Tvoice Vol 1&2

TVOtcl
Volume 1
Course Introduction
Table of Contents
Overview '
Learner Skills and Knowledge 1
Course Goal and Objectives 3
Course Flow
Additional References
Cisco Glossary of Terms
Your Training Curriculum 6
Introduction to Troubleshooting Cisco UnifiedCommunications Solutions M
Overview 1_1
Module Objectives 1"1
Identifying Cisco Unified Communications Deployments 1^3
Objectives 1"3
Cisco Unified Communications Systems 1-4
Network Infrastructure 1_5
CiscoUnified Communications Manager 1-6
Voice Clieris 1~'
Cisco Applications 1"8
Summary 1"^
Using Troubleshooting Methodology HI
Objectives 1~11
Prepare Your Network for Troubleshooting andRecovery 1-13
Troubleshoot Systematically 1"15
Problem-Solving Model 1~16
Define the Problem 1"17
Gather Facts 1'19
Consider Possibilities 1"23
Create an Action Plan 1-25
Implement anAction Plan 1"27
Observe Results 1"28
Restart the Problem-Solving Process 1-29
Document Results 1"3u"
Summary 1-31
Using Troubleshooting and Monitoring Tools 1-33
Objectives 1"33
Cisco Unified CommunicationsSystem TroubleshootingTools 1-35
Cisco Unified Serviceability 1"36
Controlling Cisco Unified CommunicationsManager Services 1-37
Alarms 1"38
Alarm Definitions 1-40
Traces 1-41
Trace Configuration Options 1-43
Trace Configuration 1-44
Customizing Trace 1-45
Trace Output 1-46
Communication Between Cisco Unified Communications Manager and SCCP
Cisco IP Phones 1-47
Call Setup Phase 1-49
Cisco Unified Communications Manager Dialed Number Analyzer 1-52
Cisco Unified Communications Manager Dialed Number Analyzer Output 1-55
Cisco Unified RTMT 1-56
Alerts 1-58
Custom Alerts on Performance Counters 1-60
Syslog Viewer 1-61
Trace&Log Central 1 co
Performance Monitor and Data Logging i eg
Performance Data Logging i gg
Cisco Unified Reporting ^.g8
Cisco Unified Communications Manager CLI 1_7q
Cisco IOS Troubleshooting Tools 1.71
Generic Call Filter Module -j_73
Sniffer Traces ^_74
Summary .
References 17c
Module Summary ^_-,-,
References 1_77
Module Self-Check 1.79
Module Self-Check Answer Key \ .37
Cisco Unified Communications Manager Troubleshooting 2-1
Overview 2-1
Module Objectives 2-1
Troubleshootinq Common Gateway and Endpoint Registration Issues 2^3
Objectives 2-3
IP Phone Initialization 2-4
Common DHCP and TFTP Issues 2-8
Troubleshooting Endpoints Using Cisco Unified Communications Manager Tools 2-9
Verify TFTP Server Configuration 2-10
Troubleshooting Endpoints from Endpoints 2-12
Cisco Unified IP Phone Network Configuration 2-17
MGCP Gateway Initialization and Communication 2-18
Cisco IOS MGCP Gateway Registration 2-20
Cisco IOS MGCP Gateway Registration Issues 2-22
Cisco IOS MGCP Gateway and Endpoint 2-24
Cisco IOS MGCP Gateway Configuration Elements 2-25
Verifying MGCP Gateway Status 2-26
MGCP Gateway Monitoring Commands 2-28
Cisco Unified Communications Manager Event Log 2-40
Cisco IOSMGCP Gateway Unsuccessful Registration 2-43
H.323 and SIP Gateway Communications 2-45
Major H.323 and SIP Gateway Monitoring Commands 2-46
Summary 2-48
References 2-48
Troubleshooting Cisco Unified Communications Manager Availability Issues 2-49
Objectives 2-49
CiscoUnified Communications ManagerSystemStops Responding 2-50
Major Causes of Not Responding 2-51
Cisco Unified Communications Manager System Log 2-52
Server Utilization 2-53
Cisco Unified Communications Manager Services 2-56
Cisco Unified CommunicationsManager Administration Does Not Display 2-57
Major Causes of Not Displayingthe AdministrationPage 2-58
Testing Network Connectivity 2-59
Verifying the Cisco Tomcat Service 2-60
Slow Server Response 2-61
Major Causes of Slow Server Response 2-62
Verifying LANConnectivity 2-63
Summary 2-64
References 2-64
Troubleshooting Database Replication Issues 2-65
Objectives 2-65
Database Replication Issues 2-66
ii TroubleshootingCisco Unified Communications(TVOICE) v8.0 2010 Cisco Systems. Ino
Database Replication Issues 2-67
Typical Database Replication ProblemScenario 2-68
Diagnosing Database Replication Issues with Cisco Unified Communications Manager 2-69
Cisco Unified Reporting 2-70
Cisco Unified RTMT Database Summary 2-71
Cisco Unified RTMT Performance Counters 2-72
Cisco Unified Communications Manager CLI 2-73
Resolving Database Replication Issues with Cisco Unified Communications Manager 2-77
Repairing Database Replication 2-78
Resetting Database Replication 2-79
Resetting the Cluster 2-83
Summary 2-84
References 2-84
Troubleshooting LDAP Integration Issues 2-85
Objectives 2-85
LDAP Integration Options with Cisco Unified Communications Manager 2-86
Cisco Unified Communications Manager LDAP Integration Options 2-88
LDAP Integration Considerations 2-89
Major LDAP Integration Issues 2-91
LDAP Integration Fictitious Issues 2-93
Resolving Synchronization Issues in Cisco Unified Communications Manager Using
Active Directory 2-94
Verify Services 2-95
Verify Service Account 2-96
Verify LDAP Directory Configuration 2-97
Troubleshooting LDAP Synchronization 2-98
Cisco DirSync Service Parameters 2-99
Setting Up Default Password and PIN for Synchronized End Users 2-100
Some End Users Not Synchronized 2-101
Resolving Authentication Issues in Cisco Unified Communications Manager Using
Active Directory 2-102
Troubleshooting LDAP Authentication 2-104
Summary 2-105
References 2-105
Module Summary 2-107
References 2-107
Module Self-Check 2-109
Module Self-Check Answer Key 2-113
Troubleshooting Call Setup Issues 3^
Overview 3-1
Examining Call Setup Issues and Causes 3^3
Objectives 3-3
Call Setup Issues 3-4
Single-Site Call Setup Failure 3-5
Intracluster Call Setup Failure 3-6
Intercluster Call Setup Failure 3-8
Summary 3^10
Troubleshooting On-Premises Single-Site Calling Issues 3-11
Objectives 3-11
On-Premises Call Setup Issues 3-12
Digit Collection in Cisco Unified Communications Manager 3-13
Digit Collection and Digit Analysis Overview 3-15
Digit-by-Digit Analysis 3-16
Digit Collection Example 3-17
Partitions and CSSs 3-18
Partition and CSS Considerations 3-20
ie, 2010 Cisco Systems. Inc Troubleshooting Cisco Unified Communications (TVOICE) v80 iii
Device and Line CSS 3-21
Time-of-Day Routing Review 3-23
ToD Routing Example 3-24
Troubleshooting Single-Site Call Setup Failure 3-26
SCCP Call Setup Flow 3-28
Tracing CSS Problems 3-31
One-Way Calling 3-36
Call-Forwarding Issues 3-37
Destination Unregistered 3-38
Forwarding to Voice-Mail Issues 3-39
Summary 3-43
References 3-44
Troubleshooting On-Net Multisite Calling Issues 3-45
Objectives 3-45
Multisite Dial Plan Issues 3-46
Overlapping Dial Plan 3-48
Intercluster Call Setup 3-50
Intercluster On-Net Call Anatomy 3-51
Tracing Call Setup via a SIP Trunk 3-53
ICT and H 323 Trunk Issues 3-64
Cisco Unified Border Element Call-Setup Issues 3-65
CCD Operation Review 3*66
Typical CCD Issues 3-69
Troubleshooting Gatekeepers 3-70
Gatekeeper H.225 RAS Messages 3-72
Gatekeeper Call-Routing Flowchart 3-74
Common Gatekeeper Issues 3-75
Troubleshooting Cisco IOS H 323 Gatekeeper 3-76
Troubleshooting Gatekeeper Registration 3-81
Verify Configuration That Can Affect the Registration 3-82
Troubleshooting Gatekeeper Call Routing 3-84
H.323 Gatekeeper CAC 3-85
Troubleshooting Gatekeeper CAC 3-87
Troubleshooting Cisco Unified Border Element 3-90
Common Cisco Unified Border Element Issues 3-92
Cisco Unified Border Element Troubleshooting Commands 3-93
Troubleshooting Cisco Unified Border ElementH.323 Side 3-99
Troubleshooting Cisco Unified Border Element-SIP Side 3-102
Cisco Unified Border Call Flow-H.323 Initiated 3-104
Cisco Unified Border Call Flow-SIP Initiated 3-105
immediate Remote-Call Drops 3-114
Summary 3-115
References 3-116
Troubleshooting Off-Net Calling Issues 3-117
Objectives 3-117
Common Off-Net Calling Issues 3-119
Gateway Troubleshooting 3-121
H.323Gateway Troubleshooting Procedure 3-124
SIPGateway Troubleshooting Procedure 3-127
Gateway Digit Collection and Analysis 3-128
Digit Collection and Dial-Peer Matching Review 3-129
Gateway Dial-Peer Matching Review 3-130
Issues with Discard Digits Instruction 3-134
Incoming Number Transformation Considerations 3-136
Outgoing Number Transformation Considerations 3-137
Cisco Unified Communications Manager Digit Manipulation Operation 3-139
Misconfigured Discard Digits Instructions Example 3-142
Troubleshooting Cisco Unified Communications (TVOICE)vS.O 2010 Cisco Systems. Inc
Overview of Cisco IOS Gateway Digit Manipulation 3-144
Voice Translation Profile Search-and-Replace Example 3-145
Dial Plan Issues ^-146
Review of Local Route Group Feature -3"14'
Cisco Unified Communications Manager Dial Plan 3-148
Verify Cisco Unified Communications Manager Dial Plan 3-150
Cisco IOS Gateway Dial Plan 3"155
Verify Cisco IOS Gateway Dial Plan 3-158
Troubleshooting Common Voice Call Issues 3-164
CallerID Issues 3'165
No-Ringback Issues: Scenario 1 3-167
No-Ringback Issues: Scenario 3 3*170
One-Way Audio Issues 3-173
Dead Air Issues 3-176
Dropped Calls 3"177
Second Dial Tone Issues 3-179
PSTN Call Setup Failure, MGCP Gateway 3-180
Tracing PSTN Call SetupDigit Analysis 3-182
Tracing PSTN Call SetupPath Selection 3-184
Tracing PSTN Call SetupISDN Call Setup, Media Setup 3-185
PSTN Call Setup Failure, H.323 Gateway 3-187
Debugging the H.323 Gateway Dial Plan 3-189
Globalized Call-Routing Issues 3-192
Inbound Call with Globalized Call Routing 3-195
Normalization of Localized Call Ingresson Gateways 3-196
Outbound Call Using Callback from Call Lists 3-197
Localized Call Egress at Gateways 3-198
Outbound Call Using Local Dialing Rules 3-200
Normalization of Localized Call Ingresson Phones 3-201
Globalized Call-Routing Issues 3-202
Verify the Configuration ThatAffects the Globalized Routing 3-203
Verifying Globalized Call-Routing Configuration Elements 3-205
Tracing Inbound Globalized Call Setup 3-208
Tracing Outbound Globalized Call Setup 3-212
Outbound ISDN Call Debug 3-216
Summary 3-217
References 3-218
References 3-219
Module Self-CheckAnswer Key 3-229
2010 Cisco Systems, Inc Troubleshooting Cisco UnifiedCommunications (TVOICE) v8 0
vi Troubleshooting Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems
TVOICE
Course Introduction
mm Overview
Troubleshooting Cisco Unified Communications (TVOICE) v8.0 prepares network
professionals with the knowledge and skills that arc required to troubleshoot Cisco Unified
Communications systems and solutions in enterprise, midmarket, and commercial deployments
in single-site and multisite environments. The course teaches troubleshooting methodology,
triage, resources, tools, and fixes at the integrated system or solution level for Cisco Unified
Communications Manager.
Learner Skills and Knowledge
This subtopic lists the skills and knowledge that learners must possess to benefit fully fromthe
course. The subtopic also includes recommended Cisco learning offerings that learners should
first complete to benefit fully from this course.
Learner Skills and Knowledqe
Working knowledge of converged voice and data networks
Working knowledge of MGCR SIR and H.323 as well as a working
knowledge of their implementation on Cisco IOS gateways
Working knowledge of Cisco Unified Communications Manager,
Cisco Unified Communications features and applications, and
Cisco IOS voice gateways in single-site and multisite
environments
amer
Cisco learning offerings:
Implementing Cisco Voice Communications and QoS
(CVOICE)
implementing Cisco UnifiedCommunications Manager,
Part 1 (CIPT1)
ImplementingCisco UnifiedCommunications Manager,
Part 2 (CIPT2)
Troubleshooting Cisco Unified Communications (TVOICE| v8 0 2010 Cisco Systems. Inc
Course Goal and Objectives
This topic describes the course goal and objectives.
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'^vtohetwork professionals with the knowledge
idsfdtlstKat are required totroubleshoot Cisco
lifted Communications systems andsolutions,
iudirt;g;jridividual components of Cisco Unified
Intriurfcafions solutions"
Upon completing this course, you will be able to meet these objectives:
Describe a systematic methodology to troubleshoot Cisco Unified Communications
solutions
Isolate and troubleshoot reported issues that relate to Cisco Unified Communications
Manager
Diagnose a call setupissueand resolvethe issuesas you discoveror reveal them, givena
trouble call for which the source of the problem is unknown
Solve the common issues of an SAF-enabled network and CCD
Troubleshoot issues that arerelated to Cisco Unified Communications Manager features
and applications
Troubleshoot voice quality issues and issues that are related to media resources
2010Cisco Systems, Inc
Course Introduction
Course Flow
This topic presents the suggested flow of the course materials.
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Commumcatons Troubleshooting
A introductitu to
Manager Troublesliooling Cisco Unified
Voice Quality and i
M
Troubleshoffling
Troubleshooting Cal-Setup Issues Communications
Media Resources i
Cisco Unified
(Cont.) (Cont.) Manager Feature
Issues (Cont.) >
Communicalions
Solusoos
Trmbleshooting
CaU-Sebp Issues
Lunch
SAF and CCD
Issues
and Application
Issues (Cont )
Troubleshooting
Cisco Unified
!
Cjsco Unified
C Ofll ITIu m C3 ll OT1S
Communications Voice Quality and
P
Troubleshooting
Troubleshooting
Manager Feature Media Resources
Manager
Troubleshooting
CaS-Setup issues
Cisco Unified
and Application Issues! Cont)
M
(Com.)
Communications
Issues (Cont.)
Manager Feature Voice Quality and
and App 6cation
Media Resources ; '
Issues Issues
*
The schedule reflects the recommended stmcture for this course. This structure allows enough
time for the instructor to present the course informationand for you to work through the lab
activities. The exact timing of the subject materials and labs depends on the pace of your
specific class.
TroubleshootingCisco Unified Communications (TVOICE) v8 0
2010 Cisco Systems, Inc
Additional References
This topic presents the Cisco icons and symbols that are used in this course, as well as
information regarding where tofind additional technical references.
Cisco Icons and Symbols
Cisco Unified
Presence
Cisco Unity
Connection
Cisco Unified
Messaging
Gateway
Cisco Adaptive
Security Appliance
Cisco Unified
Communications
Ma nagBr
Cisco Unified
Border Element
"I Cisco Unfed
*H / Personal
ltd Communicator
Cisco Unified
SRST Router
SAF-Enabled
Router
Network
Cloud
Gatekeeper
Voice Router
Cisco Unified
Communications
Manager Eipress
Cisco Unified
Communications
Manager Express with
Cisco Unity Express
Cisco Glossary of Terms
For additional information on Cisco terminology, refertothe Cisco Internetworking Terms and
Acronyms glossary of terms at
hup: doeuiki.eisco.com wiki Category:lntcmetwotkitig_ferms_and_Acronyms_(l'l A).
2010 Cisco System: , Inc Course Introduction
Your Training Curriculum
This topicpresents the training curriculum for this course.
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You areencouraged 10 jointhe Cisco Certification Community, a discussion forum open to
anyone holding a valid Cisco Career Certification (such as Cisco CCIF,\ CCNA\ CCDAk,
CCNP-. CCDP\ CCIP", CCVP\ orCCSP*), Itprovides agathering place for Cisco certified
professionals to share questions, suggestions, and infonnation about Cisco Career Certification
programs and other certification-related topics. For more information, visit
hi;p: uuw.ci^v com gu certifications.
Troubleshooting Cisco Unified Communications (TVOICE] v8 0 2010 Cisco Systems Ino
Cisco Career Certifications: CWco CCI
Expand your professional options and advance your career.
Professional-level recognition in voice networking.
| Expert 1
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Voice Nelwo rising
) 2010 Cisco Systems, Inc
Recommereled Irarang through
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and QoS
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Communications
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Appicafjons
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Course Introduction 7
Troubleshooting Cisco Unified Communications (TVOICE) v8 0 2010CtscoS st
Module 1
Introduction to Troubleshooting
Cisco Unified Communications
Solutions
^ Overview
1m*
Because of the complexity of a Cisco Unified Communications system, you must have a solid
understanding of the various elements of the voice network to troubleshoot effectively inthis
environment. This module will identify the major elements of a Cisco Unified Communications
system.
In addition, you must use a systematic troubleshooting approach to provideconsistent network
services and minimize service interruptions. This module will teach you howto prepare a
systematic troubleshooting method and howto createand implement an actionplan. Youwill
also learnhowto investigate and defineproblems, use varioustools and techniques to gather
facts, assemble an effective troubleshootingplan, and observe and document the solutions.
Module Objectives
Upon completing this module, youwill beabletodescribe a systematic methodology to
troubleshoot Cisco Unified Communications solutions. Thisability includes being abletomeet
these objectives:
Identify the major components of a Cisco Unified Communications solution to isolate
problemareas quickly during troubleshooting
Explain the steps, considerations, andrequirements to systematically troubleshoot a
problem in a Cisco Unified Communications solution
Identify- the troubleshootingand monitoringtools that can be used in a Cisco Unified
Communications solution
Troubleshooting CiscoUnified Communications (TVOICE) v8.0 2010CiscoSystems, Inc
Lesson 1
Identifying Cisco Unified
Communications Deployments
Overview
Because of the complexity of Cisco Unified Communications systems, you must have asolid
understanding of how to troubleshoot networks. This lesson provides abrief overview of the
many areas of troubleshooting that can be found in Cisco Unified Communications systems.
Before beginnmg to troubleshoot anetwork, you must recognize the broad areas that can
malfunction, for example, the failure of aCisco Unified IP phone to connect to Cisco Unified
Communications Manager docs not necessarily mean that the Cisco Unified Communications
Manager server itself is malfunctioning. The failure could be caused by network connectivity
issues asaver problem, or aphone problem. This lesson describes the major elements ofthe
Cisco Unified Commun.cations system that can help you to isolate problem areas quickly
during troubleshooting. M '
Objectives
Upon completing this lesson, you will be able to identify the major components of aCisco
Unified Communications solution to isolate problem areas quickly during troubleshooting This
ability includes being able tomeet these objectives:
Identify typical problem areas in the network that normally need troubleshooting to resolve
- Explain which part of the Cisco Unified Communications system is considered to be
network infrastmcture," explain its purpose in the Cisco Unified Communications system
and describe the types ofissues that require troubleshooting
Explain the purpose of the Cisco Unified Communications Manager in aCisco Unified
t ommumcations system and list common issues that occur with it
" w!thC T/ f Tl CMem Comf,onents that can e*ist i" "Cisco Unified Communications
system and desenbe the types of voice client issues that require troubleshooting
Explain the purpose of Cisco Unified Communications applications in aCisco Unified
Communications system and list common issues that occur with them in general
Cisco Unified Communications Systems
This topic identifies typical problem areas in the network that normally need troubleshooting to
1-4
SRST \
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_/ /Communicator i
< \ 'Cisco Unified
jr Peisondl
.f,ed Conrnunical.ons Manager Unified CME - Cisco Unified Communications Manager Expn
U-UedCU = r.iiLoUri
Cisco Unified Communications systems can be extremely complex because they not only
introduce their on set of troubleshooting elements, but they also rely on the existing network
infrastructure Because of the reliance on existing network infrastructure, aCisco Unified
Commumcations systems administrator must be adept at troubleshooting infrastructure issues
as well as application layer components such as Cisco Unified Communications Manager and
Cisco Unity.
There are four common elements ofCisco Unified Communications systems for
troubleshooting:
Cisco Unified Communications Manager.
. Cisco Unitv Connection. Cisco Unified Presence, or other Cisco Unified Communications
application's The Cisco Unified Commumcations applications are covered in the
Integrating Cisco Unified Commumcations Applications (CAPPS) v8.0 course.
Network infrastructure.
Voice clients.
Depending on the complexity of the voice network and services that you would like to add you
can introduce additional troubleshooting areas. For example, ,t you want to add auto-attendant
services to the voice network on aseparate server using Cisco Unified Communications
Manager vou could install aCisco Customer Response Solutions (CRS) server, which would
introduce its own issues on the network for future troubleshooting.
Troubleshooting Cisco Unified Communications (TVOICE) v8 0
2010 Cisco Systems, Ire
Network Infrastructure
This topic explains which part ofthe Cisco Unified Communications system isconsidered to be
"network infrastructure," explains its purpose in the Cisco Unified Communications system,
and describe the types ofissues that require troubleshooting.
Cisco IP Comnxmpcalor
o Unified Personal
^Communicator
/Cisco Unified
Personal
icator
| United CM =Cisco Unified Communications Manager Unified CME =Cisco Unified CcmmunIcations Manager Express |
The Cisco Unified Communications system infrastructure, indicated in red squares, establishes
the foundation for your entire voice network. The infrastructure isacommon source of
troubleshooting because it supports not only the new voice features and traffic patterns, but also
the existing data network. The merging of the voice network and the data network has many
benefits. It also carries considerable risk because any existing data network issues will affect
the voice network as well. For example, ifyour routing protocol requires 60 seconds to
converge and a primary link is lost, both the data and the voice networks will bewithout
service for 60 seconds. In most environments, this type offailure is unacceptable.
Because the voice network relies completely on the existing network foundation, you must
understand how to troubleshoot all current network issues in atimely manner. You may need to
upgrade many of your routers and switches to support new voice functionality such as inline
power, auxiliary VLANs, and voice interface cards (VICs). You must also understand the
operation ofthese new features and equipment and the troubleshooting methods to use during a
malfunction.
2010 Cisco Systems. Inc
Introduction toTroubleshooting Cisco Unified Communications Solutions
1-5
Cisco Unified Communications Manager
This topic explains the purpose ofCisco Unified Communications Manager in aCisco Unified
Communications svstem, and lists common issues that occur with it.
1-6
,'OscolPConin.j-
Cisco UnfiedPe-sonal
- Com-numcato'
United CM = Cisco U-iiteJ Co
al onsManage? Unified CME =Cisco Unified Communications Manager E.pn
Cisco Unified Communications Manager servers, indicated inthefigure with a red square,
provide the core voice functionality in the Cisco Unified Communications system. Cisco
Unified C-ommunications Manager isthe software-based, call-processing component that
provides the same functionality as legacy PBX systems. When troubleshooting common VoIP
issues such as one-way voice, no dial tone, or reorder tones, Cisco Unified Communications
Manager is typically the first place to look for problems. Because ofthe complex
configurations that are stored in Cisco Unified Communications Manager, you can usually
attribute most of thecommon voice issues to a configuration problem.
Troubleshooting Cisco Unified Communications (TVOICE) v80
Voice Clients
This topic lists the range of voice client components that can exist in a Cisco Unified
Communications system and describes the types of voice client issues that require
troubleshooting.
Voice Clients
Communicator t
Unified
Personal
Communicator
Unified CM=CiscoUnified Communications Manager Unified CME- CiscoUnified Communications Manager Expres
In some instances, voice clients can cause voice network malfunctions. Voice clients include,
but are not limited to, the following:
Cisco Unified IP phones, including Cisco IP Communicator, Cisco Unified Video
Advantage, and Cisco Unified Presence Client
Cisco Unified Communications Manager Attendant Console, Cisco WebAftendant,or both
Analog devices such as fax machines or legacy telephones
Either the end user or the network administrator might incorrectly configure or connect these
devices tothenetwork. Youmight alsofind certain devices tobeincompatible withyourCisco
Unified Communications system. In any event, you shouldconsiderclient deviceswhen
performing any troubleshootingon the voice network.
The remainder of this course does not focus on troubleshooting voice client issues; these issues
are typically simpler in nature, andanadministrator canusually solve these problems bya
physical check of the device.
Introduction to Troubleshooting Cisco UnifiedCommunications Solutions 1-7
Cisco Applications
This topic explains the purpose of Cisco Unity Connection and other Cisco Unified
Communications applications in a Cisco Unified Communications system and lists common
issues that occur with Cisco Unity in general.
..iilli
Un.l'Sd Cf.' = Cisco U-nfud Cut ations Manaoei u-vfied CME - Cisco Unified Communications Manager Fiin'i
Almost even,' corporate voice network that requires IP telephony services also needs voice-matl
services. CiscoUnityprovides unified messaging services, suchas voice mail, email, andfax,
to your IPtelephony network. CiscoUnitycan alsobe a common troubleshooting area because
of its advancedconfiguration. When you troubleshoot issues such as Message Waiting
Indicators (MWis), voice-mail audio levels, or automatic call transfers, you usually examine
the Cisco Unity server first.
Because Cisco Unity is integrated with Cisco Unified Communications Manager, much
troubleshooting is necessary because of configurationerrors between the two devices.
TroubleshootingCisco Unity problems can lead to troubleshootingCisco Unified
Communications Managerproblems. The initial configuration and communication between
Cisco Unified Communications Manager and Cisco Unity is critical, because that process sets
the stage for your Cisco UnifiedCommunications systemtroubleshooting.
In some instances, other Cisco Unified Communications applications, such as Cisco Unified
Contact Center FxpTess, Cisco Unified MeetingPlace, or CiscoUnifiedPresence, can cause
voice network malfunctions.
Troubleshooting Cisco Unified Communications (TVOICE] v8 0
>2010 Cisco Systems, Inc
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
There are four common areas for troubleshooting Cisco voice
networks: Cisco Unified Communications Manager, Cisco
Unified Communications applications, network infrastructure,
and voice clients.
Cisco Unified Communications Manager is typically the first
place to look for problems when troubleshooting common
VoIP issues.
The network infrastructure, which supports both voice and
data networks, is a common source of problems that require
troubleshooting.
Incorrect configuration of voice clients is a common source of
problems that require troubleshooting.
The advanced configuration of Cisco Unity Connection is a
common source of problems that require troubleshooting.
In this lesson, you have learned to identify the major components of a Cisco Unified
Communications solution to isolate problem areas quickly during troubleshooting.
2010 Cisco Systems,
1-10 TroubleshootingCisco Unified Communications (TVOICEl v8 0 201(3 Cisco Systems. Inc
Lesson 2
Using Troubleshooting
Methodology
Overview
Preparing a systematic troubleshooting method requires the identification ofessential tools and
procedures for troubleshooting IP telephony and data network systems. This lesson teaches you
basic procedures for identifying network system problems. You will learn touse these methods
toassist you introubleshooting nearly every type of network issue thatyoumay encounter.
Totroubleshoot your environment effectively, you must systematically define the problem,
gather the pertinent data, and consider the potential causes. It is important todevelop astandard
troubleshooting procedure within your organization. This lesson serves as anexample template
on how to troubleshoot a problem.
Objectives
Upon completing this lesson, you will beable toexplain thesteps, considerations, and
requirements totroubleshoot systematically a problem ina Cisco Unified Communications
solution. This ability includes being able to meet these objectives:
Explain how you can simplify the troubleshooting process bymaking certain preparations
Describe thegeneral problem-solving model that youshould follow to help resolve
troubleshooting issues systematically
Explain how tocreate a clearproblem statement in theCiscoUnified Communications
systemto helpto start withthe troubleshooting procedure
Explain the processof collecting information about a reportedCiscoUnified
Communications system problem to help to isolate possible causes
Explainthe process of identifying the possible causesof a reportedCiscoUnified
Communications systemproblem that is basedon the factscollected
Explain howtocreate an action planforthe troubleshooting task that is based on theknown
causes
Explain howto ensurethe greatestpossibility of successof the troubleshooting planduring
its implementation
Explain howto confirm the success of the troubleshooting task before closing down the
trouble ticket
Explain whatto do if the troubleshooting task fails
Explain why and how to keep a record of your troubleshooting activities
Troubleshooting Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems, Inc
Prepare Your Network for Troubleshooting and
Recovery
This topic explains how you can simplify the troubleshooting process by making certain
preparations.
Cisco Unified Communications System
Possible sources of problems:
* Complex deployments
Various protocols
Various endpoints and services
Network infrastructure
^, CiscoP Communicator
^ Ufi*&d Personal
Communicator
Beforeconsidering systematic troubleshooting methods, you must answerthe following
questions regarding your preparedness to manage a VoIP network outage:
Do youhave an accurate physical and logical mapof your internetwork, showingthe
physical locationof all VoIP devices on the network and how they are connected, and a
logical map of the network addresses, network numbers, and subnetworks?
Do you havea list of all network protocols that are implemented inyour networkand a list
of the network numbers, subnetworks, zones, and areas that are associated with those
network protocols?
Do you knowwhich protocols are being routed and the correct, up-to-date configuration
information for each protocol?
Do know which protocols are being bridged? Arc there any filters configured in any of
these bridges, and do you have a copy of these configurations? Arc any of these protocols
applicable to Cisco Unified Communications Manager?
Do you know all of the points of contact to external networks, including any connections to
the Internet? For each external network connection, do you know which routing protocol is
being used?
Has your organization documented normal network behavior and performance so that you
can compare current problems with a baseline?
Faster recovery from a network failure should result if you can answer yes to several of these
questions.
12010 Cisco Systems, Inc Introduction to Troubleshooting Cisco Unified Communications Solutions 1-13
This figure describes why a systematic troubleshooting methodology is important and what
such a troubleshooting model could look like.
Cisco Unified Communications systems can be extremely complex because they not only
introduce their own set of troubleshooting elements, but they also rely onthe existing network
infrastructure. Because of the reliance on the existing network infrastructure, a CiscoUnified
Communications systems administrator must beadept at troubleshooting infrastructure issues
aswell asapplication-layer components such asCisco Unified Communications Manager,
Cisco Unity Connection, Cisco Unified Presence, Cisco Unified Communications Manager
Express, or Cisco Unity Express.
There arefour common elements of Cisco Unified Communications systems for
troubleshooting:
Call processing systems such asCisco Unified Communications Manager, Cisco Unified
Communications Manager Express, Cisco Unified Border Element, or gatekeeper
Unified communications applications such as Cisco Unity Connection, Cisco Unity
Express, Cisco Unified Presence, or Cisco Unified Contact Center Express
Unified communications endpoints such as Cisco Unified IPphones, Cisco IP
Communicator. CiscoUnified Personal Communicator, or third-party Session Initiation
Protocol (SIP) endpoints
Network infrastructure
Typically, the complexity and. consequently, thepossible causes of issues depend on thesize of
thedeployment, thenumber of protocols, thenumber of applications, andthenumber of
countries (consider differences inpublic switched telephone network [PSTN] numbering plans,
for example),
1-14 Troubleshooting Cisco Unrtied Communications (TVOICE) v8 0 2D10 Cisco Systems, Inc
Troubleshoot Systematically
This topic describes the general problem-solving model thatyoushould follow tohelp resolve
troubleshooting issues systematically.
Troubleshooting Methodoiof
Required in Complex Enviro
Advantages ofa systematic troubleshootingmethod:
Easier to identify potential problems.
* Learning effect:
Understanding what is going on when there is no problem:
Tracking down internal processes such as call flows
Knowing comer case
Better ability to spot anomalies
Systematic documentation helps to solve future issues.
Asystematic troubleshooting method makes it easier for you toidentify potential problems on a
data network and aCisco Unified Communications system. You can use atroubleshooting
model to reduce methodically a largeset of possible causes of trouble toa smaller set or toa
singlecause. Then youcan fixthe problemand restoretheCisco Unified Communications
system.
After you resolve aproblem, asystematic process ofdocumenting the case helps you capture,
preserve, and communicate the experience that you gained while solving the problem. You can
also refer to this document ifsimilar problems arise later. Such amethod helps you increase the
expertise ofthe organization and reduces the time that you will spend solving future problems.
12010 Cisco Systems, Inc.
Introduction toTroubleshooting CiscoUnified Communications Solutions 1-15
Problem-Solving Model
The figure shows an example of a problem-solving model that uses a systematic approach for
troubleshooting.
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This troubleshooting model consists of these steps:
Step 1 Analyze the network problem and create a clear problem statement. Define
symptoms and potential causes.
Step 2 Gather the facts that you need tohelp isolate possible causes and use the results to
narrow the list of potential causes tothe most likely source. Ideally, youwould
narrow the problem down tothe root cause. Ifyoucannot determine one potential
source of the problem, exclude as many options as possible bygathering facts by
using tools ot methods that areappropriate for theproblem.
Step3 Basedon the facts that you gather, consider the most likelypossible causes.
Step 4 Create anaction plan for the possible causes. Begin with the most likely problem
and devise a planwhereyou manipulate onlyone variable.
Step 5 Implement the action plan.
Step 6 Analyze the results todetermine ifthe problem has been resolved.
Step 7 Ifthe problem has not been Tesolved, create anaction plan that is based onthe next
most probable cause onyour list. Return to Step4 andrepeat theprocess.
Step 8 Ifthe problem isresolved, consider ihe process complete and document any
changes, the root cause ofthe problem, and the steps that resolved the problem.
Note If you have not previously approached problems systematically orhave not considered using
a problem-solving model, you may find that it takes longer initially. Ultimately. II saves time.
1-16 Troubleshooting CiscoUnified Communications (TVOICE) v8 0
Define the Problem
This topic explains how to create aclear problem statement in Cisco Unified Communications
system tohelp start the troubleshooting procedure.
Define the Problem
Network Problem:
Auser in cluster NY initiates a call toa user in cluster FRA
The IP phone in FRA rings, but as soon asthe user answers
the call, the user hears a fast-busy signal.
Cluster NY
Cluster FRA
Example: Network Problem
The figure shows asample network problem. You receive aproblem report that states the
Following:
Auser in Cluster NY initiates acall to auser in Cluster FRA. The IP phone in FRA rings but
as soon as the user answers the call, the user hears afast-busy signal.
Define the Problem
Refer to the problem-solving model as you go through the problem-solving process.
Asystematic approach to troubleshooting consists of asequence of steps. First, define the
problem clearly and sufficiently. To make aclear problem statement, define the problem ,n
terms of aset of symptoms and the associated causes. Ideally, when defining the problem
compare your current network configuration and performance against anetwork baseline.'
Abaseline is aset of data that is collected from targets after installation. Use the baseline
data for comparison with real-time data.
Note
orobut i " ' ^ ,dT,fy the genera' SymptmS' NeXt' detene what possible pobems thcse m]ght mdjcatc ^ ^^^^^ P
established network baselines. You should be able to identify the characteristics of the netTo k
nttwo kehnemh 'S P/rf0n"7 "XPeCtCd In 3ddit]0n' ^ mUSt know *** "^
network have changed since the last record of baseline performance.
Introduction to Troubleshooting Cisco Unified Communications Solutions
1-17
1-18
When referring to the sample network problem, your definition of the problem should be very
like the problem report itself. For example, the user in Cluster NY reported that when
attempting to call another user in Custer FRA, the telephone rings but returns afast-busy signal
when answered.
While vou are defining the problem, you should already be formulating the possible causes. For
example you might remember recently modifying the Cisco Unified Commumcat.ons Manager
region configurations. This can give you ahead start in the right direction and might provide a
quickfix to the problem.
" m,nirp\vfin 2010C,sc0 systems' lnc
Troubleshooting Cisco Unified Communications (TVOICE) v8.0
Gather Facts
This topic explains the process of collecting information about a reported Cisco Unified
Communications system problem to help to isolate possible causes.
Gather Facts
Yes
The next step in the problem-solvingmodel is to gather facts. When you gather facts,
accurately interviewend users to get all of the pertinent details of the problem. Here are some
example questions to use when interviewing an end user:
When did the problem first occur?
Is the problem intermittent or does it always occur?
If it is intermittent, is there a pattern to the time of day that it occurs?
Are thereany messages or tones that play, or does thedevicesimplydisconnect?
What does the display of the device read when the problemoccurs?
Is anyoneelse that youknowhavingthis problem, either near youor near the partythat you
are catling?
Which digits arc being dialed? Is the call supposed to go across an intercluster trunk (ICT)
or through the PSTN?
This list is not exhaustive butprovides a place tostart. Based onyour knowledge of thesystem
andtheproblem that isbeingreported, youmight come upwithadditional questions.
Some businesses have front-line interviewers who use a triagescript with end usersto
determine at which level to manage theproblem. This course does not require the use of scripts;
however, they might work well in your environment.
Note
)2Q10Cisco Systems, Inc.
Itis also important for you to know whatworks forthe user. Forexample, ifa user is having
problems calling one extension, find out whatextensionsthe user cancall. This knowledge
will assist in narrowing the scope of the problem.
Introduction to Troubleshooting Cisco Unified Communications Solutions 1-19
In addition, you should question other key people that are involved with the network, such as
network administrators and managers. These queries will help you understand the desired call
path that the failed call is taking.
To gather information, you can use internal or external tools to help collect data. Internal tools
are methods that you can use directly on Cisco equipment. Some examples of internal tools are
the show and debug commands on routers and switches, the device and search pages of the
Cisco Unified Communications Manager server, the Cisco Unified Real-Time Monitoring Tool
(RTMT) Performance Monitor, and Cisco Unified Communications Manager traces.
External tools are the methods that you can use to gather information from sources that are not
directly related to Cisco cquipmenl. Some examples of exiemal tools are protocol analyzers and
network management systems.
1-20 Troubleshooting Cisco UnifiedCommunications (TVOICE) vB0
Gather Facts (Cont.)
Collect the facts:
Interview end users and place test calls. Any progress tone heard?
VerifyCisco Unified Communications Manager configuration.
Verifyconfigurations of routers, run debugs.
* Use external tools such as network analyzers or Cisco Unified
Communications Manager trace.
Cluster NY Cluster FRA
As the sample network problemshows, you should first interviewthe end user. The user can
provide critical information that you might not find in an email trouble ticket or from a
telephone call. Here is a sample interviewof the user of the IP phone in NY:
Question: Doyou alwaysreceivea fast-busy signal whencallingIPphoneB?
Answer: This problemseems to happen intermittently. It started about two months
ago, but onlyoccurred fromtime to time. Recently, it has beenoccurring quite
frequently.
Question: Have youchanged theconfiguration of yourtelephone inanyway?
Answer: No.
Question: What time of day does the problemoccur?
Answer: The problem seemsto occur most duringbusytimesof the day.
Question: Do calls to IP phones at the same site as phone Awork?
Answer: Yes.
Question: Do calls to other phonesat the samesite as phoneFRAwork?
Answer: Sometimes.
Afier yougatherandconsider theinformation that theenduser provided, youcanbegin to
verifyyour CiscoUnified Communications Managerand router configurations. Yourun
debugs or CiscoUnifiedCommunications Managertracesto gather additional facts. After
verifying the collected data, you might realize that theCisco Unified IPphones inCluster NY
are all first-generation CiscoIPphones that support G.711 and G.723coder-decoders (codecs).
The Cisco IPphones in Cluster NY are second-generation Cisco IPphones, which support
G.711 and G.729 codecs. Upon verifying the Cisco Unified Communications Manager region
configuration, you seethat you have Cisco Unified Communications Manager that is
configured to require a G.729 codec when communicating between clusters. However, because
the user informed you that the problem isintermittent, do you think the problem directly relates
to this configuration issue?
2010CISCO Systems, Inc
For these types of problems, tools that can gather information in real lime can also be useful.
For example, you could have the Cisco IP phone user in NY attempt to call the Cisco Unified
IP phone user in FRA while logging the call setup process through a Cisco Unified
Communications Manager trace, or capturing data through a network analyzer.
1-22 Troubleshooting Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems, Inc
Consider Possibilities
This topic explains the process of identifyingbased on the facts collectedthe possible
causes of a reported Cisco Unified Communications system problem,
Slart Finished
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Yes
No
This topic describes the process of determining the potential problems on your IP telephony
network.
After you have gathered all of the available facts, you should consider the potential problems
that are based on those facts. You can set boundaries to help isolate the network problems.
Remove irrelevant network details from the set of items to check. You can also eliminate entire
classes of problems that are associated with system software and hardware.
Possible problems:
' Cisco Unified Communications Manager:
Incorrect region configured
No transcoding resource
Cluster NY Cluster FRA
By brainstorming with the data that is gathered from the sample network problem, you might
come up with a few possible causes thai could include the following;
Incorrect region definitions
No transcoding resource
Use the appropriate tools or methods to gather facts to try to eliminate as many of the options
as possible. In this example, the following information is gathered:
You use a network topology diagram and discussions with the network administrator to
discover the call path. The call path goes across an ICT (Frame Relay-based) between the
two clusters.
Through further discussions with the administrator, you learn that the endpoints in cluster
NY can use G.711 or G.729, and the devices in cluster FRA only support G.711 and G.723.
Because of bandwidth constraints. G.71 I cannot be used. The region settings are ruled out.
and the administrator confirms that the codec settings are correct and have worked in the
past.
You eliminate the "router access list" and "RTP header compression mismatch" options
because, on occasion, the calls do connect.
The most likely remaining possibility is a lack of transcoding resources. Use the
Performance Monitor counters in Cisco Unitied RTMT to confirm this possibility. In
addition, to help confirmthe root cause of the problem, you can set an alert to send a
message when no transcoding resources are available.
After considering these possibilities, you should create an action pian for the most likely
solution.
Create an Action Plan
knl(!irP,8inS hW t0 CrCatC a" aCti" Pla" fr the troubleshoting task that is based
Create an Action Plan
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When creating an action plan, you should employ a"divide and conquer" policy if multiple
hTmethoTs thaT ^ ShU,d ^ <* M^ *** ** ^ -7 *Le
me methods that you can use tocorrect the problem.
Break the problem into small steps. Start from the troubled device and work outward At each
Cm determine ,f the network is functioning properly. This process will help you trl apath
from the source ofthe problem to the destination. P m
sFpcc!ffc IS!^ttheFS VeVel0P " aCtin P'an' eSpecial1^ **>* Pti" in a
specific area. It will save time and give you experience in other areas.
2010CiscoSystems, Inc
introducton to Troubleshooting Cisco Unified Communications So.u.ions
1-25
1-26
Possible action plan:
- Add additional transcoding resources to the larger cluster NY.
Cluster NY
Cluster FRA
After vou consider all possibilities and narrow your options down to the most likely cause 01.
de 1vto the confirmed root cause, the action plan should reflect asolution that ,s directly
Sattdto the problem. In the sample problem, it was confirmed that the transcoding resou ces
were runnin* short during heavy usage times. The only solutions to this problem are to add
additional transcoding resources or to limit the number ol calls.
Depcnditm on the overall cost, you might decide to replace the Cisco Unified IP phones in the
luster *ith Cisco Unified IP phones that support the G.729 codec mstead. Ot course, before
vou make any purchases, you should perform extensive monitoring of the transcoding resource,
toensure that this is the true cause of the problem.
Implement an Action Plan
This topic explains how to ensure the greatest possibility of success of the troubleshooting plan
during its implementation.
Implement an Action Plan
Finished
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No
When developing and executing the action plan, be as specific as possible. The plan must
identify a set of steps that you will execute, and you must carefully implement each step. Keep
track of exactly what you are testing. It is best that you list the action plan in a step-by-step
process on paper. Use this documented, systematic action plan to take notes while
implementing the plan, tracking successes and failures. Never change more than one variable at
a time, because, otherwise, it will be difficult to determine the ultimate solution to the problem.
The following arc other items that you should consider during the implementation of the action
plan:
Make sure that the changes that you made do not make the problemworse. If the changes
do make the problem worse, you should reverse the changes.
Limit the impact of the changes on other users.
Minimize the extent or duration of potential security lapses, such as removing an access
list.
In addition to fully backing up Cisco UnifiedCommunications Manager, you should maintain
backup configurations of the routers and switches in your network.
Observe Results
1-28
This topic explains how to confirm the success of the troubleshooting task before closing down
the trouble ticket.
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After manipulating a variable to find a solution to a problem, gather results that are based on
the action plan and determine if you have resolved the problem. If you do resolve the problem,
document the solution in addition to the action plan process.
| 2010 Cisco Systems Inc
Restart the Problem-Solving Process
This topic explains what to do ifthe troubleshooting task fails.
Restart the Problem-Solving Pn
- - -lT
Use (tie Process
After you observe the results and determine that the problem still exists, restart the problem-
solving process with the remaining possibilities that you identified when gathering facts. With
the result ofthe last action plan, you can narrow the possibilities. Your narrowing ofthe
possibilities should be anongoing process.
Ifthe previous action plan is avalid action that results in adesirable configuration, but does not
solve the root cause, then leave the action implemented and create another action plan. Ifthe
previous action does not solve the problem, and you do not consider the results adesired
permanent state, then back out ofthe action before you reiterate the process ofcreating anew
action plan.
Introduction to Troubleshooting CiscoUnified Communications Solutions 1-29
Document Results
This topic explains why and how to keep arecord ofyour troubleshooting activities.
focumonl Results
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As soon a-, you resolve the problem, you must document your work according to the practices
and procedures of the enterprise.
Reasons for creating documentation include the following;
Documentation maintains the exact steps that you took to solve the problem.
Documentation provides you with a back-out plan in case the fixes applied worsen the
situation over time.
Documentation of the problem and resolution serve as a historical record for future
reference.
1-30 Troubleshooting CiscoUnified Communications (TVOICEl v80
fi^^
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my
Summary
' You can recover more easily from a failure if you are
prepared in advance.
To troubleshoot a Cisco Unified Communication system or
data network, you must use a problem-solving model This
model includes eight steps.
Define the symptoms of the network problem clearly and
understandably.
Use internal and external tools togather facts.
Summary (Cont.)
Isolate the network problem and consider the possible
causes.
' Develop an action plan.
1Implement anaction plan, and be asspecific as possible.
Evaluate the effectiveness of theaction plan.
If theproblem persists, restart the process from the
possibilities that are based on the facts previously gathered.
Document the results oftheaction plan.
^nbWh T yUMaVe leamed t0 eXPlam thG StepS' codîons, and requirements to
troubleshoot aproblem systematically in aCisco Unified Communications solution.
)2010 Cisco Systems. Inc
1-31
,Tl..r,, n " 2010 Cisco Systems Inc
1-32 Troubleshooting Cisco Unified Communications (TVOICE) vB u
Lesson 3
Using Troubleshooting and
Monitoring Tools
Overview
Jcôrnifi^r " a" VemeWf,hC COmmn **"* ft0,S that yU CM to troubleshoot
aCisco Unified Commun.cat.ons system deployment. Many tools exist on the Cisco Unified
Communications Manager and most of them are in the Cisco Unified SerîceabiiiN menus the
Cisco Unified Real-Time Monitoring Tool (RTMT), or the command-line intert" On
hv'n n y;Van0U r W and debU CmmandS Xist This cêrs these commandos
issues arise with aCisco Unified Commun.cations Manager deployment.
Objectives
Upon completing this lesson you will be able to identify the troubleshooting and monitoring
^:;^ÛmfiCd C^nssolution. This abfiity mcludeSg
' tD7CriH û CTUmfied Serviceability "and the associated tools that you can use-
to troubleshoot Cisco Unified Communications Manager
Use the Cisco Unified Serviceability Control Center to stop, start, restart and view the
status of services on the Cisco Unified Communications Managed server
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Describe how to interpret basic trace output
" tmuhl ' hCi?,Urfd Collunicatioils Manager Dialed Number Analyzer tool to
troubleshoot dial plans ,n the Cisco Unified Communications Manager configuration
^;^^rUnifiedRTMT and exp,ain how hcan be - *
Describe how to view, set, and modify both predefined and customized alerts in Cisco
Unitied RTMT
Describe how to view the local syslog files in the Cisco Unified RTMT
Describe how to collect trace and log tiles and view them in the Cisco Unitied RTMT
Enable and use Performance Monitor data and monitoring to assist you and Cisco TAC
with troubleshooting
Explain how to use the Cisco Unified Reporting tool of the Cisco Unified Communications
Manager toaid introubleshooting
Provide an overview of how to use the CI.I of Cisco Unified Communications Manager
Version 8 to aid inthe troubleshooting process
Provide an overview of how to use the Cisco IOS Software to support in troubleshooting
Describe how to use ageneral network sniffer to troubleshoot protocol issues
Trouble
shooting Cisco Unified Communications (TVOICE) v8 0
Troubleshooting Tools
Th
topic provides an overview ofthe different types oftools that are useful when
troubleshooting aCisco Unified Communications system.
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CiscoUnified Serviceability
Alarms
Selling Trace
Csco Unified Communeatens
Manager CAR
- Control Center
DialedNumber Analyzer
' Cisco Unified RTMT
Alerts
Viewing Trace
Syslog Viewer
Performance Monitonng
CLI
Other Troublestiooling Tools
* Packet Sniffer
* Cisco Unified Operations
Manager
Galeway Troubleshooting Tools
show commands
debug commands
Many tools are available to help an administrator troubleshoot aCisco Unified
Communications system. Cisco Unified Serviceability is aweb-bascd troubleshooting tool for
Cisco Unified Commun.cations Manager that provides many functions to help troubleshoot.
Csco IOS gateways have many show and debug commands that can also be useful when
troubleshooting Cisco Unified Communications system issues.
froubllshoTr'r"' CnCtW,0r,kS SOftWarC 3re CXamplCS fther S0fttools that can help you
troubleshoot a Cisco Unified Communications system.
Many Cisco web-based tools can help you troubleshoot issues with Cisco Unified
Communications systems:
Tech Tips: The most common issues that Cisco Technical Assistance Center fTAC)
manages aretypically written upinTech Tips.
" SZTACC"s? Co,lertion: ThiS SCarchabic col,et--ti0" ^Cisco TAC cases includes the
ability tosearch byentering a sentence.
' n 'TPreIer: ! ^Wcshooting tool reports potential problems by analyztng
supported showcommand output. *
Search Engine: The Cisco.com website is indexed, and you can search it for documents.
)2010Cisco Systems, Inc.
Introduction to Troubleshooling Cisco Unified Communications Solutions
1-35
Cisco Unified Serviceability
1-36
This topic describes the Cisco Unified Serviceability menus and the associated tools that you
can use to troubleshoot Cisco Unified Communications Manager.
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Cisco Unified Serviceability is aweb-based troubleshooting tool for Cisco Unified
Communications Manager that provides the following benefits:
Saves Cisco Unified Communications Manager Service alarms and events for
troubleshooting and provides alarm message definitions.
. Sixes Cisco Unified Communications Manager Service trace information to various log
files for troubleshooting. You can configure traces in Cisco Unified Serviceability and
collect and view trace infonnation in the Cisco Unified RTMT.
. Pros ides feature services that you can activate and deactivate in the Service Activation
window.
. Provides an interface for the starting and stopping feature and network services.
Provides an interface to view the status offeature and network services.
. Generates reports for Call Detail Record (CDR), Cisco Unified Communications Manager
CDR Analysis and Reporting (CAR), and Cisco Unified RTM I.
. Provtdes CDRonDemand. which allows you to retrieve the CDR and Call Management
Record (CMR) files from Cisco Unified Communications Manager.
. Mlows Csco Unified Communications Manager to work as amanaged device for Simple
Network Management Protocol (SNMP) remote management and troubleshooting.
. Monitors the disk usage of the log partition on aserver, or all servers in the cluster.
Troubleshooting
Cisco Unified Communications (TVOICE) v80
Controlling Cisco Unified Communications
Manager Services
This topic describes how tousethe Cisco Unified Serviceability Control Center tostop, start,
restart, andview the status of services on theCiscoUnified Communications Manager server.
Cisco Unified Serviceability Control Center
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Youactivateanddeactivate feature services in the ServiceActivation windowin CiscoUnified
Serviceability. Services that display in the Service Activation window do not start until you
activate them,
Cisco Unified Serviceability allows you to activate and deactivate only feature services (not
network services). You can activate ordeactivate as many services as you want at the same
time. Some feature services depend onother services, and the dependent services areactivated
before the feature service activates.
The Control Center in Cisco Unified Serviceability allows you toview the status, refresh the
status, and to start, stop, and restart the feature and network services.
Starting, stopping, or restarting a service causes all Cisco Unified IP phones and gateways that
are currently registered to that service to fail over to their secondary Cisco CallManager
service. Devices and phones need torestart only ifthey cannot register with another service.
Starting, stopping, orrestarting a service causes other installed applications (such as a
conference bridge or Cisco Messaging Interface) thatarc homed to theCiscoUnified
Communications Manager to startandstopas well.
Caution
Stopping a service alsostopscall processing for all devices thatthe service controls. When
a service isstopped, calls from anIP phone toanother IP phone stay up; calls in progress
from anIP phone to a Media Gateway Control Protocol (MGCP) gateway also stay up, and
other types of calls are dropped.
Introduction toTroubleshooting CiscoUnified Communications Solutions 1-37
Alarms
This topic defines alarm destinations and levels, lookup alarm definitions, and describes how to
add user notes to alarm definitions.
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Cisco Unified Serviceability Alarms assist system administrators and support personnel in
troubleshooting Cisco Unitied Communications Manager problems. Administrators use alarms
to receive notification that an event has takenplace. Alarms contain information suchas an
explanation ofthe event and recommended action. Alarm information includes application
name, machine name, cluster name, recommended action, and user-definedtext that can be
custom-defined.
You can configure alarms for Cisco Unified Communications Manager servers that are in a
cluster andservices for each server, such as Cisco Unitied Communications Manager, Cisco
TFTP. and Cisco CTIManager. You configure the alarm interface tosend alarm information to
multiple destinations, and each destination can have its own alarm event level (from debug to
emergency). Then, you use the Cisco Unified RTMT to collect and view the alarms.
Whena service issuesan alarm, the alarminterface sendsthe alarmto thechosenmonitors,
such as, for example, system diagnostic interface (SDI) trace, or Cisco Real-Time Information
Server (RIS) data collector. The monitor forwards the alarm or writes ittoits final destination,
such as a log file.
Configure alarms per sener and per service, which allows for setting specific alarm levels to
assist introubleshooting. Follow these sleps toconfigure alarms:
Step 1 Choose Alarm >Configuration.
Step 2 Choose the server.
Step 3 Choose the service.
1-38 Troubleshooting CiscoUnified Communications (TVOICE) v8.0
Step4 Check the box or boxes foryourdesired alarm destination.
You can set alarms to go to any offour destinations, and you can set the level of
alarms individually for each of the four destinations, as follows:
Local Syslogs: Use Cisco Unified RTMT to collect or view this output.
Remote Syslog: Go to the syslog server to view the output.
SDI Trace: Use Cisco Unified RTMT to collect orview this output.
Signal distribution layer(SDL) Trace: Use Cisco Unified RTMT tocollect or
view this output.
Step5 In theAlarm Event Level drop-down menu, click thedown arrow.
Step6 Click thedesired alarm event level for each of thedestinations.
The following eight levels ofalarms are listed in the order ofoutput that is
generated, startingwiththe least output:
Emergency(least output)
Alert
Critical
Error (Default)
Warning
Notice
Informational
Debug(most output)
2010 Cisco Systems, Inc. Introduction to Troubleshooting Cisco Unified Communications Solutions
1-39
Alarm Definitions
1-40
This section describes how to create yourown, customized alarm definitions.
*i.*u.-s2&^l^feato3&&SS3&>siB^'#ta^^Wi, mu^****.*,
\\4 * M
'*n*j*i' ,,.-. I.^CJ ttj.,^' td-^"-,^! . E.it.-alai.f. N3T.* hind '
^ i; u- mi fs' TiibMjr-age1
- ifeWnnwmW*k*
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SDL "I T; rufKrtfe a^LltTon
IrirOimar.HLi.al.-ir'lT, no action
^l. -1 *" -*r-ot* arrferJ^-Jn
in rue Cism unified flaprtmrij
-"'.ear Fr,rjte Uiîs.-i CM ticn#s..
Thi iarff: d&io 1*nuaufls IT.?
' *-, -!-, rrfUi ou" or ,t--*>c
ti^ji tha Css.* LaHMar.u *d
Intanmaticisl ourpM onhf. n
thed th* conntrton ef the Ti .
Nctiftcation purooas* only, no.
i Wvi'.v jT*TtptetitO iffjite'
K the O&'.rj Unified Pepo-tmfl
iir.a-atL'-t atttt(iutd it reei*
liivtitigatu jiiv ng[-urVt lUtntc
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?.ge I of 15
Cisco Unified Communications Manager stores alann definitions and recommended actions in
aStructured Query Language (SQL) server database. You can search the database for
definitions ofall ofthe alarms by using aweb-based interface in Cisco Unified Serviceability.
The definitions include the alann name, description, explanation, recommended action,
severity, parameters, and monitors. This information helps you troubleshoot problems that
Cisco Unified Communications Manager encounters.
Traces
Thistopic describes how to configure tracesettings inCiscoUnified Serviceability andexplain
where the generated trace output can be read.
Cisco Unified Communic;
Manager Traces
Cisco Unified Serviceability provides trace tools to
assist in troubleshooting.
Trace options:
System Diagnostic Interface provides log tiles that contain
the run-time events that occurred at the involved software
routines. Used for administrator troubleshooting.
' Signaling Distribution Layer logs call-processing information
from Cisco CallManager and Cisco CTIManager services.
Used by Cisco engineers to find the cause of an error.
* Log4J is used for Java applications.
Trace log files are collected and viewed by Cisco
Unified RTMT.
Cisco Unified Serviceability provides trace tools to assist you in troubleshooting issues with
your voice application. Cisco Unified Serviceability supports the following trace options:
SDI trace: SDI traces are local log files. Every Cisco Unified Communications Manager
Service includes a default trace log file. The IP address, TCP handle, device name, the time
stamp, and many more can be used when reviewing the SDI trace to monitor the occurrence
or the disposition of a request. SDI traces log the run-time events that occurred at the
involved software routines.
SDI traces will be explained in detail throughout this course.
SDL trace: (For Cisco CallManager and Cisco CTIManager services, applicable to Cisco
Unified Communications Manager and Cisco Unified Communications Manager Business
Edition only.) The SDL trace log file contains call-processing information from services
such as Cisco Unified Communications Manager and Cisco CTIManager. The system
traces the SDL of the call and logs state transitions into a log file.
Cisco engineers use SDL traces to find the cause of an error. You are not expected to
understand the information that is contained in an SDL trace; however, while working with
Cisco TAC, you may be asked to enable the SDL trace and provide it to the Cisco TAC.
Log4J trace: Log4J trace is used for Java applications. You use the Trace Configuration
window to specify the level of information that you want traced as well as the type of
information that you want to be included in each trace file.
In the Alarm Configuration window, you can direct alarms to various locations, including SDI
trace log files or SDL trace log files. If you want to do so, you can configure trace for alerts in
ihe Cisco Unified RTMT.
i 2010 Cisco Systems. Inc Introduciion to Troubleshooting Cisco Unified Communications Solutions
After you configure information that you want to include in the trace files for the various
services, youcancollect andview trace files byusing theTrace & Log Central option in the
CiscoUnified RTMT. With the Trace& LogCentral option, you can collect SDLand SDI
traces, application logs, system logs (such as event view application, security, andsystem logs),
and crash dump files. Trace files can also be collected from the Cisco Unified Communications
Manager servers to the PCof the administrator and then viewed locally.
Tip Do not use Notepad to view collected trace files.
1-42 Troubleshooting Cisco Unified Communications (TVOICE) v8.0 2010 Cisco Systems, inc
Trace Configuration Options
This section describes how to use Cisco Unified Communications Manager trace files for
troubleshooting.
Trace Configuration Options
Two ways to configure traces:
mvwmwsm
Configured per server and service in
Cisco Unified Serviceability > Trace >
Configuration
Turns on all possible trace options for
the selected servers and services
Less information to be processed
during trace analysis, more to the
point; higher risk of missing relevant
information
Configured per server and service in
Cisco Unified Serviceabiiay >Trace >
Troubleshoottng Trace Settings
Allows the configuration of trace
filters for the selected servers and
services
More data to be processed during
trace analysis; includes information
not required for troubleshooting a
specific issue; less risk of missing
relevant information
Less impact to server performance Higher rnpact to server performance
Cisco Unified Communications Manager traces can be enabled in two ways:
Activate troubleshooting traces: You can activate so-called troubleshooting traces from
Cisco Unified Serviceability > Trace > Troubleshooting Trace Settings. This turns on
(almost) all possible trace options for the selected servers and services. Such
troubleshooting traces are extremely detailed and generate much trace information that
might not be useful for tracing a specific problem. You will have to search for relevant
informationwithin the enormous amount of data that is generated by troubleshooting
traces. In addition, troubleshooting traces can impact server performance.
Enable and configure traces: This option is available from Cisco Unified Serviceability >
Trace > Configuration and allows you to enable and disable traces per server and service
(just like troubleshooting traces). However, in addition, you can apply trace filters by
enabling and disabling so-called trace fields or applying device-based traces. Such traces
generate less infonnation and, therefore if configured properlyare more to the point by
including less irrelevant information. Consequently, trace analysis is more efficient, and
impact to serverperformance canbe drastically reduced, compared to troubleshooting
traces.
Trace files can either be collected fromthe Cisco Unified Communications Manager servers to
the PCof theadministrator and then viewed locally, or they can be remotelybrowsed from the
PCof the administrator while being located at the server. For both options, the Cisco Unified
R1MT is required. Cisco Unified RTMTis an application plug-in that can be downloaded from
Cisco Unified Communications Manager Administrationand then installed to the PC of the
administrator.
Trace Configuration
This section describes how to configure basic trace.
M^^LIL
rsurcttto
Select the server and
the service to trace
Debug trace level
settings Influence how
verbose the trace
output will be.
Activate trace for
specific components,
instead of enabling all
trace for the service.
Trace > Configuration
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b..,!-*.'.. . Hill ~h l.,11, UCK-.-.^M^.. - . | .
s. ,uer Hi-:.-*'-oj., ,..,.!.!. f.L~,:to r>..,f,.,.
F .....I .. .,, F,*i,i>sv-.
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. J..i.1.j i~nui
i*)..H. '...^ *.,r. TrM-i. (., 1
, ,<,
...If.. f,.,tl.[.
When choosing Cisco Unified Serviceability > Trace > Configuration, the first parameter to
supply is the server that is going to perform tracing. Trace contiguration can also be configured
for all servers at the same time. Select a service group and an individual service to trace.
Then, select the debug trace level to determine how verbose the trace output will be.
Finally, activate the trace for specific components.
1-44 Troubleshooting Cisco UnifiedCommunications (TVOICE)v8.0
Customizing Trace
Csco Unified Communications Manager lets you customize the tracing to include the
components that are needed for troubleshooting with the minimum size of trace log files
Customizing Trace
You can customize traces as follows:
Change digit analysis complexity to include translations and
alternate pattern analysis (default is standard complexity).
Configure the maximum number of devices that are
concurrently traced(default is 12).
Generate an alarm when the specified search strinq exists
m a monitored trace file.
Configure trace for specific servers only.
Most troubleshooting is usually performed on Cisco CallManager service targeting dial plan
and^devicc issues. When troubleshooting digit analysis in Cisco" Unitied Communfc ns
delude h' rTT CT Umfied C0mmunica^ Manager traces by default do no
inc ude details about translation patterns and alternate matches- only the finally matched
pa tern shown. Th.s behavior does not depend on the configured trice level. Ilow you
car, choose to create more detailed information dunng digit analysis by changing the Cbco
CallManager service parameter Digit Analysis Complexity from its default value
StandardAnalysis (shown in trace output as dac="0") to
TranslationAndAlternatePatternAnalysis (shown as dac-"I").
By modifying enterprise parameter Max Number of Device Level Trace in Cisco Unified
Communications Manager Administration, you can specify how many devices can Id
concurrently ,fdevice name-based trace is chosen in Trace Configuration in Cisco
Communications Manager Serviceability. The default setting is 12 devices
If you want to generate an alarm when the specified search string exists in amonitored trace
file, enable the LogFtleSearchStnngFound alert in Cisco Unified RTMT. You can fmd he
cho0F;^:?>'jst1alrinthe LpmTctcata^ ^, unified s^-J^
choose Alarms >Delations. In the Fmd Alarms Where drop-down menu, choose the Svitem
Alarm Catafog. In the Equals drop-down menu, choose LpmTctCatalog
ivaZ cTsco I^rcHrC ^ SCttmg fr thC Sen'iCe fr WhJCh yU waTlt to co"* traces. If you
ha,e Cisco Unified Commun.cations Manager clusters, you can configure trace for the serlke
on one server or on all servers in the cluster. You can configure the seUing at he Tr ce
Configuration screen of Cisco Unified Serviceability.
2010 Cisco Systems. Inc.
1-45
Trace Output
1-46
This topic describes how to interpret basic trace output.
'LlH.Hi'
, ^*-
online: lOOQQODl) SoftK.yBvit sof CK.yEv.nt.2 (mwCall)
CLID. !StndMon.Clut.r;.*lID:! 10.1. 5.2
2/15/20;S 07:58,15.849 CCK fit
io.Intlt..l) e.llSlf.r.M.e.-O
. . tr-oii=tf -
02/15/20CS 07.5B.15 85! CCH Dlglt .ly.. -tchlpi-'a-. ^.-^Hl-,
MxDEV: SEP 500000000001 LIT,:, IUU Tl<mltion.
WIS*: 0800.
;,15/203S 03 =12 =41 4 J!> CCM
llt.iinri!n>C'lli^'"TB"b"-!ml callingP.rtyMuBb.1
1.lil^P.=tlClor-f.95S..-4[.4-91=.-0624-tH462.daf,l-ll
jl.linjl'it t.rn.S . 14 0BJXJKISJ
FunyQu-lAfldC'.llaP.ltyBaim..r.S14 1)8 555 444 4
0J/15ô"s%8!l2 =41.435 CCH SKDHSh.r.dD.= =flDdLolD.v. -
N.M-TEHO-40! r^.30U15.d..(7b-(OJ2.935.-blefl3**4 i.Actvi.-l r id. (1, 59, II
found eajD= =unii\lon.Clut.rMlUD:.lfi.l.S-2>
r^^.lwl.1-."!" ,g,IE.,St^dM.CluBt.rxHID. =10.1.5.2.
85553001
This figure shows an example oftypical trace output.
Trace output can be very long, and this figure shows only some of the important entries the-
rest have been excluded. The trace output that is shown depends on which service is traced,
what the level of tracing is. and which events you try to identity by using the trace.
This figure starts with an IP phone off-hook event that shows that the softkey NewCall was
pressed at the phone. Then the trace output continues with the start ot digit analysis. Here yoi
ensee that the fully qualified calling number is 4085552001 and the short number (directory
Ito is 2M1. This digit analysts also shows acalling search space (CSS) configured at t e
IP phone. The CSS is shown as alist of partitions that the IP phone can reach in the dial plan
(pss - partition search space).
The rest of digit analysis has been excluded for this trace output, but it will be discussed in
detail later in the course.
You can see that Cisco Unified Communications Manager has matched the route pattern. For
the called number 914085554444, the best match has been found in the route pattern
9 1408XXXXXXX.
The bottom of the trace output shows device selection. After the call-routing decision.ismad.
(best match selected) an appropriate device for path selection is determined, the route list that
^called TFIIO-4a8.,n this particular case. The trace output would continue with the selection
ofroute groups and sending signaling messages to voice gateway.
Each of the lessons that follow ,n this course introduces you to aspecific trace output that can
helpto isolate problems.
Troubleshooting Cisco Unified Communications (TVOICE] v8.0
Communication Between Cisco Unified Communications
Manager and SCCP Cisco IP Phones
UnifiM r briefly e*Plai?,S thC tFaCe UtpUl th3t identifieS ,he <ieation between Cisco
[p phonesmmUri,Ca,,0nS ger 3nd Skmny C,iCnt Cntr01 Pr0tC01 (SCCP Clsco Unified
Communication Between Cisco ......._
Communications Manager and SCCPCisco
SSHlg^bnlnrt (meBsa9es from lp Phcf,e to Cisco Unified Communications
SottKeyEvent
K9ypadButton
Stimulus (phone buttons)
CCM|Stat.onD(messages from Cisco Unified Communicalions Manager to IP ptone)
Phone Display '
SetLamp.CallSlale.SelectSoftKeys
Phone Tones
SetRinger, StartTone, StopTone
Call Information
DialedNumber
Calllnfo (includes callingParty. alternateCallngParty, etc )
DisplayNotify (for example Not Enough Bandwidth)
RTP Flows
' CSS:?StopMediaTransmission, OpsnReceiveChannd,
CCM Stationlnit messages are sent from aCisco Unified IP phone to Cisco Unified
Communications Manager. The events that trigger such amessage include the following:
" tt?a?e fiir* ^^3USCr PrESSCS 3Sftkey bUttn' 3SoftKc*E will be shown in
" ^ZÎ^^3" 3* ta* ^ypad, ^Button
Phone button pressed: This event causes aStationlnit message with the string Stimulus in
Unif!e^Pf ^nrT"* ^ ^ CiSC Unified Cnications Manager to aCisco
tt;-^dlffercnt stationD mfor di^*. * --
Phone display-related messages: SetLamp, CaMStatc, or SelcctSoftkeys messages are
used to turn on or off phone button lights, to update the call state (ringing S and
so on), or to change the softkeys according to the current call state.
Phone tone-related messages: SetRinger tells the phone to ring. StartTone and StopTonc
Introduction to Troubleshooting Cisco Unified Communicalions Solutions
1-47
Call information messages: The DialedNumber that specifies the number that should be
shown at the display. The number can be updated while processing an outgoing call
depending on where the digit manipulation is configured (route pattern versus route list
versus global transformations). The Calllnfo is used to signal the callingParty (as shown at
,he phone display) and the altemateCallingParty (used for callbacks) to the IP phone
DisplavNotify messages include notification text that should be displayed at the IP phone
(for example Not Lnough Bandwidth when acall is not admitted by Call Admission
Control [CAC]).
Real-Time Transport Protocol (RTP)-related messages: SiartMedial ransmission,
StopMediafransimssion. OpenReceiveChannel, and CloscRceeiveChannel messages are
usecl to start or stop sending or receiving RTP packets and include the peer IP address and
the RTP port numbers to use.
2010 Cisco Systems Inc
Call Setup Phase
This section briefly explains the trace output that identifies the events during the call setup
phase.
CCMjDigit analysis (during analysis):
pss and ToDFilteredPss, en, fqcn, dd, dac
CCM]| {after matching a route pattern):
PretransformCalingPartyNumber, Call ngPartyNumber,
FullyQualHiedCalledPartyNumber. DialingPattem.
Pret ransform DigitSt ring
CCM|RouteUst:
CCM| RouteList: RouteListName shows route list by name.
CCM|RouteList: RouteGroup shows number of route groups in
route list by count number.
CCM]RouteListCdre: String skipToNextMember indicates that next
route group is examined.
CCM|RouteListCdrc: String executeRouleActon indicates current
device is attempted.
CCM|SM DMSharedData shows device name after findLocaDevice:
Shows device or route list referenced from route pattern
Shows device while processing route list
Each call starts with a digit analysis section. During digit-by-digit analysis, the appropriate
trace output includes the following important information, which is found in CCM|Digit
analysis trace records:
pss: The list of partitions that is available to the calling device. The pss shows partitions in
prioritized order, which is especially important when using the line or device CSS
implementation model.
ToDFilteredPss (time-of-day filtered pss): The same list but reduced by those partitions
that are currently inactive because a time schedule is applied to them that does not include
the local time. This is the actual list of partitions that is used for looking up the dialed
number in the call-routing table.
en (calling number): This is the caller ID; in the case of an IP phone, it is the directory
number.
fqcn (fully qualified calling number): This is the calling number of an IP phone as it
looks after applying the external phone number mask.
dac (digit analysis complexity): Indicates whether translation pattern and alternate pattern
analysis is active.
After the best match is found, detailed informationabout the match is displayed. The
information includes the following fields that are found in CCM|| trace records:
PretransformCallingPartjNumber: This is the caller ID before any digit manipulation is
applied.
CallingPartyNumber: See en.
FullyQualifiedCalledPartyNumber: See fqcn.
2010 Cisco Syslems. Inc.
Introduction to Troubleshooting Cisco Unified Communi cat ions Solutions
DiatingPattern: The pattern that is matched.
PretransformDigitString: The called number before any digit manipulation is applied.
After the call-routing decision, path selection is performed. The following information
regarding path selection can be found in Cisco Unified Communications Manager trace tiles:
CCM RouteList records can include the RouteListName, which shows the applicable route
list by its name. Under RouteGroup, you can find the number of route groups in the route
list bv a count number. After the string RouteListCdrc, you can find the string
skipToNextMember (which indicates that the next route group is examined) or the string
executeRouteAction (which indicates that the current device is attempted).
CCM SMDMSharedData trace records can include the string findLoealDevice. If present, it
refers to a route list or device (if shown after matching a route pattern) or the device (if
shown while processing a route lis!).
TroubleshootingCisco Unified Communications (TVOICE) v8.0 2010 Cisco Systems, Inc
Call Setup Phase (Cont
' CCM|Locations:
CCM|RegionsServer:
" SnSm^'1"6,5 Sh,WS maXimUm permil,ed bandwidth (kp/s)
consumed by codec (ignoring packetization overhead).
CCM| MediaResourceManager:
JrTquileT^^
CapCount.Region shows region for both endpoints.
mrgl shows MRGL to be used for media resource allocation
CCM|MediaTerminationPointControl:
oZltZTofmpTri DheviceriName indlcale* successful allocation
ui iranscoaeror MTP and shows device byname.
.ha, are rnidcol, (for exampie conferences or Lie I^2^ MO , t" '^"'^
Theselec.ed device is snn by its name in CCMlMediaTcrmia,ionP,,Co,ro, trace
2010 CiscoSystems, Inc.
introduction to Troubleshoo.i.g Cisco Unrfied Communications Solutions
1-51
Cisco Unified Communications Manager Dialed
Number Analyzer
1-52
This topic describes how to use the Cisco Unified Communications Manager Dialed Number
Analyzer tool to troubleshoot dial plans in the Cisct
configuration.
,coUnifiedCommunications Manager
o Unified Communication
Csco Unified Communications Manager Dialed Number Analyzer installs as a eatuu^n ice
ll ,w u hCisco Unified Communications Manager. The tool allows you to test aC.sco
uSCreations Manager dial plan configuration before deploying it. You can aiso
use the tool to analyze dial plans after the dial plan is deployed.
"-:- S :::r:;"^-:SS-=-"-
srssrossKi-.s.s;=:-::s:u
called-party transformations that are applied to the dialed digits.
Troubleshooting CiscoUnified Commun
cations (TVOICE! v8.0
The following list describes specific features ofthe tool:
. Analysis: Cisco Unified Communications Manager Dialed Number Analyzer allows
selection of specific devices that act as, calling parties and called parties to test the dial plan.
It allows analysis of calls from devices such as IP phones, computer telephony integration
(CTf) ports and eateways. The tool allows you to perform asimple analysis by directly
entering dialed digits as input and choosing aCSS within which the analysis must be
performed If you choose adevice, the tool uses the CSS that is associated with the device
to perform the analysis. The tool also allows analysis of calling-party numbers that arc not
bound to any device.
. Digit Discard Instructions and Dialing Patterns: The Cisco Unified Communications
Manager database stores called-party transformation information such as discard digits
instructions (DDls) that are specified for Cisco Unified Communications Manager dial
plans. Because Cisco Unified Communications Manager Dialed Number Analyzer uses the
Cisco Unified Communications Manager database to analyze dial plans, the tool also
allows you to view DDIs that are specified for the dial plans. Cisco Unified
Communications Manager uses route patterns to route or block internal and external calls.
Cisco Unified Communications Manager Dialed Number Analyzer allows you toview
dialing patterns that are associated with devices that are configured in the Cisco Unified
Communications Manager dial plan that you arcanalyzing.
Analysis Output: Cisco Unified Communications Manager Dialed Number Analyzer
displays analysis results in anew browser window that you use to perform analysis. You
can either view the results online or save the output that displays in the form ofan XML
file for easy retrieval and use.
After Cisco Unified Communications Manager Dialed Number Analyzer installs as aservice, it
must be actuated. Use Cisco Unified Serviceability and activate the Cisco Unified
Communications Manager Dialed Number Analyzer service in Service Activation
To access Cisco Unified Communications Manager Dialed Number Analyzer, go to Cisco
Unified Serviceability and choose Tools >Dialed Number Analyzer. You can also use the
following URL: https://<cm-machine>/dna.
You can use the Cisco Unified Communications Manager Dialed Number Analyzer to perform
the analysis that is based onthe following options:
Analysis by using the analyzer window (shown in the figure) allows simple analysis that
involves entering calling-party and called-party digits in Cisco Unified Communications
Manager Dialed Number Analyzer and choosing aCSS for the analysis. Cisco Unified
Communications Manager Dialed Number Analyzer uses this CSS and analyzes the dialed
digits. You do not need tochoose specific devices or provide any other input. Cisco
Unified Communications Manager Dialed Number Analyzer allows analysis ofa route
pattern, translation pattern, phone directory number, or CTI route point.
Analysis by using phones when Cisco Unified Communications Manager Dialed Number
Analyzer provides aPhones window (shown in the figure) where you can find and list
phones by device name, description, directory number, CSS, device pool, device type, and
call pickup group. You can find aphone and choose itas acalling device for the analysis
that you want to perform. You can further choose aconfigured phone line (directory
number) and use it as a calling-party number.
)2010 Cisco Systems, Inc Introduction to Troubleshooting Cisco Unified Communications Solutions 1-53
Ana ysis by using gateways when Cisco Unified Communications Manaeer Dialed Number
Analyzer allows you to find and list gateways through which Cisco Unified
Commumcations Manager receives inbound calls. From the list of gateways you can
choose gateway endpoints to dial digits and analyze the cail flow of inbound calls to a
t lSco Unified Commumcations Manager system. You can choose gateway endpoints that
are configured in the Cisco Unified Communications Manager system.
Analysis by using trunks when Cisco Unified Communications Manager Dialed Number
Analyzer provides a1runks window where you can find and list trunks through which
inbound dialed digits can beanalyzed.
Analysis by using multiple analyzers allows you to perform multiple analyses and bulk
testing ot dial plans. Cisco Unified Communications Manager Dialed Number Analyzer
provides aMultiple Analyzer window where you can choose acomma-separated values
(C. S\ )tde that contains alist ofdata that is required for analysis. Cisco Unified
Communications Manager Dialed Number Analyzer will then process the CSV file and
display the bulk output results.
1-54 Troubleshooting Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems Inc
Cisco Unified Communications Manager Dialed Number
Analyzer Output
^8rTtTr.the eXamPle ^^f ClSC Umfkd Communiti Manager Dialed
Cisco Unified Communications fVfpnaqpi
Dialed Number Analyzer Output
tt - M.K^k^jcrj
f PattFfli Pattern, gflni
O OldKu
nlHi' Maim -
" *-iiref*n.itf AH-MiDnzah-ln Co**
o jiuihD-iranHi L*o*i -n
rrt1'****+*"t (aftlng Hr*y ftuntiirr
"J PrfTTintlci-n f illdg Pirty IWwi.brf
CallKiTBrtr iwmbr' - *?J*!|!iinoi
r**4 pt> TFMriBn-#f1o.
i*UBd ran* Mm* -
OllCtrd DEgbHJitTtrurptwi - I>|.bi
- t'ftf*ljl
Analyses of En-Bloc Dialing
This output was genera ed from the Phones analysis window, where from the BRI phone 1
with directory number 3001 (public switched telephone network [PSTN] number 5215553001)
simulated dialing of the number 9011445653200 was analyzed. 5553001),
The figure shows segments of asingle output window that starts with Results Summary and
en swtth Route L,siThe content of analysis output depends on the dial plan eon"gurat"n and
*hich analyzing window was used to generate that output.
This particular analysis shows that the directory number that was used for dialing belongs to
Summ " r,CTT^ tHC Phne iS CnfigUred With device * ne CSS The Results
Summa* shows abnef summary of the call that includes the match result, matched palm
information, and the end device that is used for call touting (route list US^PSTN_rl in thi'
2010 CiscoSystems, Inc.
Introduction to TroubleshoCing Cisco Unified Communications Solutions
1-55
Cisco Unified RTMT
Th topic describes ihe functions of the Cisco Unified RTMT and explains how it can be used
to collect facts for troubleshooting.
f\ I m
This tool provides real-time information about Cisco Unified Communications Manager
devices and performance counters and enables you to collect traces.
Performance counters can be system-specific or specific to Cisco Unified Com.mini cations
Manager. Objects comprise the logical groupings of like counters tor aspec tedeuce o
feature such sCisco Unified IP phones or Cisco Unified Communications Manager system
Xnanee. Counters measure various aspects of system performance punters measure
statistics such as the number of registered phones, calls that are attempted, and calls in
progress.
Cisco Unitied RTMT. which runs as aclient-side application on the PC ^^^
uses H1TPS and TCP to monitor device status, system performance, device d.sco*cr>, and CT1
apfications maCsco Unified Communications Manager cluster. The tool also connects
directly to devices via 1ITTPS to troubleshoot system problems.
You can install Cisco Unified RTMT. which works for resolutions 800*600 and abewe, on a
Xu that is runnm, on most of the Microsoft Windows plattorms or Linux with KDE or
Gnome client. Csco Unified RTMT requires at least 128 MB in memory to run on aWindow
operating svstem platform.
you It clco Unifed RTMT, ,, uses ,ta Dcfcul. profile and d,splayS .he sys.cm summary
page inthe monitor pane.
)2010 Cisco Systems
Cisco Unified Communications Manager supports the creation ofaCisco Unified RTMT user
with restricted access to Cisco Unified Communications Manager Administration You can
create auser with aprofile that is limited to Cisco Unified RTMT usage only. The user will
have full access to Cisco Unified RTMT but will not have permission to administer aCisco
Unified Communications Manager server.
The Cisco Unified RTMT window comprises the following main components:
Menu Bar, which includes some or all of the following menu options, depending on your
configuration:
File: Allows you to save, restore, and delete existing Cisco Unified RTMT profiles
monitor Java Heap Memory Usage, go to the Serviceability Report Archive window
in Cisco Unified Serviceability, log off. orexit Cisco Unified RTMT.
System: Allows you to monitor system summary, monitor server resources, work
with performance counters, work with alerts, collect traces, and view syslog
messages.
- Communications Manager: Allows you to view Cisco Unified Communications
Manager summary information on the server; monitor call-processing information;
and view and search for devices, monitor services, and CTI.
Unity Connection: Allows you toview the Port Monitor tool.
IME Service: Allows you to monitor server and network activity ofthe Cisco
Intercompany Media Engine (IME) server.
- Service Advertisement Framework and Call Control Discovery (CCD): Allow
you to monitor ifSAF forwarder has registered with Cisco Unified Communications
Manager orifCCD hosted patterns are being learned.
- Edit: Allows you to configure categories (for table format view), set the polling rate
for devices and performance monitoring counters, hide the quick launch channel
andedit thetracesetting forCiscoUnified RTMT.
Window: Allows you to close asingle Cisco Unified RTMT window or all Cisco
Unified RTMT windows.
- Application: Depending on your configuration, allows you to browse the applicable
web pages for Cisco Unified Communications Manager Administration, Cisco
Unified Serviceability, Cisco Unity Connection Administration, and Cisco Unity
Connection Serviceability.
Help: Allows you to access Cisco Unified RTMT documentation online help or to
view the Cisco UnifiedRTMTversion.
" HU? LaUnCh Channe,: You can click this pane with tabs on the left side of the Cisco
Unified RTMT window to display information about the server or information about the
appheations. The tab contains groups of icons that you can click to monitor various objects.
Monitor pane: Pane where monitoring results display.
The following pages explain individual functional components of Cisco Unified RTMT in more
u Grill 1.
2010CiscoSys,emS,lnc. Introduction to Troubleshoo.ing Cisco Unifed Communications Solutions
1-57
Alerts
1-58
This topic describes how to view. set. and modify both predefined and customized alerts in
Cisco Unified RTMT.
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Two types of alerts in Cisco Unified RTMT;
* Preeonfigured read only
* User-defined
Csco Unified RTMT. which supports alert defining, setting, and viewing, contains
preeonfigured and user-defined alerts. Although you can perform configuration tasks tor both
types, you cannot delete preeonfigured alerts (whereas you can add and delete user-detmed
alerts).'The Alert menu comprises the following menu options:
Alert Central: Comprises the status and history of every alert in the Cisco Unified
Communications Manager cluster.
Set Alert/Properties: Allows you tosetalerts and alert properties.
Remove Alert: Allows you to remove an alert.
Enable Alert: Allows you to enable an alert.
Disable Alert: Allows you to disable an alert.
Suspend Cluster/Node Alerts: Allows you to temporarily suspend alerts on aparticular
Cisco Unified Communications Manager node oron the entire cluster.
Clear Alert: Allows you to reset an alert to signal that an alert has been taken care of, such
as for example, changing the color of an alert item from red to black. After an alert has
been raised, it.color will automatically change to red in Cisco Unified RTMT and will stay-
that way until you manually clear the alert.
Clear All Alerts: Allows you to clear all alerts.
Alert Detail: Provides detailed infonnation on alert events.
Config Email Sener: Allows you to configure your email server to enable alerts.
Config Alert Action: Allows you to set actions to take for specific alerts; you can
configure the actions to send the alerts to desired email recipients.
In CiscoUnified RTMT, you canconfigure alert actions for everyalert that is generated. The
alert actions that are sent to email recipients that you specify can be placed in the alert action
list.
Cisco Unified RTMThas a set of preeonfigured alerts that cannot be deleted. However, you can
change their properties, and you can disable them.
) 2010 Cisco Systems, Inc Introduction lo Troublespooling Cisco Unified Communications Solutions 1-59
Custom Alerts on Performance Counters
This figure shows an example of how to set a custom alert on a Performance Monitor counter.
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Custom Alert--
Setting a custom alert on a performance counter:
Select the counter and nghl-clck on the seiecied counter
Enable the alert, set the severiiy level, and optionally add a custom description.
Sei the desired threshold values and v*en the alert should be triggered.
Set limitson the frequency and time that the alert can be sent
Changing the propeities of a predefined alert is very similar, follow ihese steps to set a custom
alert:
Note The properties pages will be unique to the counter or predefined alert.
Step 1 Select the performance monitoring counter, ensure that it is highlighted, and then
right-click the counter and select Set Alert/Properties.
Step 2 Check the Enable Alert check box, select the severiiy level, enter an optional
description, and click Next.
Step 3 Set the desired threshold values that will trigger the alert, and choose whether to
trigger the alert immediately. Click Next.
Step 4 Set any desired limits on alert generation and, optionally, set the hours that the aiert
can be generated. Click Next.
Step 5 Add any desired email addresses for the alert to go to and click Activate.
Syslog Viewer
This topic describes how to view the local syslog files in the Cisco Unified RTMT.
Cisco Unified RTMT Syslog Viewer
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You can use the Syslog Viewer to view messages from these logs:
System Logs: Hardware, operating system, and systems messages are logged here.
Application Logs: Cisco UnifiedCommunications Manager alarms and alerts as well as
application messages are sent here.
Security Logs: Logins, login attempts, and security-related messages are sent here.
To display messages in the Syslog Viewer, perform these steps:
Step 1 Perform one of the following tasks:
Inthe QuickLaunch Channel, clickthe Tools tab; thenclickSysLog Viewer tab
and the SysLog Viewer icon.
Choose Tools >SysLog Viewer >Open SysLog Viewer.
Step 2 From the Select a Node list box, choose the server where the logs are stored that you
want to view.
Step 3 Click the tab for the logs that you want to view.
Step 4 After the log displays, double-click the log icon to list the filenames in the same
window.
Step 5 To view the contents of the file at the bottom of the window, click the filename.
Step 6 Click the entry that you want to view.
Step 7 To view the complete syslog message, double-click the syslog message.
Trace & Log Central
This topic describes how to collect trace and log files and view them in ihe Cisco Unified
RTMT.
Use Trace & Log Central, an option within the Cisco Unified RTMT, to collect, view, and zip
various Cisco Unified Communications Manager Service traces and other Cisco Unitied
Communications Manager log files. With the Trace & Log Central option, you can collect
Cisco Unified Communications Manager SDL and SDI traces, Cisco Unified Communications
Manager application logs (such as Cisco Unified Communications Manager Bulk
Administration Tool [BAT] logs), system logs (such as the livcnt View application, security,
and system logs), and crash dump files.
The Trace & Log Central option provides these methods to collect and view trace tiles:
Remote Browse: After the system has generated trace files, you can view them on the
server by using the viewers within the Cisco Unified RTMT. You can also use the Remote
Browse feature to download the traces to your PC.
Collect Files: Collects and downloads iraces for services, applications, and system logs on
one or more servers in the cluster for an absolute date and time range (such as between July
8, 2010 at 12:30 and August 8, 2010 at 12:30) or for a relative time (such as within the last
30 minutes).
Query Wizard: Collects trace files for services, applications, and system logs for an
absolute or relative time range, which contains text strings that you specify. You can view
the collected trace files or download the trace files to your PC. You can also save the trace
collection query criteria for later use. If you save the query as a regular query, you can run
the query only on the node upon which it was created. If you save the query as a generic
query, you can run it on any node in any cluster.
1-62 Troubleshooting Cisco Unified Communications (TVOICE) v8 0
Schedule Collection: Schedules a recurring trace collection and allows users to perform a
specified action, including running another query, generating a syslog, or downloading the
trace files on a Secure FTP (SFTP) server.
Local Browse: After you have collected trace files and downloaded them to your PC, you
can view them with a text editor that can manage UNIX variant line terminators such as
WordPad on your PC, or you can view them by using the viewers within the Cisco Unified
RTMT.
Collect Crash Dump: Collects a crash dump file for one or more servers on your network.
Real Time Trace: Comprises two options: View Real-Time Data and Monitor User
Events. The View Real-Time Data option allows you to view the current trace file that is
being written on the server for an application. The Monitor User Events option enables the
system to monitor real-time trace files and perform a specified action when a search string
displays in the trace file. Actions include generating an alert, generating local or remote
syslogs, or downloading trace files via SFTP.
Job Status: Allows you to view the status of the trace collection jobs that are running on
the system as well as recently processed jobs.
After the system has generated trace files, you can view them on the server by using the
Remote Browse option.
You can collect individual trace files or zip multiple traces into a single file. You can manually
delete the collected trace files from the server, or you can set the Trace & Log Central option to
delete the trace files from the server after collection.
After you collect the files, you can view them in the Local Browse option. The file displays in
the appropriate viewer such as the Quality Report Tool (QRT) Viewer, Q.931 Translator, Log
Viewer, or Generic Viewer.
2010 Cisco Systems, Inc. Introduction to Troubleshooting Cisco Unified Communications Solutions 1-63
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After the system has generated trace files, you canviewthemon the server by usingthe
viewers u ithin the Cisco Unified RTMT, You can also use the Remote Browse feature to
download the traces to your PC.
Performthe following procedure to display or download the log files on the server with the
Trace & Log Central feature:
Step 1 Display the Trace & log Central options.
Step 2 Double-click Remote Browse.
Step 3 Choose the appropriate service for which the trace output has been generated and
click Next. In the next window, click Finish.
Step 4 After the traces become available, a message displays.
Step 5 To display the results, navigate to the file through the tree hierarchy. After the log
filename displays in the pane on the right side of the window, you can either right-
click to select the type of program that you would like to use to view the file or
double-click the file to display the file in the default viewer.
To download the trace files, choose the tiles that you want to download, click Download,
specify the criteria for the download, and click Finish. After you download the trace files, you
can view them by using the Local Browse option of the Trace & Log Central feature.
1o delete trace files from the server, click ihe file that displays in the pane on the right side of
the window and then click Delete.
To refresh a specific service or a specific server in a cluster, click the service or server name
and then click Refresh. After a message displays that the remote browse is ready, click Close.
To refresh all services or all servers in a cluster that displays in the tree hierarchy, click
Refresh All. After a message slates that ihe remote browse is ready, click Close.
If trace and log files were collected to the local machine thai is running the Cisco Unified
RTMT, you can view, search, and filter them by using the Voice Log Translator (VLT) tool.
You can download this tool from Imp: www.cisco, coin'egi-bin'iablebu ild.pl' \oice~tool.
1-64 Troubleshooting Cisco Unified Communications (TVOICE) v8.0 2010 Cisco Systems, Inc.
*
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Performance Monitor and Data Logging
This topic describes how to enable and use Performance Monitor data and monitoring to assist
you and Cisco TAC with troubleshooting.
Cisco Unified RTMT Performas
Monitor and Data Logging
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You can monitor the performance of Cisco Unified Communications Manager by choosing the
counters for any object by using Cisco Unified RTMT. The counters for each object display
when the folder expands.
You can log performance counters (known as perfmon) locally on the computer and use the
performance log viewer in Cisco Unified RTMT to display the perfmon CSV log files that you
collected, the Alert Manager and Collector (AMC) perfmon logs, and Cisco RIS data collector
perfmon logs.
You can enable the troubleshooting perfmon data logging feature to collect statistics
automatically from a set of perfmon counters that will provide comprehensive information on
the system state. Be aware that enabling troubleshooting perfmon data logging can impact
system performance on the server.
Cisco Unified RTMT can display perfmon counters in chart or table format. The chart format
that is shown in the figure displays the perfmon counter information by using line charts. For
each category tab that you create, you can display up to six charts in the Cisco Unified RTMT
Perfmon Monitoring pane.
2010 Cisco Systems, Inc Inlroduction to Troubleshooting Cisco Unified Communications Solutions 1-65
Performance Data Logging
Ihe troubleshooting perfmon data logging feature helps Cisco TAC to identify system
problems.
* Enables collection of performance monitoring statistics.
' Use as directed by TAC
* Can impact cluster performance.
System > Service Parameters > Cisco RIS Data Collector
When you enable troubleshooting perfmon daia logging, you initiate the collection of a set of
Cisco Unified Communications Manager and operating system performance statistics on the
selected node. The statistics that the system collects include comprehensive infonnation that
you can use for system diagnostics and information from a set of counters that is not a part of
the current set of preeonfigured counters in the Cisco Unified RTMT.
An extensive amount of infonnation is collected in a short time; therefore, you should not
enable troubleshooting of perfmon data logging for any extended time. In addition, you should
enable the Log Partitioning Monitor to monitor disk usage while the troubleshooting perfmon
data logging is enabled.
These available settings enable and disable troubleshooting perfmon data logging:
Enable Logging: From the drop-down menu, choose True to enable or False to disable
troubleshooting perfmon data logging. The default value specifies True.
Polling Rate: Enter the polling rate interval (in seconds). You can enter a value from 5
(minimum) to 300 (maximum). The default value specifies 35.
Troubleshooting Cisco Unified Communications (TVOICEl v8 0 2010 Cisco Systems, Inc
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Maximum No. of Files: Enter the maximum number of Troubleshooting Perfmon Data
Logging files that you want to storeon disk. Youcan enter a valuefrom1 (minimum) up to
100(maximum). The default value specifies 50. Consider your storage capacity when
configuring the Maximum No. of Files1 and Maximum File Size parameters. Cisco
recommends that younot exceed a value of 100MBwhen youmultiply theMaximum
Number of Files value bythe Maximum FileSizevalue. When thenumber of files exceeds
the maximum numberof files that you specifiedin this field, the system will deletelog files
with the oldest time stamp.
Maximum File Size: Enter the maximum file size (in megabytes) that you want to store in
a perfmon logfile before a newfileis started. Youcanentera value from 1(minimum) to
500 (maximum). The default value specifies 5 MB. Consider your storage capacity when
configuring the Maximum No. of Files andMaximum FileSizeparameters. Cisco
recommends that you not exceeda valueof 100MBwhenyou multiplythe Maximum
Number of Files value by the MaximumFile Size value.
2010 Cisco Systems, Inc Introduction to Troubleshooting Cisco Unified Communications Solutions 1-67
Cisco Unified Reporting
This topic explains how to use theCisco Unified Reporting tool of Cisco Unified
Communications Manager to aid in troubleshooting.
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The Cisco Unitied Reporting web application, which is accessed at the Cisco Unified
Communications Manager console, generates reports for troubleshooting or inspecting cluster
data.
1hisconvenient tool provides a snapshot of cluster data without requiring multiple steps 10 find
thedata. Thetool design facilitates gathering datafrom existing sources, comparing thedata.
and reporting irregularities.
Areport combines data from one or moresourceson one or moreservers intoone output view.
for example, you can view a report that shows ihe hosts file for all servers in the cluster.
The application gathers information from the publisher server and each subscriber server. A
report provides data for all activecluster nodes that are accessible at the time that the report is
generated.
Cisco Unified Reporting includes the following capabilities:
A user interface for generating, archiving, and downloading reports
A notification message if a report will take excessive time to generate or consume
excessive CPU
For a complete description of reports that are available on your system and the data that is
captured in a report, access the Report Descriptions report.
Troubleshooting Cisco Unified Communications (TVOICE] v8.0 2010 Cisco Systems, Inc
Here is how you can access the application:
Choose Cisco Unified Reporting in the Navigation menu in Cisco Unified
Communications Manager Administration.
Choose File > Cisco Unified Reporting at the Cisco Unified RTMT menu.
Enter https:. '<servernameor IPaddress>:8443/cucrcports/ and thenenter your authorized
username and password.
2010 Cisco Systems, Inc. introduction to Troubleshooting Cisco Unified Communications Solutions 1-69
Cisco Unified Communications Manager CLI
This topic provides an overview of how to use the CLI of Cisco Unified Communications
Manager vo\()to aid in the troubleshooting process.
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For troubleshooting, show and utils command
categories are usually used.
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You can use the CLI to access the Cisco Unified Communications Manager systemfor basic
maintenance and failure recovery. To obtain access to the system, either use a hardwired
terminal (a system monitor and keyboard) or open a Secure Shell (SSH) session. Ihe account
nameand password that youuse to log inare createdat installation time. Youcanchangethe
password after installation, but you can never change the account name.
A command represents a text instructionthat causes the systemto performsome function.
Commands can be standalone, or they can have mandatory or optional arguments.
To get detailed help, at the CI.I prompt, enter the help command where the command specifies
ihe command nameor thecommand and the parameter. To queryonlycommand syntax, at the
CLI prompt, enter command?. For example, to get help with thecommand file list active log,
enter file list activetog?.
For complete informationon the Cisco Unified Communications Manager CLI commands, sec
the Command Line Interface Reference Guide for Cisco Unified Communications Solutions at
hup: www.ciiCO.com en 1!S docs voice ip commaicmcli ref'.S 0 I cli ref.siH.hli.il.
1-70 Troubleshooting Cisco Unified Communications [TVOICE) v8 0 2010 Cisco Systems, Inc
Cisco IOS Troubleshooting Tools
This topic provides an overview of howto use the CiscoIOS Software to support in
troubleshooting.
Cisco IOS show and debug Comi
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The figure shows an example of Cisco IOS show and debug commands. Major commands that
are used for troubleshooting of individual functionality will be covered in respective individual
lessons.
Cisco IOS Software contains several useful show commands that can aid in problem isolation.
The figure shows an example of how to display information for voice dial peers by using the
show dial-peer voice <tag>summary command. The command that is used depends on what
subsystem of a device that is running Cisco IOS Software that you want to inspect. Various
show commands will be presented in individual lessons of this course.
The output from privileged EXEC debug commands provides diagnostic information about
various internetworking events that relate to protocol status and network activity in general.
Set up your terminal emulator software (such as IlyperTcrminal) so that it can capture the
debug output to a file.
Before running any Cisco IOS voice gateway debugs, make sure that service timestamps
debug datetime msec is globally configured on the gateway.
Note Avoid collecting debugs in a live environment during operation hours.
Preferably, collect debugs during nonworking hours. If you must collect debugs in a live
environment, configure no logging console and logging buffered. To collect the debugs, use
show log.
Because some debugs can be very long, collect themdirectly on the console port (default
loggingconsole) or on the buffer (logging buffer). Collecting debugs over a Telnet session can
impact the device performance, and the result couldbe incomplete debugs, whichrequires that
you recollect them.
To stopa debug, use the no debug all or undebug all commands. Verify that thedebugs have
been turned off by using the show debug command.
1-72 Troubleshooting Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems. Inc
Generic Call Filter Module
The generic call filter module (GCFM) manages the filtering for call control application
programming interface CCAPI modules. These CCAPI modules arc dial peers, H.323, ISDN,
interactive voice response (IVR), MGCP, and Session Initiation Protocol (SIP).
Generic Call Filter Module
The GCFM providesa centralizedfiltering system for voice call
processing debug sand traces.
Call control components register withthe GCFM.
Match lists define the match criteria.
Debug commandsproduceoutputbased on upto 16 matchlists.
Each match list can be individually activated and deactivated and
all activated match list conditions must be met for output to be
produced.
Enabling some debug commandsduring normal business hourscould
senously impact call processing.
The filtering conditions are configured inthe GCFM, and then the individual modules are
informed whena call has to be filtered. The GCFMcoordinates between multiple modulesto
manage filtering conditions.
Toenterthe call filter match list configuration mode andcreate a call filter match list for
debugging voice calls, use the call (liter match-list number voice command, number isa
numeric label that uniquely identifies the match list, where the range is 1to 16. The following
example shows the voice call debug filter that isset to match outgoing called number 8288807:
call filter match-list 1 voice
outgoing called-number 8288807
To configure debug filtering for incoming called numbers, use the incoming called-number
[+]string[T] command incall filter match list configuration mode, for example.
To configure debug filtering for outgoing called numbers, use the outgoing command in call
filter match list configuration mode.
To configure debug filtering for the source carrier ID ortrunk-group-label, use the source
command in call filter match list configuration mode.
To filter debugging output for certain debug commands that isbased onspecified conditions,
use the debug condition command.
Introduction to Troubleshooting Cisco Unified Communications Solutions 1-73
Sniffer Traces
This topic describes how to use ageneral network sniffer to troubleshoot protocol issues
Requires third-party software.
Capture packets
H.323
SIP
SCCP
MGCP
RTP, SRTP
- Requires the use of
SPAN port.
1 Capturing encrypted data
requires additional steps.
13 & - * 1*1 rr
As data streams travel back and forth over the network, a sniffer captures packets, decodes, and
analyzes the contents. You can use asniffer in aCisco Unified Communications system to
apture packets that are used for call control. The commonly captured protocols include the
following:
H.323
SIP
SCCP
MGCP
In addition tothe call control protocols, you can also capture the actual media streams in the
form ofReal-Time Transport Protocol (RTP) orSecure RTP (SRTP) packets. Many packet
snitters can then recreate the conversation inthe form ofa waveform (WAV) orMPEG file.
Cisco Unified IP phones can be configured to replicate all signaling traffic at their PC port
(Port SPAN). Ifa sniffer isconnected to a Cisco Unified IP phone PC port, you can collect
stutter trace fromall conversations between the phone and Cisco UnifiedCommunications
Manager, for instance.
Cisco Catalyst switches also allow Switched Port Analyzer (SPAN) port configuration.
TroubleshootingCisco Unified Communications (TVOICEl v8 0
M
Summary
-ummary
Troubleshooting tools include Cisco Unified Communications
Manager tools, Cisco IOStools, and tools like packet sniffer.
Cisco Unified Serviceability is a web-based troubleshooting tool
that provides alarm and trace configuration, Cisco Unified
Communications Manager service management, and SNMP
agent.
Cisco Unified Serviceability Control Center stops, starts, restarts,
and views the status of services.
Administrators can use alarms to receive notification that an event
has taken place.
Cisco Unified Serviceability provides trace tools that assist in
troubleshooting: SDLand SDI traces can be enabled.
Summary (Cont.)
i 2010 Cisco Syslems, Inc
Amount of trace output depends on which service is traced,
what the level of tracing is, and the number of events included
in the trace.
Cisco Unified Communications Manager Dialed Number
Analyzer lets you testa Cisco Unified Communications
Manager dial plan prior to deploying it or after the dial plan is
deployed.
Cisco Unified RTMT provides real-time information about
Cisco Unified Communications Manager devices and
performance counters and enables you to collect traces.
Cisco Unified RTMT, which supports alert defining, setting, and
viewing, contains preeonfigured and user-defined alerts.
You can use the syslog viewer to view messages from system
logs, application logs, and security logs.
Introduction to Troubleshooting Cisco Unified Communi cations Solutions 1-75
nummary
Use Trace & Log Central to collect, view, and zip various
Cisco Unified Communications Manager service traces and
other log files.
You can monitor the performance of Cisco Unified
Communications Manager by choosing the counters for any
object by using Cisco Unified RTMT.
Cisco Unified Reporting web application can generate
various reports for troubleshooting or inspecting cluster data.
Use the CLI to access the Cisco Unified Communications
Manager system for basic maintenance and failure recovery.
The output from the Cisco IOS show and debug commands
provides diagnostic information about a variety of
internetworking events.
You can use a sniffer in a Cisco Unified Communications
system to capture packets that are used for call control.
In this lesson, you have learned to identify' the troubleshooting and monitoring tools that can be
used in a Cisco I'nified Communications solution.
References
For additional information, refer to ihese resources:
Command Line Interface Reference Guide for Cisco Unified Communications Solutions at
Imp: www.uisco.coin en I !S does voice ip comm cucrn.'cli ref'-S ()_! cli ref ISO I.html.
'-76 Troubleshooting Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems. Inc
Module Summary
This topic summarizes the key pointsthat werediscussedin this module.
e hummai
The four common areas for troubleshooting Cisco Unified
Communications systems are Cisco Unified Communications
Manager, Cisco Unified Communications applications, the
network infrastructure, and voice clients.
Totroubleshoot a Cisco Unified Communications system, you
must use a systematic troubleshooting method. This includes
defining a problem, gathering symptoms, creating and
implementing an action plan, and observing the results of
your actions.
Many tools exist to aid in the troubleshooting process of
Cisco Unified Communications systems. Some of these tools
are Cisco Unified RTMT, Cisco Unified Serviceability, Cisco
IOS debug and show commands, and CLI.
Inthis module, you have learned todescribe a systematic methodology totroubleshoot Cisco
Unified Communications solutions.
References
For additional information, refer to these resources:
Command Line Interface Reference Guide for Cisco Unified Communications Solutions at
http: wuw.cisco.com/en US docs.'voice ip eomm/cuem/cli ref'8 0 1/cli ref801.html.
) 2010 Cisco Systems. Inc
1-78 Troubleshooting Cisco Unified Communications (TVOICE) 8.0 2010Cisco Systems. Inc
Module Self-Check
Use the questions here to review what you learned in this module. The correct answers and
solutions are found in the Module Self-Check Answer Key.
There are four common elements of the Cisco Unified Communications system for
troubleshooting. Which two of the following do not belong to the Cisco Unified
Communications system" (Choose two.) (Source: Identifying Cisco Untfied
Communications Deployments)
A) Cisco Unified Communications Manager
C) DscÛnity Connection, Cisco Unified Presence or other Cisco Unified
Communicationsapplications
D) network infrastructure
E) file or application server
F) voice clients
02) When you are troubleshooting common VoIP issues such as one-way voice, no dial
tone or reorder tones, Cisco Unified Communications Manager is typically the first
place to look for problems. (Source: Identifying Cisco Unified Communications
Deployments)
A) true
B) false
Cm In some instances, voice clients can cause voice network malfunctions. Voice clients
include, but are not limited to, which three of the following? (Choose three.) (Source:
Identifying Cisco Unified Communications Deployments)
A) CiscoMobile Client .
B Cisco Unified IP phones, including Cisco IP Communicator, Csco Unified
Video Advantage, and Cisco Unified Presence Client
C) Cisco H.323 Communicator
D) Cisco Unified Communications Manager Attendant Console, Cisco
WebAttendant, or both
E) CiscoVoiceGateway
F) analog devices such as fax machines or legacy telephones
G) digital facilities as an ISDN PRI
QD
1-79
041 C^ZTfo"0W'"g K* *> (Che* .wo, (Sourcc: u,^
t- tsco Unified Commumcations Deployments)
A) Because Csco Unity Connection is integrated with Cisco Unitied
Commumcations Manager, much troubleshooting ,s necessary because of
configurate errors between the two devices '
B) Because Csco Unity Connection is integrated with Cisco Unified
Commumcations Manager, much troubleshooting is not necessary because
configuration errors are practically not possible
Troubleshooting Cisco Unity Connection problems always leads to
troubleshooting Cisco Unified Communications Manager problems
OseUo Jni?H r CiSC Unlty C0nnCCti0n Problcms Can iead t0 troubleshooting
l. isco Unitied Commumcations Manager problcms
Troubleshooting Cisco Unity Connection problems is always aseparate
process and ,tnever involves the troubleshooting ofCisco Unified
Before considering systematic troubleshooting methods, which three of the following
questions must you answer regarding your preparedness to manage aVoIP network
outage. (Choose three,, (Source: Using Troubleshooting Methodology)
A) Do you know all ofthe points ofcontact to end users''
B) Do you have an accurate physical and logical map of your internetwork
showing the physical location of all VoIP devices on the network and how they
are connected and alogical map of the network addresses, network numbers
and subnetworks.'
C) Do know which application protocols are being used within the network to
provide mission-critical services?
D) Do you know which protocols arc being routed and the correct, up-to-date
contiguration information for each protocol?
E) Can you separate voice and data traffic in your network during the periods of
network issues?
F) Mas your organization documented the behavior of file servers during the
network outage? 6
G' !I,Ta?vrri2ati0n d0CUmcnted nomal "**<>* behavior and performance
so tliat >ou can compare current problems with a baseline'.1
What are two advantages of the systematic troubleshooting method'.' (Choose two ,
(Source: Using Troubleshooting Methodology)
A) Asystematic troubleshooting method defines afallback mechanism that can be
used mcase ot network problcms.
Asystematic troubleshooting method makes it easier for you to identify
potential problems on adata network and aCisco Unified Communications
C) Asystematic troubleshooting method creates abaseline mechanism that can be
used when a network is in a functional state
D) You can use atroubleshooting model to instantly fix the problem and restore
me l isco Unitied Commumcations system.
E) You can use atroubleshooting model to methodically reduce alarge set of
possible causes of trouble to asmaller set or to asingle cause. Then you can fix
the problem and restore the Cisco Unified Communications system
05)
Q6)
C)
Di
B)
system
Q7) Match thetroubleshooting model stepto itsdescription. (Source: Using
Troubleshooting Methodology)
A) Define Problem
B) Gather Facts
C) Consider Possibilities
D) Create Action Plan
E) Implement Action Plan
F) Observe Results
G) Document Facts
1. Based on the facts that you gather, consider the most likely possible
causes.
2. Implement the action plan.
3. Analyze the networkproblemand create a clear problemstatement.
4. Create an action plan for the possible causes. Begin with the most likely
problem and devise a plan where you manipulate only one variable.
5. Gather the facts that you need to help isolate possible causes and use the
results to narrow the list of potential causes to the most likely source.
6. If the problem is resolved, consider the process complete and document
any changes, the root cause of the problem, and the steps that resolved the
problem.
7. Analyze the results to determine if the problemhas been resolved.
Q8) To define the problem, first identify the general symptoms. Next, determine what
possible problems these symptoms might indicate. Your problem statements must refer
to the previously established network baselines. You should be able to identify the
characteristics of the network when the network is performing as expected. (Source:
Using Troubleshooting Methodology)
A) true
B) false
Q9) When you gather facts, accurately interviewend users to get all of the pertinent details
of the problem. Which three of the following are example questions to use when
interviewing an end user? (Choose three.) (Source: Using Troubleshooting
Methodology)
A) How long did it work?
B) Is the problem intermittent or docs it always occur?
C) When did you arrive to the office?
D) Are there any messages or tones that play, or does the device simply
disconnect?
E) Is anyone else that you know having this problem, either near you or near the
party that you are calling?
F) Have you mentioned the issues to your manager?
>2010 Cisco Systems, Inc. Introduction lo Troubleshooting Cisco Unified Communications Solutions 1-81
QIO) When creating an action plan, which policy should you employ if multiple possibilities
remain? (Source: Using Troubleshooting Methodology)
A) analyze and include
B) analyze and conquer
C) divide and exclude
D) divide and conquer
Qll) What are twoconsiderations whenyouare developing and executing theaction plan?
(Choose two.) (Source: Using Troubleshooting Methodology)
A) Be less specific to proceed quicker.
B) Be as specific as possible.
C) Make sure that the changes that you made do not make the problemworse. If
the changes do make the problem worse, you should consider each change as a
new cause of issue.
D) Do not consider the impact of the changes on other users.
E) Keep track of exactly what you are testing. It is best that you list the action
plan in a step-by-stcp process on paper.
Q12) What should you do if the previous action plan is a valid action that results in a
desirable configuration but does not solve the root cause? (Source: Using
Troubleshooting Methodology)
A) Remove the action that was implemented and continue with the next action.
B) Leave the action that was implementedand create another action plan.
C) Leave the action that was implemented and redefine the root cause.
013) Which three statements are reasons for creating documentation? (Choose three.)
(Source: Using Troubleshooting Methodology)
A) Documentation maintains the exact steps that you took to solve the problem,
B) Documentation is used to prove that the network is back in order.
C) Documentation provides you with a back-out plan in case the fixes that were
applied worsen the situation overtime.
D) Documentation is used at a later stage to redesign your network.
L) Documentation of the problem and resolution serve as a historical record for
future reference.
) Documentation is distributed among the users to notify them of how to behave
in ease of network problems.
1-82 Troubleshoolmg Cisco Unified Communications (TVOICE] v8 0 2010 Cisco Systems. Inc
Q14) Cisco Unified Serviceability is a web-based troubleshooting tool. What are four typical
functions provided by Csco Unified Serviceability that are used during
troubleshooting? (Choose four.) (Source: Using Troubleshooting and Monitoring
Tools)
A) saves Cisco Unified Communications Manager Service alarms and events, and
provides alarm message definitions
B) provides backup and restore options
C) saves Cisco Unified Communications Manager Service trace information to
various log files
D) reads Cisco Unified Communications Manager Service trace information from
various log files
E) can be used as a packet analyzer tool
F) provides an interface to view the status of feature and network services
G) monitors the disk usage of the log partition on a server or of all servers in the
cluster
It) monitors the CPU and memory utilization of all servers
Q15) Cisco Unified Serviceability allows you to activate and deactivate feature and network
services. You can activate or deactivate as many services as you want at the same time.
(Source: Using Troubleshooting and Monitoring Tools)
A) true
B) false
Q16) Which three actions can you perform for alarms? (Choose three.) (Source: Using
Troubleshooting and Monitoring Tools)
A) Define new alarms.
B) Add user-defined notes for predefined alarms.
C) Set the destination for alarms to a syslog server.
D) Set the level of alarms.
E) Set the event ID.
F) Set the address of the SNMP server.
Q17) Which three statements regarding trace are true? (Choose three.) (Source: Using
A) Trace is disabled by default.
B) Trace can be sent to a syslog server.
C) Trace output can be viewed from the CCMService web page.
D) Trace output can be viewed from the Cisco Unified RTMT.
E) Trace output is in a binary format that needs to be interpreted by the Cisco
Unified RTMT.
F) Trace output can be collected and sent to Cisco TAC.
G) Trace output goes to the hard drive by default.
Q18) Which statement is true when the Digit Analysis Complexity service parameter is set to
its default value of StandardAnalysis? (Source: Using Troubleshooting and Monitoring
Tools)
A) The Digit Analysis Complexity parameter is not shown together with the
generated alarm.
B) Depending on the configured irace level, Cisco Unified Communications
Manager traces do not include details about translation patterns and alternate
matches. Only the finally matched pattern is shown.
C) Cisco Unified Communications Manager traces do not include details about
translation patterns and alternate matches. Only the finally matched pattern is
shown. This behavior does not depend on the configured trace level.
Q\9) Match the trace messages and fields with their descriptions. (Source: Using
A) CCM Stationlnit
B) pss
C) CCMStationD
D) CCM Locations
El CCM MediaResourceManager
I. phone display-related messages, phone tone-related messages, RTP-related
messages, and so on, that are sent from Cisco Unified Communications
Manager to a Cisco Unified IP phime
2. shows if CAC has to be performed, and includes the required (bw), current
(curr), and maximum (max) bandwidth
3. shows information about required media resources such as MTPs or
transcodcrs. as well as media resources that are set up at midcall
_ 4. events such as softkey pressed, key from keypad pressed, phone button
pressed, and so on, that are sent from a Cisco Unified IP phone to Cisco
Unified Communications Manager
5. lists partitions that are available to the calling device
Q20) What is the tool that allows you to test a Cisco Unified Communications Manager dial
plan configuration before deploying it? (Source: Using Troubleshooting and
Monitoring Tools)
Q21) Which ihree functions can be found in the Cisco Unified RTMT? (Choose three.
A) viewing trace output
B) starting services
C) setting alarms
D) viewing alerts
E) CAR
F| performance monitoring
1-84 Troubleshooting Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems, Inc.
Q22) Which three statements about alerts aretrue? (Choose three.) (Source: Using
A) Alerts are configured in the Cisco Unified RTMT.
B) No alerts are configured by default.
C) Alertscanbe sent to a definable email address.
D) You candefinethe time of day that alertscan senda message.
E) Youconfigure alerts in theCisco Unified Serviceability webinterface.
F) You can configure alerts to be written to the trace output.
Q23) Which three statements regarding Trace &Log Central are true? (Choose three.)
(Source: Using Troubleshootingand MonitoringTools)
A) The Remote Browse feature allows a web browser to view trace output.
B) TheRemote Browse feature allows traceoutput tobe viewed within theCisco
Unified RTMT.
C) The CollectFiles feature is usedonlyto collect high-level traces for TAC.
D) YoucanusetheCollect Files feature tocollect traceoutput totext files that
you can use with Windows applications.
E) The Real Time Trace feature allows trace output to be viewed as it occurs
within the Cisco Unified RTMT.
F) The Real TimeTrace feature allows traceoutput to be viewedas it occurs
within supportedJava browsers.
Q24) You can logperformance counters (known as perfmon) locally onyourcomputer and
usetheperformance logviewer in Cisco Unified RTMT to display theperfmon CSV
logfiles that youcollected. Which twotypes of logscanyoucollect? (Choose two.)
A) Cisco CallManager service perfmon logs
B) Alert Manager and Collector perfmon logs
C) Gateways and Trunks perfmon logs
D) Cisco R1S data collector perfmon logs
E) Cisco TFTP perfmon logs
Q25) Inwhich twowayscanyouusetheCiscoUnified Reporting tool of Cisco Unified
Communications Manager to aid in the troubleshootingprocess? (Choose two.)
(Source: Using Troubleshootingand Monitoring Tools)
A) This application gathersinformation fromthe publisherserver only, but not
subscriber servers.
B) This tool provides a snapshot of cluster datawithout requiring multiple stepsto
find the data.
C) Areport that this tool generates combines data from one or more sources on
one or more servers into one output view.
D) This tool can identify potential sources of issues and report them.
2010 CiscoSystems, Inc. Introduction to Troubleshooting CiscoUnified Communications Solutions 1-85
Q26) Which three statements about the Cisco Unified Communications Manager CLI are
true? (Choose three.) (Source: Using Troubleshooting and Monitoring Tools)
A) You can use the CLI toaccess the Cisco Unified Communications Manager
system for basic maintenance and failure recovery.
B) You can use the CLI toaccess the Cisco Unified Communications Manager
system for advanced maintenance andfailure recovery.
C) I o obtain access to the system, open a Telnet session to the Cisco Unified
D) Toobtain access tothe system, either use a hardwired terminal (asystem
monitor and keyboard) or open an SSHsession.
E) For troubleshooting purposes, the CLI set and test command categories are
usually used.
F) For troubleshooting purposes, the CLI show and utilscommand categories are
usually used.
Q27) Which three statements describe how to use the Cisco IOS Software to support
troubleshooting? (Choose three.) (Source: Using Troubleshooting and Monitoring
Tools)
A) Csco IOS Software can write diagnostic infonnation to Cisco Unified
Communications Manager trace log files.
B) Various show commands can aid in problemisolation.
C) Output fromdebug commands provides diagnostic information about various
internetworking events.
D) Use terminal emulator software (such as HyperTerminal) tocollect debugs.
E) To stopa debug, use the debug stop or no debug start command.
F| Before running any Cisco IOS voicegatewaydebugs, makesure that service
timestamps debug datetime msec isglobally configured onthe gateway.
Q28) What doyou need toconfigure onCisco Unified IP phones or onCisco Catalyst
switches to collect a sniffer trace from conversations between the phone and Cisco
Unified Communications Manager? (Source: Using Troubleshooting andMonitoring
Tools I
Troubleshooting CiscoUnified Communications (TVOICE) v8.0 2010CiscoSystems, Inc
Module Self-Check Answer Key
on B.F.
Q2) A
03) B. D. F
04) A. D
0?>
B. n. G
Q6) B.E
0^) A-3. B-5, C-l.D-4, E-2, F-7.G-6
08) A
Q9) B.D.r
Q10) I)
QUI B. h
Q12I B
OH) a, c. a
014) A, C. F, G
Q1S) B
016) B. C. D
017) D. F. G
018) C
QI9) A-4. B-5.C-l.D-2. E-3
Q20) Dialed Number Analyzer
021) A, D. F
022)
C. D. E
023) B. D. E
Q24) B. D
025) H.C
026) A, D. F
Q27) B.C.
Q2K) Sw itched Port Analy/er (SPAN)
2010 Cisco Systems. Inc. Introduction to Troubleshooting Cisco UnifiedCommunications Solutions 1-87
Module 2
Manager Troubleshooting
_ Overview
After you migrate your voice network to include Cisco Unified Communications Manager
functionality, the entire structure of your voice network changes. Nearly all devices should now
mm communicate across the data network structure, keeping the number of analog devices used at a
minimum. Because the PBX functionality has been replaced with the Cisco Unified
Communications Manager cluster, much of your troubleshooting focus nowchanges to include
the Cisco Unified Communications Manager system.
The Cisco Unified Communications Manager is an appliance that is based on Linux and uses
IBM Informixas the database for configuration, historical data, and all user information. This
5' module will cover issues that are related to these areas: Cisco Unified Communications
*""' Manager availability, database replication issues, Lightweight Directory Access Protocol
(LDAP) issues, endpoint registration issues, and gateway registration issues.
Module Objectives
Upon completing this module, you will be able to isolate and troubleshoot reported issues that
^ relate toCisco Unified Communications Manager. This ability includes being able to meet
^^ these objectives:
Define the issues that are related to gateway and endpoint registration in Cisco Unified
Communications Manager and that enable the learner to describe common solutions to
^^ these issues
Explain issues that can cause Cisco Unified Communications Manager to become
unavailable in the network and describe how to isolate and troubleshoot these issues
mm Define how to identifydatabase replication issues in Cisco Unified Communications
Manager clusters and how to repair or re-create database replication
^ Explain howto troubleshoot LDAP synchronization or LDAP authentication issues when
^^ using LDAP integration
Troubleshooting Cisco UnrfiedCommunications (TVOICE)v8.0 2010 Cisco Systems. Inc
Lesson 1
Troubleshooting Common
Gateway and Endpoint
Registration Issues
^ Overview
Endpoint registration issues are some of the most common issues you will face when
troubleshooting Cisco Unified Communications Manager. Many issues can prevent IP phones
fromregistering correctly, includingnetwork connectivity, IP phone settings, DHCP settings,
and Cisco Unified Communications Manager configuration settings.
Objectives
Upon completing this lesson, you will be able to define the issues that are related to gateway
and endpoint registration in Cisco Unified Communications Manager and describe common
solutions to these issues. This ability includes being able to meet these objectives:
Describe how a Cisco Unified IP phone that operates with SCCP or SIP initializes to a
Cisco Unified Communications Manager and list the common issues that can prevent an
endpoint from being able to register
Use ping from Cisco Unified Communications Manager to troubleshoot endpoint
connectivity and explain the procedure to verify Cisco Unified Communications Manager
TFTP settings and to create a new configuration file to help resolve the issues
Explain how to use the Status Message screen on an IP phone to view recent messages and
explain how to check network and device settings from the IP phone
Explain the MGCP gateway registration process and the basic MGCP gateway
communications in a Cisco Unified Communications system and list the most typical issues
that might be encountered during these processes
Explain the steps for verifying the MGCP gateway status and the steps for viewing and
interpreting registration and communication errors
Explain issues with H.323 and SIP gateway communication in a Cisco Unified
Communications system
IP Phone Initialization
2-4
This topic describes howa CiscoUnified IPphone, that operates with SkinnyClientControl
Protocol (SCCP) or Session Initiation Protocol (SIP), initializes to a Cisco Unified
Communications Managerand lists the common issuesthat can prevent an endpoint frombeing
able to register.
Two PoE mechanisms supported.
Most common cause of problems is a mismatch of the inline power
between the endpoint and the switch.
Modified FLP
Reflected FLP
Resistive Detection and Classification
Return Current
I Cisco Discovery Protocol message from
IP Phone 7960. "I need 6.3 W
Cisco Discovery Protocol (VLAN number)
Cisco Prestandard
-c^^ Cisco
|Sr.?til Catalyst
l!^ Switch
802.3af
These steps take place during phone bootup for all Cisco Unified IP phones that use the Cisco
prestandard Power over Ethernet (PoE):
Step 1 The switch sends a special tone, called a Last Link Pulse (FLP), out of the interface.
The FLP goes to the powered device, in this case, a Cisco Unified IP phone.
Step 2 The Cisco Unified IP phone has a special relay that connects its Ethernet receive
pair with its Ethernet transmit pair. The relay is closed when no power is being
supplied to the phone, and the closed circuit allows the FLP that is sent in Step l to
arrive back at the switch. This will not happen when the attached device is a non-
PoE-capable device, such as a PC device, because there is no relay. In this case, the
FLP does not make it back to the switch and no power is applied.
The switch applies power to the line, and the link should go up within 5 seconds.
The powered device (Csco Unified IP phone) boots.
Step 3
Step 4
Step 5 The Cisco Unified IP phone uses Cisco Discovery Protocol to tell the switch
specifically how much power it needs.
Some Csco Unified IP phones use the standards-based IEEE 802.3af PoL. These steps take
place for these phones during bootup:
Step 1 The switch constantly applies direct current to all ports that have a powered device
that are attached to them. The powered device is connected and will have a
resistance of 25 ohms if it is PoE-complianl.
Step 2 The switch detects that the device is a PoE-capable device.
Troubleshooting Cisco Unified Communications (TVOICE] vB 0 2010 Cisco Systems, inc
m*
iJm
Step 3 The switch applies power to the link in low-power mode, which is 6.3 W.
Step 4 The powered device (Cisco Unified IP phone) boots.
Step 5 The Cisco Unified IP phone uses Cisco Discovery Protocol to tell the switch
specifically how much power it needs.
Ifthe Cisco Unified IP phone does not receive power from the switch, first verify which
mechanisms are supported by the switch and ensure that the endpoints have acommon
mechanism supported. The most common cause of problems is amismatch of the supported
inline power between the endpoint and the switch.
Cisco UnifiedCommunications Manager Troubleshooting 2-5
2-6
C'
DHCPDISCOVER
DHCP
DHCTOFFER(|PAddr Def.GW, TFTP. DNS*) f^M Se^r
TFTP GET (SEP003094C3AD7E cnf xml)
TFTP Data(SEP003094C3AD7E.cnf xml)
SCCP Registration with Unified CM'
DNS .5 oplionai
" Uniiied CM - Cisco Unitied Communications Manager
TFTP
: v W Server
I
VK
i
Cisco Unified
Communications
Manager
Cisco Unified IP phones continue with these postpowcr steps:
Step 1 The switch uses Cisco Discovery Protocol to inform the Cisco Unified IP phone of
the voice VLAN (auxiliary VLAN) to which the phone belongs.
Step 2 TheCisco Unified IPphone initializes theIPstack andsends out a
DHCPDISCOVER broadcast requesting an IP address on the voice VLAN scope.
Note
Step 3
Step 4
Step 5
It ispossible to hardcode the IP address, subnet mask, default gateway, Domain Name
System (DNS), and TFTP server on the Cisco Unified IP phone and skip the DHCP steps
However, the DHCP should be used to minimize the administrative load that is required to
hardcode these settings
The DHCP server hears the broadcast and assigns an IP address from the scope for
the voice VLAN subnet, subnet mask, default gateway, DNS (optional), and address
of the 1FTP server (typically the Cisco Unified Communications Manager publisher
server). The DHCP server then sends all of the settings back tothe Cisco Unified IP
phone in the form ofa DIICPOFFER message.
The phone receives the Dl ICPOFFER and applies the values that it receives.
One value that isearned inthe DIICPOFFER message isthe address of the TFTP
server. TheCsco Unified IPphone uses this information to make a connection to
the TFTP server.
Troubleshooting CiscoUnified Communications (TVOICE] v8.0
tm
Step 6
Step 7
The TFTP server contains configuration files and profile files. Aconfiguration file
includes parameter for connecting to Cisco Unified Communications Manager and
information about which image load aphone should run. Aprofile file contarn
various parameters and values for phone and network settings The C,sco Unified IP
phone first requests its SEP<mac>.cnf.xml file from the TfTP server. If the
SFP<mac> cnf.xml file is not found on the server, the phone requests the
XMLDefault.cnf.xml file. From this file, the Cisco Unified IP phone obtains its list
of Cisco Unified Communications Manager servers and then registers to the pnmary
server.
The configuration file defines how the Cisco Pphone communicates with Cisco
Unified Communications Manager. After obtaining the file from the TFTP server
the phone attempts to make aTCP connection to the highest priority Cisco Unified
Communications Manager on the list. If the phone was manually added to the
database Cisco Unified Communications Manager identifies the phone. If the phone
was not manually added to the database and autoregistratmn is enabled in Cisco
Unified Communications Manager, the phone attempts to autoregister itself in the
Cisco Unified Communications Manager database. Cisco Unified Communcations
Manager informs devices that use .cnf or .cnf.xml format configuration files of their
load ID. Devices that use .xml format configuration files receive the load ID in the
configuration file.
Note The extension numbers, speed dials, and other settings are assigned when the SCCP IP
phone registers. They are not contained in the SEP XML file.
>2010 Cisco Systems. Inc Cisco Unified Communications Manager Troubleshooting
Common DHCP and TFTP Issues
This section describes the most common DHCP and TFTP issues
Common DHCP issues:
- DHCP server is unavailable or not reachable.
1DHCP server does not have ascope that isdefined for the
subnet of the phone.
All addresses in the scope arecurrently leased out.
- The phone is on a different subnet than the DHCP server
DHCP relayis not enabled on the router.
Common TFTP issues:
DHCP scope does not define TFTP server option 150.
DHCP option 150 points tothe wrong address.
- TFTP service on the address specified is stopped or hung.
If the phone does not receive an IP address from DHCP, one of the following is likely the
'""lie: J
The DHCP server is unavailable or cannot be reached by an endpoint.
The DHCP server does not have ascope that is defined for the subnet of the phone.
All addresses in the scope are currently leased.
The phone is not on acommon subnet with the DHCP server, and DHCP relay is not
enabled on the router.
Ifthe phone does not receive its configuration file from the TFTP server, one of the followim-
might bethecause of the problem:
The Dl ICP scope did not have the TFTP server that is defined as option 15(1.
The option 150 in the Dl ICP scope points to the wrong address.
The TFTP service on the specified address is stopped or hung.
>2010Cisco Systems, Inc
It*
Troubleshooting Endpoints Using Cisco Unified
Communications Manager Tools
This topic describes how to use Cisco Unified Communications Manager to troubleshoot
endpoint connectivity and explains the procedure to verify Cisco Unified Communications
Manager TFTP settings. This topic also explains howto create a new configuration file to help
resolve the issues.
Using Ping to Cisco If
Cisco Unified Operating System Administration > Services > Ping
CLI
4dBin=UCili Datwork ping 10.1 5.39
PIMC 10.1.5.19 (10 1.5.291 5<4> byt.. o{ 4t.
64 bytti frrx 10.1 5.29, ICVp *q-0 ttl.64 tin. 0 503 mi
64. byt.. IroM 10.1 5.25, icnp qui ttl.64 tin 0 SSI K
64 byt.. Ires 10.1 5.29.
l"P q.2 ttl.64 tin. 0 46B n
64 byt. ([a> 10.1 5.29: inp a*q>3 ttl,61 tin. 2 29 a.
--- 10.1.5.IS plug it.el tic.
p.ck.t. t.. initlil, 4 r.c. v.d, 0% p.ct.t la tin. 3 0 02m.
rtt min/tvg/miW-d iv - 0 tCt/C.Hl/i. 295/0 771 pip. 2
To verify connectivity to IP phones, use the ping or traceroute tool from either the
administrator PCor the Cisco IOS router. The ping should be performed from a server that is in
the Cisco Unified Communications Manager group. In Cisco Unified Communications
Manager, performping fromthe Cisco Unified Operating SystemAdministrationweb pages.
Alternatively, youcan perform a pingfromthecommand-line interface (CLI) by usingthe
command utils network ping <ip address>.
Note Use the Cisco Unitied Communications Manager server to initiate the ping, because Cisco
Unified IP phones do not have that option.
If the ping is unsuccessful, try pingingfromanotherdevice inthe path to try to isolatethe
network connectivity issue.
2010 Cisco Systems, tnc
Verify TFTP Server Configuration
Verify that the TFTP server from which the Cisco Unified IP phone will receive its
configuration tile is started and activated.
"UV .01
ip
dhcp exclu ded addras 10 1.5 1 10.1.
s.iSjS'
ip dhcp exclu ded addrea 10 1.5 31 10.1 .5^*54
ip
dhcp pool Voi
e
necw ik 10 .1. .0 255 255 255
dea lc-rc uter 10.1. .1
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ip 10.1.5 2
a'sii.=.yiJ-Jki&!'>
The configuration file that the Cisco Unified IP phone receives from the TFTP service defines
which Cisco Unitied Communications Manager servers to register with and the order of
preference of those servers. To verify that both the Cisco CallManager and the Cisco TFTP
Services arc activated and started, perform these steps:
Step 1 Open the Cisco Unified Communications Manager Serviceability web interface.
Step 2 Choose Tools > Control Center - Feature Services.
Step 3 Choose the desired Cisco Unified Communications Manager server from the drop
down menu.
Step 4 View the status of the Cisco TFTP and Cisco CallManager Service.
If the TFTP server is running, make sure that the correct TFTP option 150has been offered to
the Cisco Unified IP phone. This is the option of DHCP subnet settings, and it can be
configured either on Cisco UnifiedCommunications Manager or on a Cisco IOS router as seen
in the figure.
Verity' that the Cisco Unified IP phone has received the correct address of the TFTP server:
Step 1 Press Settings at the Cisco Unified IP phone.
Step 2 Choose Network Configuration.
Step 3 Choose IPv4 Configuration.
Step 4 Scroll the Cisco Unified IP phone screen down to locate TFTP Server entry.
If you continue to have problems with a particular phone, the configuration file may be corrupt.
The next step is to re-create the configuration file to try to resolve the problem.
To create a newconfiguration file, performthese steps:
Step 1 Fromthe CiscoUnifiedCommunications ManagerAdministration web interface,
choose Device > Phone > Find to locate the phone for which you want to re-create
the configuration file.
Step 2 Choose Delete to remove the phone fromthe Cisco Unified Communications
Manager database.
Step3 Add thephone back totheCisco Unified Communications Manager database.
Step 4 Power-cycle the phone.
After youpower-cycle the phone,you shouldverifythat the phone is in the CiscoUnified
Communications Manager database. Search for the phone by entering any known properties of
the Cisco Unified IP phone in question at the Device > Phone > Find page.
>2010 Cisco Systems, Inc. Cisco Unified Communications Manager Troubleshooting 2-11
Troubleshooting Endpoints from Endpoints
This topic explains howto use the Status Message screen on an IPphone to view recent
messages andexplains how tocheck network anddevice settings from the IPphone.
*3ih
i i 11K I. I ihii
Settings > Status > Status Messages
7M5GSMIingi B -3
Dswco Configurmion
Sacurfty Configuration
Model WormaUon
jSWi.
tt JM
St Hi Hi
04:is:3!Tnm LiltUpdM.it
04 28 32 No IPv4 DNSStrttr
04 :e IS WNNot Condguind
04 16 1BSEP002<I1T*90D:2.crIi
1 he Status menu on the Cisco Unified IP phone allows users to view status messages on the IP
phone. To \ iew these messages, press Settings, and then choose option Status to enter the
Status menu. Next, choose option Status Messages to display any stanis messages that appear.
You can access the Status Messages screen at any lime, even if the phone has not finished
bootim;.
2-12 Troubleshooting Cisco Unified Communications (TVOICE) v8 0 )2010 Cisco Systems, Inc
The table describes the many status messages that can appear on aCisco Unified IP phone.
Status Messages on the Cisco Unified IP Phones
Message
BOOTP server used
CFG file not found
CFG TFTP size error
Checksum error
CTL installed
CTL update failed
DHCP timeout
Dial plan parsing error
Description
Possible Explanation and Action
The phoneobtained its IP
address from a Bootstrap
Protocol (BOOTP) server
rather than a DHCP server.
None. This messageis informational only.
The configurationfilewith
the MAC address in the
name and the default
configuration file were not
found on the TFTP server.
The configurationfile is too
large for the filesystem on
the phone.
Downloaded software file is
corrupt.
A Certificate Trust List (CTL)
Die is installed in the phone.
The phone could not update
its CTL file.
The DHCP server did not
respond.
The phone could not parse
the dial plan XMLfile
properly.
The configuration file for a phone iscreated when
thephoneis addedtothe Cisco
Unified Communications Manager database. If the
phone has not been added to the Cisco
Unified Communications Manager database, the
TFTP server generates a CFG File Not Found
response. Note the following:
Youmust manuallyadd the phone to Cisco
Unified Communications Manager ifyou are
not allowing phonesto autoregister.
- if you are using DHCP, verify that the DHCP
server ispointing tothecorrect TFTP server.
If you areusing static IP addresses, check the
configuration of the TFTPserver.
Power-cycle the phone to force the Cisco Unified
IP phone to download the CFG file again. If this
still shows as a bad file, trydeletingthe phone
from the configuration and readding ittore-create
the file on the TFTP server.
Obtain a new copy ofthephonefirmware and
place it in the TFTPPath directory. You should
only copy files into this directory when the TFTP
server software is shut down; otherwise, the files
could becorrupted. ^_^_
None. This message is informational only.
There isa problem with theCTL file ontheTFTP
server. Use the CTL client toverify that the CTL
has the appropriate servers init and fix any
problems found. ^^^^
Network is busy: The errors should resolve
themselves when the network load reduces.
No network connectivity between the
DHCP server and the phone: Verifythe
network connections.
DHCP server is down: Check the
configuration of the DHCPserver.
Errors persist: Consider assigning a static
IP address.
There is a problem with the TFTP-downloaded
dial plan XMLfile.
Message
DNS timeout
DNS unknown host
Duplicate IP
Error update locale
File auth error
File not found
IP address released
Load auth failed
Load ID incorrect
Description
Possible Explanation and Action
DNS server did not respond.
themselves when thenetwork load reduces.
No network connectivity between the DNS
server and the phone: Verify the network
connections.
DNS server is down: Check the configuration
of the DNS server.
DNS could not resolve the
name of the TFTP server or
Cisco Unified CallManager.
Verify thatthe hostnames oftheTFTP server
orCisco Unified CallManager areconfigured
properly in DNS.
Consider using IPaddresses ratherthan
hostnames.
Another device is using the
IP addressthat isassigned
to the phone.
One or more localization
files could not be found in
the TFTPPathdirectory or
were invalid. The locale was
unchanged
An error occurred when the
phone tried to validate the
signature of a signed file.
This message includes the
name of the file that failed.
The phone cannot locate on
the TFTPserver the phone
load filethat is specified in
the phone configurationfile.
The phone has been
configured to release its IP
address.
A signed phone load file has
been modified or renamed.
Load ID of the software file
is of the wrong type
If the phone has a static IP address, verify that
you havenot assigned a duplicate IPaddress.
If you areusing DHCP, check theDHCP server
configuration.
Check that the following files are located within"
subdirectories in the TFTPPath directory:
Located insubdirectory with same nameas
network locale:
tones.xml
Located insubdirectory with same nameas
user locale:
glyphs.xml
dictionary.xml
kate.xml
dictionary.xml
The file iscorrupted. If thefile isa phone
configuration file, delete the phonefrom the
database using Cisco Unified Communications
Manager Administration. Then add the phone
backto the CiscoUnified Communications
Manager database usingCisco
Unified Communications Manager
Administration.
There is a problemwith the CTL file, and the
keyfor the server fromwhich filesare obtained
is bad. Inthis case, run the CTLclient and
update the CTL file, making sure that the
proper TFTP servers are included in this file.
Makesure that the phone toadfile is on the TFTP
server and that the entryinthe configuration file is
correct.
The phone remains idle until itispower-cycled or
you reset the DHCP address.
Make surethat thephone load file that thephone
is downloading has not been altered or renamed.
Check theload ID assigned tothephone (from
Cisco Unified Communications Manager, choose
Device > Phone). Verify that the load ID is
entered correctly.
2-14
Message Description Possible Explanation and Action
Load rejected HC The application that was
downloaded is not
compatible with the phone
hardware.
Occurs if you attempt to install a version of
software on a phone that does not support
hardware changes on the newer phone.
Check the load IDassigned to the phone (from
Cisco Unified Communications Manager, choose
Device > Phone). Re-enter the load that is
displayed on the phone.
Load server is invalid Indicates an invalid TFTP
server IP address or name
in the Load Server option.
The Load Server setting is invalid. The Load
Server setting specifies a TFTP server IP address
or name from which the phone firmware can be
retrieved for upgrades on the phones.
Check the Load Server entry (from Cisco
Unified Communications Manager Administration,
choose Device > Phone).
No CTL installed A CTL file is not installed in
the phone.
Occurs if security is not configured, or, if security
is configured, because the CTL file does not exist
on the TFTP server.
No default router DHCP or static configuration
did not specify a default
router.
If the phone has a static IP address, verify
that the default router has been configured.
Ifyou are using DHCP, the DHCP server has
not provided a default router. Check the
DHCP server configuration.
No DNS server IP A name was specified but
DHCP or static IP
configuration did not specify
a DNS server address.
If the phone has a static IP address, verify that
the DNS server has been configured.
If you are using DHCP, the DHCP server has
not provided a DNS server. Check the DHCP
server configuration.
Programming error The phone failed during
programming.
Attempt to resolve this error by power-cycling the
phone. If the problem persists, contact Cisco
technical support for additional assistance.
TFTP access error TFTP server is pointing to a
directory that does not exist.
If you are using DHCP, verify that the DHCP
server is pointing to the correct TFTP server.
If you are using static IP addresses, check the
configuration of the TFTP server.
TFTP error The phone does not
recognize an error code that
is provided by the TFTP
server.
Contact the Cisco Technical Assistance Center
(TAC).
TFTP file not found The requested load file (.bin)
was not found in the
TFTPPath directory.
Check the load IDassigned to the phone (from
Cisco Unified CallManager, choose Device >
Phone). Verify that the TFTPPath directory
contains a .bin file with this load ID as the name.
2010 Cisco Systems, Inc. Cisco Unified Communications Manager Troubleshooting 2-15
Message Description Possible Explanation and Action
TFTP server not authorized The specified TFTP server
could not be found in the
CTL of the phone.
The DHCP server is not configured properly
and is not serving the correct TFTP server
address. In this case, update the TFTP server
configuration to specify the correct TFTP
server.
If the phone is using a static IP address, the
phone might be configured with the wrong
TFTP server address. In this case, enter the
correct TFTP server address in the Network
Configuration menu on the phone.
If the TFTP server address is correct, there
might be a problem with the CTL file. In this
case, run the CTL client and update the CTL
file, making sure that the proper TFTP servers
are included in this file.
TFTP timeout TFTP server did not
respond
themselves when the network load reduces.
No network connectivity between the TFTP
server and the phone: Verify the network
connections.
TFTP server is down: Check the configuration
of the TFTP server.
Version error The name of the phone load
file is incorrect.
Make sure that the phone load file has the correct
name.
XmlDefault cnf.xml, or
.cnf.xml corresponding to ihe
phone device name
Name of Ihe configuration
file.
None. This is an informational message indicating
the name of the configuration file for the phone.
Cisco Unified IP Phone Network Configuration
This sectiondescribes howto displayNetwork Configuration and DeviceConfiguration on a
Cisco Unified IP phone.
:isco Unified
;onfiguration
To view and verify network settings, whether manually set or obtained by using a DHCP
server, perform these steps:
Step 1 On the Cisco Unified IP phone, press Settings.
Step 2 Choose Network Configuration.
Step 3 View the phone MAC address (Host Name) and VLAN ID.
Step 4 If you choose IPv4 Configuration, view DHCP server settings, IP address, subnet
mask, default gateway, and TFTP server address.
To view and verify Device Configuration, perform these steps:
Step 1 On the Cisco Unified IP phone, press Settings.
Step 2 Choose Device Configuration and then Unified CM Configuration.
Step 3 View the Cisco Unified Communications Manager servers that the Cisco Unified IP
phone is currently using for registrations. They are shown at the Cisco Unified IP
phone in the sequence fromthe most- to the least-preferredserver.
2010 Cisco Systems, Inc. Cisco Unified Communications Manager Troubleshooting
MGCP Gateway Initialization and Communication
This topic explains the Media Gateway Control Protocol (MGCP) gateway registration process
and the basic MGCP gateway communications ina Cisco Unified Communications system and
lists themost typical issues that might beencountered during theseprocesses.
Cisco IOS MGCP gateway communication behavior
- Endpoints on the gateway register with the Cisco Unified
Communications Manager server.
' CiscoUnified Communications Managermust have the gateway
added to its configuration database.
e Endpoints are fullyunder the control of Cisco Unified
' MGCP configuration on the gateway must be performed initially.
b The ability to download configuration files to the MGCP gateway
enables adds, moves, and changes to MGCPendpoints to be
performed inthe Cisco Unified Communications Manager web
interface.
" Any dial plan that is programmed into the router is not used
when the MGCP gateway is registered to a Cisco Unified
Endpoints in MGCP represent the source for call data (Real-Time Transport Protocol [RTP] or
IP)that is flowing through thegateway. Acommon type of endpoint is found at thephysical
interface between the plain oldtelephone service (POTS) or public swilched telephone network
(PSTN) service andthe gateway; thistypeof endpoint might beananalog voice port or a
digital service level 0 (DSO) group. There are other types of endpoints as well, and some arc
logical rather than physical. Anendpoint is identified by a two-part endpoint namethat contains
the name of the entityon whichit exists (for example, an accessserver or router) and the local
name bywhich it is known (forexample, a port identifier). Theexample of an MGCP endpoint
identifier would be SO DS1-0.1(gJIQ.
Call agents manage call flow byusing standard MGCP commands that aresent tothe endpoints
that are undertheir control. The commands are delivered in standardASCII text, and can
contain sessiondescriptions that are transmitted in SessionDescription Protocol (SDP), which
is a text-based protocol. These messages are sent over IPor User Datagram Protocol (UDP).
Call agents keeptrack of endpoint and connection statusby usinggateway reporting of
standard eventsthat are detected from endpoints and connections. Call agentsalso direct
gateways to applycertainstandard signals whena POTS or PSTN connection expects them.
For example, when someone picks up a telephone handset, an off-hook event is detected on an
endpoint on the residential gateway towhich thetelephone is connected. Thegateway reports
theevent toa call agent, which orders thegateway toapply thedial tone signal totheendpoint
that is reporting the off-hook event. The person picking up the handset hears a dial tone.
2-18 Troubleshooting Cisco UnifiedCommunications (TVOICE) v8 0
When youtroubleshoot MGCP, it is important tounderstand thecommunication behavior of
MGCP gateway, which includes the following:
Endpoints on the gateway will each register with the Cisco Unified Communications
Manager server.
Cisco Unified Communications Manager must have the gateway added to its configuration
database.
Endpoints are under the control of the Cisco Unified Communications Manager server to
which the endpoint is actively registered.
Some initial configuration must be performed on the gateway, mostly related to network
connectivity.
If the abilityto download configuration files has beenconfigured on the gateway, all adds,
moves, and changes to MGCP endpoints thereafter only require configuration in the Cisco
Unified Communications Manager web-based interface.
If there is a dial planthat is programmed intothe gateway, it is not usedduringthe periods
when the MGCP gateway is registered to a Cisco Unified Communications Manager,
server.
2010 Cisco Systems, Inc Cisco Unified Communications Manager Troubleshooting 2-19
Cisco IOS MGCP Gateway Registration
2-20
This section describes howMGCP gateways register withCiscoUnifiedCommunications
Manaeer.
=v-,a,w~i~iv a, -s JWV.$KS***iA-V~.&w,i=!S.. H , -V*
Trie gateway bools
The IP slack is nni'iali-?eO
TCP conrwaion is opened
Res:al in Progress is sen' m Cisco
IJnted Communications Manager
rhe gateway is bang brought
into secvice
Acknowledgment (ACKj
Acknowledgment tACKj
OK
AUEP
Endponiis no* registered with Cisc
Unified Conimun=caiions Manager
I
Acknowledgment (ACK)
Cisco Untied Communicalions
Manager sends an Audit Endpoint
message per each endpanl.
Cisco Umhed Communications
Manager sends a Notification
Regnes! per each endpoint
The tigure describes howCisco Unitied Communications Manager registers MGCPgateways
in its database. Follow these steps to register a Cisco IOS gateway:
Step 1 Boot the gateway.
Step 2 Initialize the IP stack.
Step 3 ATCP connectionopens between the Cisco UnifiedCommunications Manager and
the Cisco IOS MGCP gateway.
Step 4 If you use the command ccm-manager config server <1P address>, the gateway
obtains its configuration file. This file contains the Cisco Unified Communications
Manager Group, which defines the ordered list of Cisco Unified Communications
Manager servers to which the gateway will attempt to register. If the ccm-manager
config server <IP address> is unavailable, a manual contiguration is applied and
the gateway registers with the Cisco Unitied Communications Manager that is
defined by the MGCP commands.
Step 5 The gateway sends a Restart InProgress (RSIP) message to the first-choice Cisco
Unified Communications Manager.
Step 6 An acknowledgment (ACK) message is sent in response to indicate that the message
was received.
Note The ACK commands are standard TCP acknowledgments of the received command.
Step 7 The Cisco Unified Communications Manager sends AudilEndpoint (AUEP)
messages to all endpoints that are configured in the Cisco Unified Communications
Manager database.
Step8 ACK messages aresent inresponse to theAUEP messages to indicate successful
reception.
Step9 TheCiscoUnified Communications Manager sends NotificationRequest (RQNT)
messages toall endpoints for thatgateway that areconfigured intheCisco Unified
Communications Managerdatabase, askingto be notifiedif any changes occur.
Step 10 Ihe endpoints will nowshowas "registered" inthe CiscoUnifiedCommunications
Manager web interface.
2010 Cisco Systems, Inc. Cisco Unified Communications Manager Troubleshooting
Cisco IOS MGCP Gateway Registration Issues
This section describes the common issues that areexperienced during theMGCP gateway
registration process.
"isco (03 MGCP Gateway Kegi&ttaii
MGCP gateway common registration issues:
1 IP network connectivity
1 IP addressing problems with DHCP or manual settings
Cisco Unified Communications Manager services not running
Cisco CallManager and Cisco TFTP
- Cisco IOS Software version incompatible with Cisco Unified
Communications Manager version
- Missing or incorrect configuration in Cisco Unified
Communications Manager
Missing or incorrect configuration in the Cisco IOS MGCP
gateway
Theseissuescan cause CiscoIOSMGCP gateway registration problems:
Networkconnectivity issues when there arc connectivity issues between the Cisco Unified
Communications Manager and the gateway.
DHCPserver or scope settings are missing or are wrong (if using DHCP). If DHCPis not
used, manually entered IPsettings are incorrect or missing.
The Cisco CallManager or Cisco TFTP Service is not running at Cisco Unified
The Cisco IOS Software version is out-of-date, and it is incompatible with the Cisco
Unified Communications Manager version.
MGCPgateway is misconfiguredat the Cisco Unified C'ommunications Manager:
1he domain name in Cisco Unified Communications Manager does not match the
domain name on the gateway (Cisco IOS gateway only).
The slot, module, or port that is specified in the Cisco Unified Communications
Manager configurationdoes not match the physical configuration of the gateway.
The gateway has not been added to the database.
Troubleshooting Cisco Unified Communications (TVOICE) v8 0 >2010Cisco Systems, Inc
The MGCP gateway is misconfigured in Cisco IOS Software:
The config server command points to the wrong IP address.
- The gateway has the modules that are installed in slots that do not match the
configuration in Cisco Unified Communications Manager.
- MGCP commands are missing, incorrect, or incomplete.
_ The interface that is used as the source IP address for MGCP packets is in the down
state and, as a result, the gateway will notregister.
Cisco Unified Communications Manager Troubleshooting
2-23
Cisco IOS MGCP Gateway and Endpoint
This section highlights the mutual relation between Cisco Unified C
jottings and Cisco IOS MGCP gateway settings.
2-24
SlotO/SubumtOVPorti
Hostname: HQ1
T1 PR|\
PSTN
SjBICk-Kieg.HiJi
HOi .,ti r. i,
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-ommunications Manager
The s!ot, mod ,c (subunit)i and pon that are ^ mfte Communications
Manager configuration mus, match with the physical configuration of the MGCPâ"
TI he domain name (gateway name) in Cisco Unified Communications Manager must match
uuhrhe hostname that Kconfined on the gateway (Csco IOS gateway onfyT, ^1the
Troubleshooting Cisco Unified Communications (TVOICE) V8 0
2010 Cisco Systems,
Cisco IOS MGCP Gateway Configuration Elements
This section reviews the relevant parts of Cisco IOS MGCP gateway configuration
Cisco IOS MGCP Gateway
Eiements
Hostname
Primary Unified CM*Server
Secondary Unified CM*Server
TFTP Configuration Server
PRI controlled by MGCP
Q.931 backhauled to Unified
CM*
Hg-l#show
mgcp
agap *U-f*aC 10-1.1-2
Bott-aasfr* ***jaa*nt-bot 1&.1.1.1
ccn-managar mgcp
ceM-waagar eoniifl r** IB.1.1,1
ccm-managec config
I
controller Tl 0/1/0
framing est
linecode bSis
fH-^totp tnii i-a* rvio mgov
i
lntnrtace Ser1*10/1/0.33
no ip address
isfln switch-type primary-fil
isdn incoming-voice voice
no cdp snable
begin mgcp
[unilied CM - Csco United Communications Manager
The figure shows the relevant MGCP configuration from aCisco IOS Software-based MGCP
gateway. You should compare these values and correlate them with the configuration in the
Cisco Unified Communications Manager Administrative web interface.
Verifying MGCP Gateway Status
This ,opic explains the steps for verifying the MGCP gateway status and the steps for vie* tng
and interpreting registration and rnmmnn.Wmn ., * h
nerpreting registration and communication errors
2-26
PodlHQ#sh
MGCP Do ma
Priori ty
Name: PodlHO
nd Backup
^kup Ready
Nc
Current active Call Hanager:
Backhaul/Bedundant link port;
Failover Interval;
Keepalive Interval:
Last keepalive sent:
1;41:10)
Last MGCP traffic time:
00:00:22)
Last failover time:
Last switchback time:
Switchback mode:
mocp Palloack mode:
Last HGCP Fallback start time:
... Output omitted ...
10.1.1.2
2428
30 seconds
15 seconds
I0:52;49 UTC Jun 17 2006 (elapsed time:
13:33:37 UTC Jun 17 2006 (elapsed time:
12:33:37 UTC Jun 17 2006 from (10.1 1 21
12:33:07 UTC Jun 17 2006 from (10.1.1.1)
Graceful
Not Selected
None
J
One of the most useful commands for verifying that the MGCP gateway has registered
normally to aCisco Unified Communications Manager server is the show ccm-manager
command and its vanations. This command will display current registration status as well as
primary, backup, and secondary backup servers. The command also shows MGCP properties
such as keepalive, tatlover interval, andswitchback mode.
Troubleshooting Cisco Unified Communicalions (TVOICE) v8.0
2010Cisco Systems. Inc
Verify the status of MGCP gateway registration:
Choose Device > Gateway > Find
Click See Endpoints on the desired gateway
Select the desired endpoint to validate the configuration and
status
(fry** "n*
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rrsry,-^v*i ow*.?--*-***-p*** *H.i*f;
You can also use the Cisco Unified Communications Manager web interface to determine the
registration status of a gateway device. Choose Device >Gateway > Find in the web interface
and use the Find function to help narrow the search of gateways. Click the See Endpoints link
of the desired gateway, which displays a list of endpoints that are on the selected gateway. The
status appears on the page, and on the page that appears if you select an endpoint link.
) 2010 Cisco Systems, Inc.
MGCP Gateway Monitoring Commands
This section describes the commands that are used for monitoring MGCP gateway registration,
,t^* &?''"*' t.&ii^^l^^JZ-iî^^î^î.&.Sh2'i'î. 'i^th willkWiau
router*
show ecu-manager
Displays a list of Cisco Unified Communications Manager servers
and their current status and availability.
routerS
show mgcp endpoint
Displays information for endpoints that are controlled by using MGCP
router#
show mgcp connection
Displays information for active connections that are controlled by
using MGCP.
To troubleshoot Cisco gateway devices, the command show ccni-manager is a good place to
start. This command displays the status of the registration of the Cisco IOS gateway to the
primary Cisco Unitied Communications Manager server. This command also displays any
configured backups as well as other configurable settings.
Here is sample output from the show ccm-manager command. Note that the primary server
10.1,1,2 has a status of Backup Ready. This status implies that the primary server is
unavailable, and the MGCP gateway is currently in the process of registering with the first
backup server at 10.1.1.1:
PodlHQ#show ccm-manager
MGCP Domain Name: PodlHQ
Priority Status Host
Primary Backup Ready 10.1.1.2
First Backup Registering with CM 10.1.1.1
Second Backup None
Current active Call Manager:
Backhaul/'Redundant link port:
Failover Interval:
Keepalive Interval;
time: 01:41:10)
time: 00 :00:22 i
Last failover time:
(10.1.1.2)
None
2428
3 0 seconds
15 seconds
10:52:49 UTC Jun 17 2006 (elapsed
12:33:37 UTC Jun 17 2006 (elapsed
12:33:37 UTC Jun 17 2006 from
Troubleshooting Cisco UnifiedCommunications (TVOICE) v8 0 >2010 Cisco Systems, Inc
(10.1.1.1)
Switchback mode:
MGCP Fallback mode:
Last MGCP Fallback start time;
Last MGCP Fallback end time:
MGCP Download Tones:
The command show mgcp endpoint displays any configured endpoints on the Cisco IOS
MGCP gateway. This command is useful when troubleshooting todetermine if anendpoint is
functioning properly. When this command does not display any output, orthe endpoint in
question is missing, it usually points toa configuration error.
Here is an example of output from theshowmgcpendpoint command:
PodlHQJtshow mgcp endpoint
Interface Tl 0/0
ENDPOINT-NAME
SO/dsl-0/l@PodlHQ
S0/dsl-0/2PodlHQ
S0/dsl-0/3SPodlHQ
S0/dsl-0/4@PodlHQ
SO/dsl-0/5PodlHQ
S0/dsl-0/6PodlHQ
SQ/dsl-0/7PodlHQ
S0/dsl-0/8'3PodlHQ
S0/dsl-0/9@PodlHQ
S0/dsl-0/10@PodlHQ
S0/dsl-0/ll@PodlH0
S0/dsl-0/123PodlHQ
S0/dsl-0/13SPodlHQ
SO/dsl-O/143/PodlHQ
S0/dsl-0/15@PodlHQ
S0/dsl-0/163PodlHQ
S0/dsl-0/17SPodlHQ
S0/dsl-Q/18'SPodlHQ
S0/dsl-0/19PodlHQ
S0/dsl-0/20SPodlHQ
S0/dsl-0/21SPodlHQ
S0/dsl-0/2 2PodlHQ
S0/dsl-0/23PodlHQ
Interface Tl 0/1
ENDPOINT-NAME
12:33:07 UTC Jun 17 2006 from
Graceful
Not Selected
None
None
Disabled
V-PORT SIG-TYPE ADMIN
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
0/0:23 none up
V-PORT SIG-TYPE
The command show mgcp connection displays data about any active calls on the configured
andregistered endpoints. In addition, the output provides a legendto assist in decodingthe
output. The following output showsan example of a send-and-receive modein whichthe state
is active, and the codec that is being used is pulse code modulation (PCM) mu-law.
PodlHQftshow mgcp connection
2010 Cisco Systems, Inc. Cisco Unpfied Communications Manager Troubleshooting
Endpoint Call_ID(C) Conn__ID(i;
(E)vent [SIFL] (R)esult[EAj (ME)dia
1. aaln/S2/SU0/0 C=AOO0000002ce9c70000000F5,22,23
S=4,4 CO=l E=2,10,0,2 R=0,0 ME=0
P=19236,32142 M=3
!P)ort (M)ode (S)tate (CO)dec
1 = 0x9
LEGEND:
Mode : 0=INVALID, l=SENDONLY, 2-RECVONLY, 3=SENDRECV, 4-INACTIVE,
5=LOOPBACK, 6=CONTTEST, 7=DATA, 8=NETWLOOP, 9-NETWTEST, 10-CONFRNCE
State : 0=IDLE, 1=SETTING, 2=CONNECTING, 3=CONFERENCING, 4=ACTIVE,
5=CQNF_DESTRGYING, 6=DISCONNECTING, 7=INACTIVE, 8-VOICE_CONNECTING,
9=VOICE_ACTIVE, 10=CONF_DISSOCIATING, 11=CALLLEGS^DISSOCIATED,
12 =-HP__ CONNECTING, 13=HP_CONNECTED, 14-HP__CONFERENCING , 15=HP_ACTIVE,
16=VOIP_CONF_DESTROY, 17=ERR0R, 18=CONNECTING_INACTIVE,
19=CONF_DESTROYING_TNACTIVE, 20=CONT_TEST, 21=SETUP WAIT,
22=WAIT_NSE_SENT, 23=TWC_ACTIVE, 24=WAIT_STATE, 25=HANDOVER
Codec : 1=PCMU, 2-PCMA, 3=G726_32K, 4=G726_24K, 5-G726_16K, 6=G729,
7=G729_A, 8=G729_B, 9=G729_B_LC, 10=G728, 11=G723, 12=G7231_HIGH_RATE,
13=G72 31_A__HIGH_RATE, 14=G7231_LOH_RATE, 15=G7231Â_LOW_RATE,
16=GSM_FR, 17=GSM_HR, 18=GSM_EFR, 19=GSM_EHR, 20=G729_A_B
12=CLEAR_CHANNEL, 129=NSE, 130-XNSE, 131=NTE, 132=T38,
133=MODEM RELAY
Troubleshooting Cisco UnifiedCommunications (TVOICE) v8.0
**
-
MGCP Gateway
(Cont.)
router#
show mgcp statistics
ionitoring Commands
Displays MGCP statistics regarding received and transmitted network
messages
routerS
show mgcp srtp {detail |summary}
* Displays information for active SRTP connections that are controlled
by using MGCP.
router#
debug mgcp [all | errors | endpoint endpointname | media
| nas j packets | parser | src | voipcac]
Enables debugging messages for MGCP errors, endpoints, events,
media, packets, parser, and CAC.
The command show mgcp statistics displays statistics regarding sent and received MGCP
messages. Use this command to determine if connectivity is a problem. To determine if a
failure has occurred, notice the failed statistics for the specific call agent servers and look for
incrementing failed messages. Notice that all of the failed messages are with the call agent at
10.1.1.1. This server may have been down for awhile or may be experiencing heavy load.
PodlHQttshow mgcp statistics
UDP pkts rx 1972, tx 1979
Unrecognized rx pkts 0, MGCP message parsing errors 0
Duplicate MGCP ack tx 0, Invalid versions count 0
CreateConn rx 8, successful 8, failed 0
DeleteConn rx 8, successful 8, failed 0
ModifyConn rx 13, successful 13, failed 0
DeleteConn tx 0, successful 0, failed 0
NotifyRequest rx 47, successful 47, failed 0
AuditConnection rx 0, successful 0, failed 0
AuditEndpoint rx 105, successful 82, failed 23
RestartlnProgress tx 28, successful 28, failed 0
Notify tx 1748, successful 1748, failed 0
ACK tx 158, NACK tx 23
ACK rx 1791, NACK rx 0
IP address based Call Agents statistics:
IP address 10.1.1.1, Total msg rx 53,
successful 29, failed 23
IP address 10.1.1.2, Total msg rx 1919,
successful 1902, failed 0
System resource check is DISABLED. No available statistic
2010 Cisco Systems, Inc. Cisco UnifiedCommunications Manager Troublespooling 2-31
DSO Resource Statistics
Utilization: 0.00 percent
Total channels: 24
Addressable channels: 24
Inuse channels: 0
Disabled channels: 0
Free channels: 24
Ihe privileged HXEX" command show mgcp srtp summary displays an overview of active
Secure Real-Time Transport Protocol (SRTP) connections that are controlled by MGCP. When
you use this command, you are looking for the number of active calls that are using SRTP
(which is highlighted in bold in the example output). This information tells you that SRTP is
working for some calls and to which endpoints it is working. Here is an example of output from
this command:
PodlHQttshow mgcp srtp summary
MGCP SRTP Connection Summary
Endpoint Conn Id Crypto Suite
aalti' S3/SU0/0 8 AES_CM_12 8_HMAC_SHA1_3 2
aaln/S3/SU0/l 9 AES_CM_128_HMAC_SHA1_32
S3/DS1 -0/1 6 AES_CM_12 8_HMAC_SHA1_32
S3/DS1 -0.,' 2 7 AES_CM_12 8_HMAC_SHA1_32
4 SRTP connections active
Use the command show mgcp srtp detail from privileged EXEC mode to obtain more detailed
infonnation about the SRTP connections. Notice that the call to endpoint aaln/S3/SU0/l has
encryption working in the Tx direction but not in the Rx direction (shown in bold).
MGCP SRTP Connect ion Detail foi Endpoint *
Definitions: CS=Crypto Suite, KS=HASHED Master Key/Salt,
SSRC=htcp: =,'www. Cisco, cot ' umve red/cc/td/doc/product/sof twrire/ iosl23/123tcr/ 12
3tvr/Syr.cronization Source, KOC=Roliover Counter, KDR-Key Her;vat ion Rate,
SFQ=Seque:.ee Nurber, FFC-FFC Older,
ML,T=Mas:er Key Lifetime. MKI=Master Key Index:MKI Size
Er.dpoi.-.t aal:v=3,<S'JC'C Call ID 2 Conn ID 8
Ix:CS=AES CM 123 HXAC SKA1_3? KS=3NaOYXS9dLoYDaBHpzRejREfhf.0=-
3SRC =ir_ tp :' '*.ciSfO .com 'urnvercd/cc/td/doc,'product /soft.ware/iosl23/123tcr/ L2
itvr/RandoT RCC= 0 KUR-1 ShQ=.Kandom FEC-FEC ?SRTP MLT=0x80000000 MKT-0:C
Rx:C3-AES_CK_12S_IiMAC_S:iAl_32 KS=21YCQoqxt.xr.dr7ECe+x +DK+G9v4=.
SSHC'-http ://www. cisco .com/univercd/cc/td/doc/product /sof tware/ ios!2 3/123tcr/ !2
3tvr/Randcm ROC-0 KDR= 1 SEQ=Random FEC-FEC->SRTP MLT-Ox80000000 MKI=G:0
Endpoint aain/ S3: S'JO/". Call ID 101 Conn ID 9
Tx:CS-AES_CK 128 HMAC_SHA1_32 KS=11YCQoqxtxtdf7ECe+xrDK+G9v4-
3SRC-htto://www.ciscc,co~/unrvercd/cc/td/doc/product/software/iosl23/123tcr/12
3tvr,Ranaom ROC=0 KDR=I SEQ-Random FEC=FEC-sSRTP MLT=Ux8000000C MKI=0:0
RxrNot Configured
:-Jndt)oinr. 3 3 'OSi -0/1 CalL ID 1 Conn 'u 6
'iX:CS-AFS_CM_I2S_HMAC_3F.Al_3 2 KS=3NaOYXS9dLoYDaBHpzRe;jREfhf.O-
SERC-http://www.crGco.com/univercd/cc/td/doc/product/software/icsl23/l23tcr/]2
3tvr/?.ando~ RCC=C KDH= 1 SEQ-Handom FEC=FEC-sSRTP MLT=0x80000000 MKI^0:C
Rx: CS=AE3_CK_12 8_HMAC_3KA1_32 KS^11 YCQoqxtxtdf 7ECe +x +DK+C,9v4-
SSRC-nttp -'/www.cisco.c^TTi/univercd/cc/td/doc/product/soft ware/iosl 23/12 jtcr/12
JLvr/Kando- HC,"=C KDR=-1 SEQ-RandO!r, I-'EC-FEC- >SRT? MLT=0x8 0000000 MKI = 0:0
iindcci::: 33.'DS. C.'I Call ID 100 Conn ID 7
***
m
w
TX:CS=AES__CM_128_HMAC_SHA1_32 KS=llYCQoqxtxtdf7ECe+x+DK+G9v4=
SSRC-http://www.cisco.com/univercd/cc/td/doc/product/software/iosl23/123tcr/12
3tvr/Random ROC=0 KDR=1 SEQ=Random FEC=FEC->SRTP MLT=Ox80000000 MKI=0:0
Rx:Not Configured
4 SHTP connections displayed
The command debug mgcp and its many variations display debugging information. The
examples show some of the most useful variations of this command. This command is helpful
when you are troubleshootingbecause it allows for detailed debugging down to the packet
level. This command can be especially useful when other troubleshooting techniques have
failed. In the output, notice that the bolded text shows the MGCP message and the device that is
involved in the call setup.
PodlHQ#debug mgcp packets
*Jun 18 00:55:12.858: MGCP Packet received from 10.1.1.2:2427-
i 2010 Cisco Systems, Inc.
This is a call setup message sent to MGCP gateway.
Verify it points to the correct endpoint
(AALN/S2/SU0/0@PodlHQ)
CRCX 406 AALN/S2/SO0/0@PodlHQ MGCP 0.1
AQ0 0000002ce9c75QOOOOQF5
12
p:20, a:PCMU, S:off, t:b8
recvonly
L/hd
L/rg, L/ci(10/17/06/09,1001,)
process,loop
! This is the response to the CRCX
*Jun 18 00:55:12.870: MGCP Packet sent to 10.1.1.2:2427 >
200 406 OK
I : A
The MGCP gateway offers the RTP port number
Check that the IP address in c- field is correct
MGCP gateway IP address.
c=IN IP4 10.1.1.101
m=audio 17740 RTP/AVP 0 100
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194,200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 192-194,200-202
a=X-cap: 2 image udptl t38
2-34
Notice that in thesection wiih NTFY bolded that follows, thelineO: L/hd appears. O: L/hd
represents an observed event (0:) of an off hook(hd) that tookplaceand was detected by the
line package (L). Notice that reason codes are sent in responses. A tabic of reason codes
follows the output. Look up reason code 200 in the table.
*Jun 18 00:55:15.118: MGCP Packet sent to 10,1.1.2:2427 >
NTFY 729336438 aaln/S2/SU0/0iPodlHQ MGCP 0.1
N: ca.ilO.l .1 .2:2427
X: 12
O: L/hd
-Jun 18 00:55:15.122: MGCP Packet received from 10.1.1.2:2427-
- - "5
200 729336438
Jun 18 00:55:15.522: MGCP Packet received from 10.1.1.2:2427-
- - >
! The MGCP gateway received MDCX message from call agent.
! It brings the IP address of another RTP peer and RTP port
! number to set the RTP stream to.
MDCX 407 AALN/S2/SU0/0@PodlHQ MGCP 0.1
A000000002ce9c75000000F5
A
14
p:20, a:PCMU, s:off, t:bS
sendrecv
L/hu, L/hf, D/[0-9ABCD*#]
process,loop
o=- 10 0 IN EPN AALN/S2/SU0/0'iPodlHQ
s=Cisco SDP 0
t = 0 G
m=audio 16384 RTP/AVP 0
c = IN IP4 10.1,2. 201
*Jun 18 00:55:15.526: MGCP Packet sent to 10 .1.1.2:2427-- ->
200 407 OK
<
*Jun 18 00:55:19.674: MGCP Packet sent to 10.1.1.2:2427---?
NTFY 729336439 aaln/S2/SU0/0;PodlHQ MGCP 0.1
N: ca -.10 1 1 2 2427
X: 14
0 : L, hu
Troubleshooting Cisco Unified Communicalions (TVOICE) v8 0 12010 Cisco Systems. Inc
'Jun 18 00:55:19.674: MGCP Packet received from 10.1.1.2:2427-
- ->
200 729336439
<
'Jun 18 00:55:19.674: MGCP Packet received from 10.1.1.2:2427-
- - >
DLCX 408 AALN/S2/SU0/0@PodlHQ MGCP 0.1
C: A000000002ce9c75000000F5
A
15
L/hd
process,loop
-Jun 18 00:55:19.682: MGCP Packet sent to 10.1.1.2:2427---=
2 50 408 OK
P: PS=207, OS=33120, PR=205, OR=32800, PL=0, JI=7, LA=0
<
The variation debug mgcp state displays state changes on MGCP endpoints.
Jun 18 00-57-25.710: //26/3DF5481A8021/MGCP|aaln/S2/SU0/0|-1|-
/<VOICE>/mgcp_set_call_state<7552):[lvl-2]callp(0x641AD808) old
State=CALL_IDLE new state=CALL_VOICE_ACTIVE
-Jun 18 00-57-31.578: //27/3DF5481A8021/MGCP1aaln/S2/SU0/0|-1|-
l/<VOIP>/mgcp_set_call_state(7552):[lvl=2]callp(0x641AD7S8) old
state=CALL_IDLE new state=CALL_CONNECTING
Jun 18 00-57-31.578: //26/3DF5481A8021/MGCP|aaln/S2/SU0/0|-1|-
/<VOICE>/mgcp_set_call_state(7552):[lvl-2]callp(0x641AD808) old
state=CALL_VOICE_ACTIVE new state=CALL_CONFERENCING
-Jun 18 00-57-31.582: //27/3DF5481A8021/MGCP|aaln/S2/SU0/0|-1|-
l/<VOIP>/mgcp_set_call_state(7552):[lvl=2]callp(0x641AD758) old
State=CALL_CONNECTING new state=CALL_CONFERENCING
Jun 18 00-57:31.582: //26/3DF5481A8021/MGCP|aaln/S2/SU0/0|-1|-
/<VOICE>/mgcp_set_call_State(7552):[lvl=2]callp(0x641AD808) old
State=CALL_CONFERENCING new state=CALL_CONFERENCING
/cVOICE,/mgcp_set_call_state(7552):[lvl=2]callp(0x641AD808) old
state=CALL_CONFERENCING new state=CALL_ACTIVE
l/<VOIP>/mgcp_set_call_state(7552):[lvl-2]callp(0x641AD758) old
state=CALL_CONFERENCING new state=CALL_ACTIVE
-Jun 18 00-57-35.194: //26/3DF5481A8021/MGCP[aaln/S2/SUO/0|-1|-
/<VOICE>/mgcp^set^call_state(7552!:[lvl-2]callp(0x641AD808) old
state=CALL_ACTIVE new state=CALL_CONF_DESTROYING
-Jun 16 00-57-35 194: //27/3DF5481A8021/MGCP|aaln/S2/SU0/0|-1|-
l/<V0IP>/mgcp_set_call_state(7552):[lvl=2]callpf0x641AD758) old
state=CALL_ACTIVE new state=CALL_CONF_DESTROYING
-Jun 18 00-57-35 194: //26/3DF5481A8021/MGCP|aaln/S2/SU0/0|-1|-
/<VOICE>/mgcp_set_call^state(7552):[lvl-2]callp(0x641AD808] old
State=CALL CONF DESTROYING new state=CALL_INACTIVE
2010 Cisco Systems, Inc. cisco Unified Communications Manager Troubleshooting 2-35
2-36
-Jun 18 00:57:35.194: //27/3DF5481A8021/MGCP|aaln/S2/SU0/0I -1|-
l,<yOIP>/mgcp_Set_call_state(7552):[lvl,2]callp(0x641AD758) old
State=CALL_CONF_DESTROYING new state=CALL_INACTIVE
-Jun 18 00:57:35.194: //26/3DF5481A8021/MGCP|aaln/S2/SU0/0|-1I -
/</0ICE>/m9cp_set_call_statef7552):[lvl=2]callp(Ox641AD808 old
StatesCALL_INACTIVE new State=CALL_VOICE_ACTIVE
-Jun 18 00:57:35.194: //27/3DF5481A6021/MGCP|aaln/S2/SU0/OI -1I -
l/<VOIP><mgcp_set_call_State<7552>:[lvl=2]callpf0x641AD758) old
=tate-^_ALL_INACTIVE new state=CALL_DISCONNECTING
-Jun IS 00:57:35.198: //27/3DF5481A8021/MGCP|aaln/S2/SUO/0| -1I -
l./<V0IP>/mgcp_seC_call_state(7552) :[lvl=2] callp {0x<i41AD758 old
state=CALL_DISCOHNECTING new state=CALL__IDLE
-Jun 18 00:57:35.198: //-
l/xxxxxxxxxxxx/MGCP/mgcp_set_call_state (7552) :[lvl==2] callp (0x641AD75f
Old State=CALL_IDLE new State=CALL_IDLE
-Jun 18 00:57:37.394: //26/3DF5481A8021/MGCP|aaln/S2/SUO/0I -1I -
/<VOICE>/mgcp_eC_call_state(7552):[lvl=2]callp(0x641AD808) old
state=CALL_VOICE_ACTIVE new state=CALL_DISCONNECTING
-Jun 18 00:57:37.398: //26/3DF5481A8021/MGCP|aaln/S2/SU0/0I -1I -
/<VOICE>./mgcp_Set_call_state(7552) :[lvl=2]callp (Ox641AD808) old
state=CALL_DISCONNECTING new state=CALL_IDLE
-Jun 18 00:57:37.398: /./- 1/xxxxxxxxxxxx/MGCP|aaln/S2/SUO/0 I-1I-
l/mgcp_set_call_state(7552):[lvl=2]callp(0x641AD808) old
state=CALL_IDLE new state=CALL_IDLE
"Jun 18 00:57:37,398: //-
i/xxxxxxxxxxxx/MGCP/mgcp_set^call_State(7552):[lvl=2]callP(0x641AD758
old State=CALL_IDLE new state=CALL_IDLE
This rable lists the return codes that can be seen in the MGCP messages.
Return Codes
Code
Description
(Oxx) Response acknowledgments:
The transaction is currently being executed. An actual completion message
will follow.
(1xx) Provisional response:
100
The transaction is currently being executed. An actual completion message will follow.
101
The transaction has been queued for execution. An actual completion message
follow
(2xx) Successful completion:
200
The requested transaction was executed normally
250
The connection was deleted.
(4xx) Transient error:
400
401
402
403
404
The transaction could not be executed, due to a transient error.
The phone is already off-hook.
The phone is already on-hook.
The transaction could not beexecuted, because the endpoint does not have sufficient
resources at this time.
Insufficient bandwidth at this time.
Code Description
405 The transaction could not be executed, because the endpoint is "restarting."
406
Transaction time-out. The transaction did not complete in a reasonable time and has
been aborted,
407
Transaction aborted. The transaction was aborted by some external action, such as a
ModifyConnection command aborted by a DeleteConnection command
409 The transaction could not be executed because of internal overload.
410
No endpoint available. A valid "any or wildcard was used, but there was no endpoint
available to satisfy the request.
(5xx) Permanent error:
500 The transaction could not be executed, because the endpoint is unknown.
501 The transaction could not be executed, because the endpoint is not ready.
502
The transaction could not be executed, because the endpoint does not have sufficient
resources.
503 "All or wildcard too complicated.
504 Unknown or unsupported command.
505
Unsupported RemoteConnectionDescriptor. This code should be used when one or
more mandatory parameters or values in the RemoteConnectionDescriptor are not
supported.
506
Unable to satisfy both LocalConnectionOptions and RemoteConnectionDescriptor.
This code should be used when the LocalConnectionOptions and
RemoteConnectionDescriptor contain one or more mandatory parameters or values
that conflict with each other or cannot be supported at the same time (except for codec
negotiation failure; see error code 534).
507
Unsupported functionality. Some unspecified functionality that is required to carry out
the command is not supported. Note that several other error codes have been defined
for specific areas of unsupported functionality (for example, 508, 511, etc.), and this
error code should only be used if there is no other more specific error code for the
unsupported functionality.
508 Unknown or unsupported quarantine handling.
509
Error in RemoteConnectionDescriptor. This should be used when there is a syntax or
semantic error in the RemoteConnectionDescriptor.
510
The transaction could not be executed, because some unspecified protocol error was
detected. Automatic recovery from such an error will be very difficult, and therefore,
this code should only be used as a last resort.
511
The transaction could not be executed, because the command contained an
unrecognized extension. This code should be used for unsupported critical parameter
extensions ("X+").
512
The transaction could not be executed, because the gateway is not equipped to detect
one of the requested events.
513
The transaction could not be executed, because the gateway is not equipped to
generate one of the requested signals.
514
The transaction could not be executed, because the gateway cannot send the
specified announcement.
515 The transaction refers to an incorrect connection-ID (may have been already deleted).
516 The transaction refers to an unknown call-ID.
517 Unsupported or invalid mode.
>2010 Cisco Systems, Inc.
Cisco Unified Communications Manager Troubleshooling
Code Description
518 Unsupported or unknown package.
519 Endpoint does not have a digit map.
520 The transaction could not be executed, because the endpoint is "restarting."
521 Endpoint is redirected to another Call Agent.
522 No such event or signal.
523 Unknown action or illegal combination of actions.
524 Internal inconsistency in LocalConnectionOptions.
525 Unknown extension in LocalConnectionOptions.
526 Insufficient bandwidth.
527 Missing RemoteConnectionDescriptor.
528 Incompatible protocol version.
529 Internal hardware failure
530 Channel associated signaling (CAS) protocol error.
531 Failure of a grouping of trunks (for example, facility failure).
533 Response too large.
534 Codec negotiation failure.
535 Packetization period is nol supported.
536 Unknown or unsupported RestartMethod.
537 Unknown or unsupported digit map extension.
538
Event or signal parameter error (for example, missing, erroneous, unsupported,
unknown, etc.).
539
Invalid or unsupported command parameter. This code should only be used when the
parameter is neither a package nor a vendor extension parameter
540 Per-endpoint connection limit exceeded.
541
Invalid or unsupported LocalConnectionOptions. This code should only be used when
the LocalConnectionOptions is neither a package nor a vendor extension.
(8xx Package-specific response codes:
800 Invalid NextEndpointName.
801 Invalid StartEndpomtName.
802 Invalid or unsupported BulkRequestinto parameter.
803 Invalid or unsupported StateType
804 Bulk Audit Type not supported.
805 Incorrectly specified endpoint range.
806 Requested StartEndpoint unknown or unavailable.
The gateway uses reason codes when deleting a connection to informthe call agent about the
reason for deleting the connection. Reason codes can also be used in a RestartlnProgress
command to inform the gateway of the reason for the restart.
Troubleshooting Cisco Unified Communicalions (TVOICE) v8 0
Reason Codes (RFC 2705)
Code Description
0
Endpoint state is nominal. (This codeis usedonly inresponse toaudit requests.).
900
Endpoint malfunctioning.
901 Endpoint is taken out of service.
902
Loss of lower layer connectivity (for example, downstream sync).
) 2010 Cisco Systems, Inc. Cisco Unilied Communica tions Mana ger Troubl e sriooti ng 2-39
Cisco Unified Communications Manager Event Log
Event log messages can provide clues to the cause ofanunregistration event that occurs after a
successful registration.
2-40
^fk.
nihed oomiiHimcduokt!
Iana<
Enum definitions for Reason Code
ttWMMIK
Sonny
PnCrHE
rctti-\
it.*f.mn.iri.ai.n.
H.'tSlîdlHT
JT
Unknown
NoEntiyi nDat a ba se
DalabaseConfi guralionrror
DeviceNarreUnresolveable
MaxD e vReg Exceeded
CooneclivilyErrof
InitialisationError
DovicelnilialedReset
CaflManagerResei
DeviceUntegislered
Enum definitions for Device Type
120
121
125
254
255
MODEl_MGCP_STATION
MODEL_MGCP TRUNK
MODEL_TRUNK
MODEL_UNKNOWN_MGCP_GATEWAY
MODEL UNKNOWN
Thekey to understanding these messages is to understand howto decodethedevicetypeand
reason codes.
Hereis an example of an error message that mightbe displayed whenan unregistration event
occurs:
%CCM_CALLMAHAGER-CALLMANAGER-3-DeviceUnregistered: Device
unregistered. Device name. AALN/S2/SU0/0fflPodlHQ Device IP
address.10.1.1.101 Protocol.MGCP Device type. [Optional]120
Device description [Optional].AALN/S2/SU0/0w>PodlHQ Reason Code
[Optional]9 App IDCisco CallManager Cluster
IDStandAloneCluster Node ID CCM2-1
This error message occurred because a device that has previously registered with Cisco Unitied
Communications Manager has unregistered. This error message can be issuedas part of a
normal unregistration event, or because of someother reasonsuchas the loss of kecpalives.
This error message references a device type of 120, which refers to an MGCP station, and a
reason code of 9. which means that a CiscoUnified Communications Manager reset message
was sent, which caused the device to imregister.
The tables list other device type values and reasoncodesthat the unregistration event message
miaht reference.
Device Type (Optional)
Value Definition
1 MODEL_30SPP
2 MODEL_12SPP
Troubleshooting Cisco Unified Communicalions (TVOICE) v8 0 ) 2010 Cisco Systems, Inc
Value
10
21
30
40
41
42
43
47
50
51
61
62
70
71
72
73
90
100
110
111
120
121
122
124
125
254
255
2010Cisco Systems, Inc.
Definition
MODELJ2SP
MODEL 30VIP
MODEL 30VIP
MODEL_TELECASTER_BID
MODEL_TELECASTER MGR
MODEL_TELECASTER_BUSINESS
MODEL_IP_CONFERENCE PHONE
MODEL VGC PHONE
MODEL^STATION_PHONE_APPLICATION
MODEL VEGA
MODEL TITAN1
MODEL_DIGITAL ACCESS T1
MODEL TITAN2
MODEL LENNON
MODEL ELVIS
MODEL_CONF BRIDGE
MODEL_YOKO_CONF BRIDGE
MODEL_H323 PHONE
MODEL_H323_GATEWAY
MODEL__MUSIC_ON_HOLD
MODEL_DEVICE PILOT
MODEL CTI PORT
MODEL_CTI_ROUTE POINT
VoicelnBox
MODEL_ROUTE LIST
MODEL_LOAD_SIMULATOR
MODEL MTP
MODEL_YOKOJv!TP
MODEL_MGCP_STATION
MODEL_MGCP TRUNK
MODEL_GATEKEEPER
MODEL_14_BUTTON SIDECAR
MODEL TRUNK
MODEL_UNKNOWNJVIGCP_GATEWAY
MODELÛNKNOWN
2-42
Reason Code (Optional)
Value
Definition
Unknown
NoEntrylnDatabase
DatabaseConfigurationError
DeviceNameUnresolveable
MaxDevRegExceeded
ConnectivityError
InitializationError
DevicelnitiatedReset
CallManagerReset
1(
DeviceUnregistered
For some .ample debug outputs, refer to the Sample ofDebug MGCP Packets document at the
following link: http. ci>eo.com en IS products'sw voices* 'ps5Vv'
product^ lech uoie0^l*6iilK.KOI74K04.sliiml
Troubleshooting Cisco Unified Communications (TVOICEl v8.0
Cisco IOS MGCP Gateway Unsuccessful Registration
This section describes Cisco Unified Communications Manager debug output for Cisco IOS
MGCP gateway unsuccessful registration.
m^^m^mm^^^^mmssm^^^^mmm^Kimmm^94iH^'^^^'gmm
Cisco IOS MGCP Gateway
Unsuccessful Registration
HQ-lidebug ccm-manager events
Jun 2 12:59:15.475: cmapp ngr process evactlvehost failed: Active
host 0 110.1.1.1) failed
Jun 2 12:59:115.415; onmpp ngr check hoetlist: xotive host is 0
(10.1.1.1)
Jun 2 12:59 ;15.475i capp try fallbac)t(Be tto mode.ON)
Jud 2 12:59:15.475: csapp shut backhaul: backhaul link shutdown is not
configured
Jun 2 12:59zl5.475: cmapp try fallback: fallback is not configured
Jun 2 12:59:15.475: cmapp host fen: Processing event
REGISTRATION HEEDED lor host 0 (10.1.1.1) in State REGISTERING
Jun 2 12:5* :15.475: cmappaacpsendrslp: ip addr=10.1.1.1 port-2427
it typa--l. Blot*0,aubuoit-0 rst_type-3
Jun 2 12:59:15.475: cmapp start host bar: Host 0 (10.1.1.1). tmr 0,
duration 30000
Jun 2 12:59:45.476: cmapp mgrprocaasevactivehoet failed; Active
host 0 (10.1.1.1) failed
Jun 2 12:59:45.176: cmapp mgr check hostlist: Active host Is 0
[10.1.1.1)
Jun 2 12:59:45.476: cmapp try fallback (set to mod*-ON)
Jun 2 12:59:45.476: cmapp shut backhaul: backhaul link shutdown is not
configured
Jun 2 12:59:45.476: cmapptry fallback: fallback is not configured
The figure shows Cisco IOS MGCP gateway trying to register with Cisco Unified
Communications Manager with the IP address 10.1.1.1.
The debug ccm-manager events command shows that the Cisco IOS MGCP gateway
repeatedly attempts to register but the registration attempt fails and the state REGISTERING
never transfers to REGISTERED. In this case, the show ccm-manager command would also
display the same state:
HQ-l#show ccm-manager
MGCP Domain Name: HQ1
Priority Status Host
Primary
First Backup
Second Backup
Registering with CM
Hone
Hone
Current active Call Manager:
Backhaul/Redundant 1 ink port:
Failover Interval:
Keepalive Interval:
00:04:01)
00:00:271
Last failover time:
Switchback mode:
MGCP Fallback mode:
Last MGCP Fallback start time:
TFTP retry count to shut Ports:
None
2428
3 0 seconds
15 seconds
13:53:41 CET Jun 2 2010 {elapsed time:
13:57:15 CET Jun 2 2010 (elapsed time:
None
None
Graceful
Not Selected
None
None
Disabled
2
Backhaul Link info:
Link Protocol: TCP
Remote Port Number: 2428
Remote IP Address: 13.1.1.
Current Lir.k St ate : DPEN
Stat.: sties :
p-i-ket s rccid : 0
Re::v fa i ". ures : 0
Packets xmitted: 0
Xvn: t tailzies: Z
PRI Port s being backhaulea:
FAX T,-ac: CISCO
Coni icurat icr. Error History :
The reason for the issue is either incorrect IP address that is configured at Cisco IOS MGCP
gateway, or incorrect hostname that does not match with Cisco Unified Communications
Manager configuration (which is the ease in this example).
2-44 Troubleshooting Cisco Unified Communications (TVOICE) v8.0 CO 2010 Cisco Systems, Inc
H.323 and SIP Gateway Communications
This topic explains issues with H.323 and SIP gateway communication in a Cisco Unified
Communications system.
Cisco !OS H.323 and SIP Gateway
Communications
H.323 and SIP gateway communications behavior:
* H.323 and SIP gateways do not register to Cisco Unified
Cisco Unified Communications Manager shows their status
always as Unknown.
* Cisco Unified Communications Manager requires
destinations reached through these gateways to be included
in the dial plan.
* The H.323 and SIP gateways require their dial plan to be
configured locallyas apart of their Cisco IOS configuration.
* Call setup issues will be covered in detail in the
"Troubleshoot Call Setup Issues" module.
H.323 and SIPgateways do not registerto CiscoUnifiedCommunications Manager servers
because of the peer-to-peer nature of the protocol. The Cisco Unified Communications
Manager Administrationweb interfacewill list H.323 and SIP gateways (SIP trunks in case of
SIP) with a status of "Unknown," because a gateway that runs either SIP or H.323 will be a
peer to the corresponding gateways (or trunkin case of SIP) that are configured on Cisco
Unified Communications Manager. The SIP or H.323 gateways and Cisco Unified
Communications Manager gateways are configured independently from each other, which has
the effect of decentralizing thedial plan. Dial peers containthedial planconfiguration on the
gateway. TheCisco Unified Communications Managergatewaydial plan is constructed by
usingroute patterns, route lists, androute groups. All destinations that go to an H.323or SIP
gateway from CiscoUnified Communications Managerwill need to be addedto the dial planof
For Cisco Unified Communications Managerto accept calls froman H.323gateway, the IP
address of that gateway must be a known device in the Cisco Unified Communications
Manager configuration. Add a gateway into the configuration of the Cisco Unified
Communications Manager. Because of this default behavior, it is recommended that youseton
the H.323 gateway which IP address should be used as the source IP address for H.323
messages. Use the command h323-gateway voip bind srcaddr <1Paddress>. This command
ensures that the address that is used to source 11.323 messages is consistent and that Cisco
Unified Communications Managerwill not reject call setupmessages because of an unknown
IP address.
Call setup issues with H.323 and SIP gateways are covered in more detail in the module
"Troubleshoot Call Setup Issues."
Major H.323 and SIP Gateway Monitoring Commands
This figure summarizes the show and debug commands that are available on Cisco IOS
Software to monitor the status of 11.323 and SIP gateways.
Major H.IJ.Z
Monitorina (
Dial peer monitoring:
' show dial-peer voice
* debug voip dialpeer
H.323 gateway monitoring:
- show gateway
- debug h225 asnl | events
* debug h225 q931
SIP gateway monitoring:
- show sip-ua calls | connections | statistics | status
' show sip service
The first set of commands is independent of gateway type, and these commands are the same
on both H.323 and SIP gateways.
The command show dial-peer voice is used to display information for voice dial peers. Use
this command to display the configuration for all VoIP and POTS dial peers that are configured
for a router. To show configurationinformation for only one specific dial peer, use the tag
argument to identifythe dial peer. To showsummary information for all dial peers, use the
summary keyword.
The command debug voip dialpeer is used to view default VoIP dial peer information.
The command show gateway is commonly used to monitor the status of 11.323 gateway
registration to H.323 gatekeeper.
Use the command debug h225 asnl to display additional informationabout the actual contents
of 11.225 Registration. Admission, and Status Protocol (RAS) messages. The command
displays the Abstract Syntax Notation One(ASN.l) contents of any 11.225 message that is sent
or received that contains ASN.l content. Not all H.225 messages contain ASN.l content. Some
messages contain bothQ.931 infonnation and ASN.l information, and if you enterthis
command, only ASN.l information will be displayed.
Use the debug h225 q931 command to display Q.931 informationclement details of 11.245
messages. Thecommand displays keyQ.931 eventslhat occur whenan 11.323 call goes from
one gateway to another. Q.931 events are carried in H.225 messages. This command enables
you to monitor Q.931 state changes such as setup, alert, connected, and released.
Use the debug h245 events command to display 11.245 events.
ruijljitssl.ooring Cisco UnifiedCommunicalions (TVOICE) v8 0
The command show sip-ua calls is used to display active user agent client (UAC) and user
agent server (UAS) information on SIP calls. This includes information about multiple media
streams, up to three media streams if it is a'media-forked call. This command is useful in
debugging multiple media streams because it is the only command that indicates whether an
active call is forked.
Use the show sip service command to display the status of SIP call service on a SIP gateway.
Summary
imai
The steps that take place during phone bootup, when issues could
potentially exist, include PoE, DHCP address assignment, and TFTP
configuration download.
Use ping to verify network connectivity and DHCP configuration
check to verify that correct TFTP address is distributed to endpoints.
Use Status, Network, and Device Configuration on the Cisco Unified
IP Phone to verify that the phone obtained correct information.
MGCP gateway registration issues include misconfiguration, network
connectivity or addressing issues, or systems incompatibility.
MGCP gateway registration status can be verified from a Cisco
Unified Communications Manager server or the Cisco IOS gateway
platform.
H.323 and SIP gateways never register with the Cisco Unified
Communications Manager, and their status is checked solely from
the Cisco IOS gateway platform.
This lesson has defined the issues that are related to gateway and endpoint registration in Cisco
Unitied Communications Manager and described the common solutions to these issues.
References
Cisco Systems. Inc. Sampleof Debug MGCPPackets, April 2008.
hr.p: u-.cu.aim en US piodticts s\\ \oiees\v ps556
product:, tech nutcfWIKbndilKOI 74Nl)4.^htii.l
Cisco Systems, Inc. Troubleshooting Guide for Cisco Unified Communications Manager.
Release S.Oil), Device Issues. February 2010
htt|v wu w.ciscu.ajii] en US partner docs voice ip coinm.'cueiit trouble
S 0 1 sbdevicehtm!
Troubleshooting Cisco Untied Communications (TVOICE)v8 0
Lesson 2
Availability Issues
Overview
This lesson describes scenarios in which the Cisco Unified Communications Manager is not
available and describes how to troubleshoot and diagnose the cause of these problems. These
scenarios include the Cisco Unified Communications Manager being unavailable to the
administrator through the web-based interface or phones, gateways being unable to
communicate withthe required CiscoUnifiedCommunications Managerservices,and a Cisco
Unified Communications Manager server that is abnormally slow.
Objectives
Uponcompleting this lesson, youwill be able to explain the issuesthat cancause Cisco Unified
Communications Managerto becomeunavailable in the networkand describe howto isolate
and troubleshoot these issues. This ability includes being able to meet these objectives:
Describe the possible causesand recommended actionsto take when the CiscoUnified
Communications Manager system stops responding
Describe the possible causes and recommended actions to take when the Cisco Unified
Communications Manager Administration web page does not display
Describe the possible causes and recommended actions to take when the Cisco Unified
Communications Manager server response is slow
Cisco Unified Communications Manager System
Stops Responding
This topic describes the possible causes and recommendedactions to take when the Cisco
Unified Communications Manager system stops responding.
2-50
Problem Report:
* Endpoints fail to register or communicate to their first choice
Primary UnifiedCM" Secondary UnifiedCM"
/
I
MGCP Gateway
*United CM=Cisco UnifiedCommunicat.ons Manager
The following issue ina Cisco Unified Communications Manager cluster canbereported: The
primary server that is configured as the preferred system in the Cisco UnifiedCommunications
Manager Groupdoes not respondto endpoint registration requests and keepalivcs. At this time,
when devices such as the Cisco Unified IP Phones and gateways unrcgister fromthe Cisco
Unified Communications Manager, users receive a delayeddial tone, and the Cisco Unified
Communications Manager server freezes because of high CPUusage.
Troubleshooting Cisco Unified Communications (TVOICEl v8 0
>2010 Cisco Systems. Inc
Major Causes of Not Responding
Analyze the following major facts that coujd cause Cisco Unified Communications Manager
not to respond.
ior Causes of
Analyze the facts to determine the most likely source of
the problem:
The system logs might show if a server crash occurred and might
provide information on the source of the problem.
The system logs might provide information on why the service
hung or failed to start. Often, a reboot or manually starting the
service can fix the problem.
CPU usage could be the problem and, based on the topology and
size of the deployment, might be determined to be unusual. A
reboot of the affected server can often solve this issue.
Aprocess could be causing a memory leak, which could be due
to an error in the application or operating system. A reboot would
be recommended.
These are the most likely sources of the problem:
The Cisco Unified Communications Manager server could have crashed, causing the server
to reboot. After the reboot, the system logs might show more details.
The Cisco CallManager service could freeze or fail to start completely. You can use the
system logs to determine if this is the cause of the problem. Also, the Cisco TFTP service is
important for registration of endpoints. Many endpoints download their configuration from
the Cisco TFTP server that is running on a Cisco Unified Communications Manager server.
Check the status of the Cisco CallManager and Cisco TFTP services.
A process in Cisco Unified Communications Manager could be consuming too much of the
CPU cycles, or the server could be heavily overused because of an ignored cluster upgrade.
Check the CPU usage at the Cisco Unified Communications Manager server.
A process with a memory leak could cause a shortage of system memory. This memory
leak will cause the server to page to the hard drive and either delay or stop responding.
Check the memory usage of the server processes.
>2010Cisco Systems, Inc Cisco UnifiedCommunicalions Manager Troubleshooting 2-51
Cisco Unified Communications Manager System Log
This section explains how to display the system event log.
2-52
co Unififcti 0.iiWi.unicdti(jr,:.
*"." '
.-.. .
nnrN*-)ij
::: .,:.;
*
Because Cisco Unified Communications Manager is an appliance that is based on Linux and no
full shell access exists, the Cisco Unified Communications Manager Real-Time Monitoring
Tool (RTMT) is the only way to view the system event log from the GUI.
Note In the limited command-line interface (CLI) that does exist, you can use the file command to
display log files.
It is useful to view log files when you troubleshoot server crashes, hung services, and other
errors. If server crashes are a recurring problem, increasing the level of trace on these logs can
provide more detail that can help the Cisco Technical Assistance Center (TAC) determine the
source of the problem.
Start the Ci>,co Unified Communications Manager RTMT and click SysLog Viewer. Then
choose the server in the Logs tab, expand the System Logs, and click Messages to display the
current system logs.
When the Ci-.co CallManager service stops responding, the following message displays in the
system event log:
The Cisco CallManager service terminated unexpectedly. It has
done this 1 time. The following corrective action will be
taken in 60000 ms. Restart the service.
You might see another message in this situation:
Timeout 3000 milliseconds waiting for Cisco CallManager
service to connect.
If Cisco Unified Communications Manager failed to start because of the following error, you
might see this:
The service did not respond to the start or control request in
a timely fashion.
Troubleshooting Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems. Inc
m
M
Server Utilization
Use the Cisco Unified Communications Manager RTMT to view the CPU and memory usaec
onthe server and the CPU usage ofthe processes.
Server Utilization
0 -. *-.
1 ***..; *<*n
*n fct*fl \tma i *'u
r, J_ ttocmi j *-?
The Cisco CallManager service can stop responding because the service does not have enoush
resources, such as CPU or memory, to function. Generally, the CPU usage in the server could
be close to 100 percent at that time.
In Cisco Unified RTMT, click CPU and Memory on the left of the screen. CPU and memory
usage appear on the right in the form of tables and graphs. Wait afew minutes for the graphs to
populate with real-time counters.
You can also view the average CPU usage from the CLI on the server by using these
commands:
show stats io
show perf query class Processor
You can view the CPU usage ofthe processes by using these commands:
showperf query counter Process "% CPUTime"
show process load
The following example output from the show stats io command shows an average idle CPU
usage of about ]3percent (shaded text). This means that the CPU usage is approximately 87
percent. You should create abaseline during normal operation and compare the current CPU
usage with the baseline to determine ifit is abnormal. If it is abnormal, you can use the
command show perf query counter Process "% CPU Time" to determine which process is
causing the problem.
admin:show stats io
avg-cpu: %user %nice
%sys %iowait %idle
2-53
Device:
hdb
hdbl
hdb2
hdb 3
hdb4
hdb5
hdb6
76.54 0.03
tps
22 .81
6 . 95
0 . 00
0.00
0 . 00
0 . 94
14 . 92
8.74 1.53 13.15
Blk read/s Blk_wrtn/s
18 . 13
12 .28
0.00
0 . 00
0.00
5.43
0 .40
420.25
119.69
0 .00
0.00
0 . 00
15.67
284.89
Blk_read
1638764
1110034
282
218
4
490472
36338
Blk_wrtr.
37985834
10818416
32
18
0
1416280
25751088
The follow ing example output from the show perf query counter Process / CPU Time
command displavs the CPU utilization per process, which can help you isolate the source ot
high CPU utilization. Look for aprocess that seems to be using alarge percentage ot the CPL.
Notice that the jaâ#i process is consuming 53 percent of the CPU time, which could indicate
that it is amisbehaving process. Compare this to your baseline to determine it it is abnormal,
admin:show perf query counter Process % CPU Time-
- Perf class Process!^ CPU Time) has values:
CiscoDRFLocal -> % CPU Time =
CiscoDRFMaster -> % CPU Time =
CiscoLicenseMgr -> % CPU Time =
CiscoSyslogSubA -> % CPU Time =
'Output omitted)
> % CPU Time =
> % CPU Time =
> % CPU Time =
> % CPU Time = Q
> % CPU Time =" 53
> \ CPU Time =
> % CPU Time =
> % CPU Time =
> % CPU Time =
> % CPU Time =
host_agent.pi
hostagt
init
ipsec_mgr
java#l
java#2
java
kapnid
keventd
khubd
(Output omitted)
The show perf querv class Processor command displays the average CPU usage. This
command ,s another wav to determine if high CPU utilization is the source ot the Cisco Unitied
Communications Manager issues. This example shows aCPU time of 78 percent (shaded). This
number is the likely source of problems.
admin:show perf query class Processor
==>query class :
- Perf class (Processor) has instances and values:
Total
- > % CPU Time
-> iOwait Percentage
> idle Percentage
-> Irq Percentage
-> Nice Percentage
-> Softirq Percentage
-> System Percentage
-> User Percentage
-> % CPU Time
= 7i
22
0
4
0
12
5
21
m>
jm
'*
jTotal
_Total
_Total
_Total
_Total
_Total
Total
-? iOwait Percentage
-> Idle Percentage
-> Irq Percentage
-> Nice Percentage
-> Softirq Percentage
-> System Percentage
-> User Percentage
= 0
= 79
= 0
= 4
= 0
= 12
= 5
Output from the show process load command is another way to view the CPU utilization.
Notice that in this output, the user CPU utilization is 55 percent. Compare this to the baseline to
determine if it is normal.
admin:ahow process load
01:26:26 up 1 day, 1:03, 2 users, load average: 0.65, 0.65,
110 processes: 109 sleeping, 1 running, 0 zombie, 0 stopped
CPU states: cpu user nice
total 55.4* 0.0*
Mem: 76892Bk av, 761944k used,
574156k actv,
501940k used,
0.78
idle
i 35.4%
54748k buff
232520k
Swap: 2047772k av,
cached
PID USER
2958 Informix
3457 database
27 688 admin
27727 admin
processOperat
27728 admin
1 root
2 root
3 root
4 root
root 7
PRI
16
15
28
35
34
15
15
15
34
25
HI
0
0
10
10
system irq
9;i* 0.9%
6984k free,
109128k ind,
1545832k free
softirq iowait
0.0% 0.0^
0k shrd,
13420k in c
SIZE RSS SHARE STAT %CPU %MEM TIME CPU COMMAND
13:02
3:52
0:01
0:00
0:00
0:04
0:00
0:02
0 : 00
0:00
0 oninit
0 dbmon
0 Java
108M 107M
13244 11M
18956
1328
1428
700
18M
1328
1428
596
5.4
5.4
0.9
14.3
1.4
2.4
0.1
0.1
0.0
0.0
0.0
0.0
0.0
19
0
106M S
6524 S
7600 S N
1036 S N
964 R N
524 S
0 sw
0 sw
0 SWN
0 SW
0.9
0.0
0 .0
0.0
0.0
0.0
0 top
0 init
0 keventd
0 kapmd
0 ksoftirqd/0
0 bdflush
Note You canalso usethesecommands tounderstand and create a baseline, wtiich helps you
determine if thesystem isbehaving abnormally.
>201G Cisco Systems, inc
Cisco Unified Communications Manager Services
This section describes howto check if core services are
running
t.&Jk ', (\ hfiH Mh ",ht'i.
CiscoUnified Serviceability> Tools > Control Center- Feature
Services
Cisco Unified Serviceability^ Tools > Service Activation
To \ iew the siatus ofCisco CallManager service and Cisco TFTP service, use Control Center
Feature Services from the Cisco Unitied Serviceability menus. Access this by choosing Tools >
Control Center- Feature Services.
It the services arereported as not running, restart them from within the Control Center -
Feature Services page and observe the system event log if the restart issuccessful.
Occasionally, the reactivation ofservices in Tools >Service Activation can help iftheir restart
fails.
Troubleshoolmg Cisco Unified Communicalions (TVOICE) v8 0
mw
Administration Does Not Display
This topic describes the possible causes and recommended actions to take when the Cisco
Unified Communications Manager Administration web page does not display.
Administration Page Does Not Display
Problem Report:
Administrator fails in an attempt to loginto
https://<cucm>/ccmadmin Cisco Unified Communications
Manager Administrator page.
Error or warning messagesaredisplayed instead.
Cisco Unified
Communications
Manager
The following problem report can be formulated:
An administrator tries to connect to the web-based Cisco Unified Communications Manager
Administration interface by using the URL https://<cucm >/ccmadmin and receives one of the
following error messages, depending on the web browser that is used:
Internet Explorer: "The page cannot be displayed."
Netscape: "There was no response. The server could be down or is not responding."
) 2010 Cisco Systems. Inc.
CiscoUnified Communications ManagerTroubleshooting
2-57
Major Causes of Not Displaying the Administration Page
This section describes the major facts that could cause the adminis
2-58
stration patje not to display.
"4 -Jl'
H j.U/iimj
Analyze the facts to determine the most likely source of
the problem:
s Clear out web browser cache.
- Testthe network connectivity.
' TesttheDNS nameresolution, ifused.
<Verify that a Cisco Tomcat service is running on the Cisco Unified
Communications Manager Publisher.
Verify that HTTPS traffic is permitted and that a firewall or router
access list exists.
CPU usage might be high, causing slow response to HTTPS.
The<-e are possible cause*of the issue:
First. tr\ to clear the web browser cache.
Network connectivity may not exist because of areconfiguration or network issues Test
the network connectivity.
Name resolution ,s optional, but if used, it might not be configured, ii is resolving to an
incorrect address, or ,tdoes not resolve at all. Test the name resolution.
ACisco Tomcat service is not running; it has failed to start or it has stopped This
represents the most frequent reason for Cisco Unified Communications Manager
Administration not displaying.
Firewall or access lists may be blocking the HTTPS traffic in one or both directions Make
sure that ,t has been permitted between your workstation and the server that you are irvinu
to reach. - -
' f PL'utl'izatlon ma>' bc h'?h th<-' server, leaving only very limited resources for the GUI
\ enry the server utilization.
'2010 Cisco Systems.
IM
Testing Network Connectivity
Make sure that the network connectivity exists toward the server.
Use the ping or trace-route (tracert) command from your workstation Use the server IP
address to test and verify network connectivity. Fix any network problems that could prevent
connectivity.
Use the ping or trace-route (tracert) command. Use the devicename to verifythat Domain
Name System (DNS) nameresolution is takingplaceproperly. If you encounter any problems,
determine the cause of the name resolution failure and address the issue.
If youare notusingDNS, yourlocal workstation will check in the "hosts" file tosec if anentry
exists for theyour-cucm-server-natne and if an IPaddress exists that is associatedto it. Open
the file and add the Cisco Unified Communications Manager server name and the IP address. If
you are using Microsoft Windows at your workstation, you can find the "hosts" file at
C:\WINDOvVS\system32\drivers\etc\hosts.
If testingnetwork connectivity by using ping succeeded, a firewall or router couldstill be
blocking HTTPS traffic. Remove the impediment if the securitypolicyallows.
CiscoUnified Communicalions ManagerTroubleshooting 2-59
Verifying the Cisco Tomcat Service
2-60
This section describes how to check that the Cisco Tomcat service is running.
admir, zutils service list
Requesting service status, please wait...
System SSH [STARTED]
Cluster Manager [STARTED!
Service Manager is running
Setting list o all services
.1 Return code = 0
A OlSCO DBISTARTEDJ
(output omitted)
Cisco Tc*cttSTOPPEE] ComBanded Out. of Service
Cisco Tomcat Stats Servlet [STOPPED] Commanded Out of Service
(output omitted1
admin,utils service start Cisco Tomcat
CiSCC Tomcat (STARTING]
CiSCC Tomcat [STARTING]
Cisco Tomcat [STARTING]
(may take several minutes to complete!
Ciacc TomcBttSTABTIHGl
CiBCO roncatlSTARTEDl
Verify that theCiscoTomcat service is running andhas not stoppedor ceased responding. Use
the CLI command utils service list to view the status of all Cisco Unified Communications
Manager services, including Cisco Tomcat.
Stan theCiscoTomcat process if it has stopped; restart the server if it is not responding and
cannot he staned. Use the CLI command utils service start Cisco Tomcat to start the service
and wait until the sen ice reports STARTED.
Troubleshooting Cisco Unified Communications(TVOICE) v8 0
Slow Server Response
This topic describes the possible causes and recommended actions to take when the Cisco
Unified Communications Manager server response is slow.
Slow Server Response
Problem Report:
Users complaining about delays, such as postdial delay.
" Users attempting to reach the web interface complain about
extremely slow response.
Cisco Unified
Perceptible delays
The following problemreport has been formulated:
Users at one site are complaining of postdial delay. When an administrator tries to connect to
theCisco Unified Communications Manager web-based interface, theresponse is extremely
slow.
Cisco Unified Communicalions Manager Troubleshooting 2-61
Major Causes of Slow Server Response
This section explains why a server could respond slowly.
Analyze the facts to determine the most likely source
of the problem:
3 Check the port settings on a switch forduplex and speed.
Check the server Ethernet settings for duplex and speed.
a Consider an upgrade if CPU or memory usage is high.
' Check for an incorrect dial plan design.
These arc the potential causes of slow server response:
Speed and duplex mismatches between the server and switch can cause droppedpackets
due to collisions. This can result in slow response time for the server. These issues are not
isolated to the web interfaces, but this might be one symptom of the problem. Verity if the
port on the switch that connects to the server is set for duplex and maximumspeed.
Verify that Ethernet settings on the Cisco Unitied Communications Manager server are set
for duplex and maximum speed.
Check if high CPUusage, a malfunctioningprocess, or a memory leak is not experienced ai
ihe Cisco Unified Communications Manager server. Also, security issues like a denial-of-
scrvice (DoS) attack on ihe server could cause a considerable slowdown.
An incorrect dial plan design at ihe Cisco Unitied Communications Manager or gateways
could also cause calling delays or mtcrdigit delays.
) 2010 Cisco Systems, inc
Verifying LAN Connectivity
This section describes how to verify LAN connectiv.ty for duplex and maximum speed.
Verifying LAN Connectivity
LAN Switch
S-itchishc
FstEthernetC/24 status
.,. vl an tel* Spd Type
FaO/24 CUCM Brvt connected 312
Cisco Unified Commuications Manager Server
adninisho- networjt sttiO
Ethernet 0
DHCP
disbled
IP Address ID 1.5.2
Link Detac ted: yes
Duplicate IP = QO
DNS
Not condf ured
Gateway
10 1.5.1
status
IP Kaslt
Hod
Ethernet 0
. 255.255.255.000
i imt(. aaX>Ua. Full. 109 *Mt*/*
Slow response could be the result of incorrect duplex configuration, such as when the settings
onLswitch do not match the duplex pon setting on the Cisco Unified Communicates
Manager server.
For optimal performance, set both switch and server to Auto and Full Duplex for the maximum
speed that is available on the Ethernet media.
On the LAN switch, use the command show interface FastEthernet slot/port status or a
similar command for aGigabit Ethernet pon to display the port settmgs.
On the Cisco Unified Communications Manager server CLI, use the command shim network
ethO todisplay the Ethernet ponsettings.
If Cisco Unified Communications Manager server Ethernet settings are changed, you must
restart the server to enact this change.
Cisco Unified Communicalions Manager Troubleshooting 2-63
Summary
Mil
Major causes of Cisco Unified Communications Manager not
responding include server crash, Cisco CallManager service
fatlure, high CPU usage, or process with amemory leak
' Major causes of not displaying the administration paqe
include network issues, name resolution issues Cisco
Tomcat service failure, high CPU or memory usage or
firewall oraccess list blocking HTTPS traffic. '
Major causes of slow server response include media speed
and duplex mismatches, high CPU usage, malfunctioning
process, ora memory leak.
Jvlil!b|S.T IT*? HCrariS '" Uh'Ch tHC riSC Unified Com-t,ons Manager is not
available and described how to troubleshoot and diagnose the cause of these problems.
References
2-64
ror additional information, refer tothese resources:
' Sw"r"' ^""ff"''1"* G'*' /-' Cisco Unified Commumcations Manager.
Kthus^S.O,/,, Cisco Unified Communications Manager Svstem /. February 2010
http- wuw.ciseu.com en IS Pa,tner docs xo.ee ip comm.cucm.'trouble
> 0 1 tbs\ Mem,html
Troubleshooting Ciscc Unified Communications (TVOICE) v8 0
Lesson 31
Troubleshooting Database
Replication Issues
i. Overview
This lesson describes database replication issues that can occur in aCisco Unified
Communications Manager cluster, how to diagnose areplication issue, and how to re-create the
mm database relationship.
Objectives
Upon completing this lesson, you will be able to define how to identify database ^plication
"Lcs in Cisco Unified Communications Manager clusters and how to repair or re-create
database replication. This ability includes being able to meet these objectives:
. Review the basics of database replication on Cisco Unified Communications Manager
*" Explain how to diagnose database replication issues with Cisco Unified Communications
Manager
_ . Explain the procedure to resolve and re-create the database relationship in aCsco Unified
Communications Manager cluster
Database Replication Issues
s topic renews the basics of database replication on Csco Unified Commumcat,
2-66
Kc-|
IDS Replication i
CTI Manager,.'
t-0,''':.,, ^ U0H Server/
<
TfTP Serv
Publishe
-. ices
::"*
IDS Database Subscribers
Call-Processing Servers
Acluster ,s aset otnetuorked services that work together to provide the Cisco Unified
Comtnunications Manager Service. Dedicated servers also provide database, applet TFTP
ublt he V, ''CeS T " COnfcrcncine * on hold (MOM). The subscrLrs and the
publisher can provide these services and share them among all servers in the cluster.
Clustering provides several benefits. It allows the network to scale to several thousands of
îz^r1încmc of ^ r *-- p-,:r^,
To process calls correctly, Csco Unified Communications Manager needs to retrieve
configuration settings for all devices. These settings are stored m'a database by un^ an IBM
hfonrnx Dynamic Server (IDS). The database is the repository for information suci as sfvL
parameters, features, device configurations, and ihe dial plan.
Susll'fTh^databaSh TCSentS *CrC hinCtl" f CisC0 Unified Communications Manager
u .The sen er wh he master copy of the database acts as the publisher (first node)
uhde the servers that replicate the database comprise the subscribers (subsequent nodi)
mL?^ Jn thC ClSCi UmflCd Communicats Manager cluster replicates nearly all
mrormation ,n astar topology ,one publisher, many subscribers). However Csco Unified
Communications Manager nodes also use asecond communication method to r phc n
ume data mamesh topology .every node updates every other node, This type of
chaTTit m!Shuscdrfor dynamic information lhat cha,lgcs more iY^y^ da**
changes. Use mesh replication to announce newly registered phones, gateways and digital
mem&T' Tr" Th,S aPPraC" mMa PtimUm * "-l"betw
numbers ot the cluster and the associated gateways.
Troubleshooting Cisco Unified Communicalions (TVOICE) -
mm Database Replication Issues
Occasionally, the copies of the database on the publisher and subscribers can go out of
synchronization.
Database Replication Issues
Why the database goes out of synchronization:
Network connectivity might not exist between the publisher
and the subscriber because of network or hardware issues.
The network does not provide the necessary QoS, which is a
typical issue when clustering over the WAN.
* Name resolution has failed due to no entry, an incorrect entry
in DNS, or an unavailable DNS server, if used.
Replicating peers is overused; no CPU power is left for the
replication.
Error occurred in the database replication processes.
Although the Cisco Unified Communications Manager cluster architecture allows each server
towork relatively independently of others, theconfiguration changes thatdo not replicate toall
servers within the cluster can cause operational issues. The IBM IDS is robust and the
replication mechanism reliable, but there could be periods when the database synchronization
does not work as expected:
The replication is performed in the IPnetwork. Broken network connectivity between the
servers or improper operation of server hardware will interrupt the replication.
The replication requires a sufficient amount of bandwidth and must occur in real time,
which means that the network quality of service (QoS) must provide the necessary
resources to supportthe transferof data betweenindividual servers. Typically, you might
encounter bandwidth issues if replication traffic needs to cross a relatively constrained IP
WAN(for example, in the clustering over WAN implementationmodel).
If domain name services are usedin the cluster, a nonresponding DomainName System
(DNS) server could cause a broken replication.
A growing number of endpoints and the complexity of the dial plan that is maintained
could result inservers having insufficient amounts of CPU or memory resources toperform
timely database replication. Follow performance recommendations closely andupgrade the
system when it reaches its limits.
Cisco UnifiedCommunicalions Manager Troubleshooting 2-67
Typical Database Replication Problem Scenario
This replication is typical of a database problem scenario.
I^'.lt ,-ti L/,llr<L) J
Problem report:
* The administrator changes an IP phone configuration.
However, after the IP phone reset, the changes are not
reflected on the IP phone that is registered to a subscriber.
Consider all described possibilities:
* Verify network connectivity.
* Verify that the DNS resolution works, if used.
* Make sure that there is enough bandwidth and a maximum of
80 ms of round-trip delay between the publisher and
subscriber
Diagnose the replication and recreate it if it has failed.
The Cisco Unified Communications Manager Administrator has changed the configuration of
an existing Cisco Unified IPphone. The cluster follows best practices and the IP phone is
typically registered to its primary subscriber. After resetting the Cisco Unitied IP phone, the
change is unavailable.
Other replication issues could be caused by configurationinconsistenciesbetween the publisher
and the subscribers or simply a "Database Communication Error" message that appears in
Cisco Unified Communications Manager logs.
Consider all described possibilities and causes, and the troubleshooting procedure could stati as
follows:
Network connecth ity issues are preventing database replication. Use ping to verily
network connectivity fromthe subscribers to the publisher. Restore network connectivity if
necessary.
Name resolution ha* failed because of no entry, an incorrect entry in DNS, or an
unavailable DNSserver. Ping by name to verify name resolution. Fix the name resolution
to pro cut future problems.
Make sure that you have enough network resources to assist in replication. The round-trip
delay betwecn the two replication peer servers must be, in the worst case, 80 ms.
If an error occurred with database replication that caused database replication to fail, re-create
the database relationship between the publisher and ihe subscribers.
Troubleshooticg Cisco Unified Communications (TVOICE) v8.0
Diagnosing Database Replication Issues with
This topic explains how to diagnose database replication issues with Cisco Unified
Diagnosing Database Replication
How to check whether the database is out of
synchronization:
You can use the CLI, Cisco Unified Reporting, or
Cisco Unified Comunications Manager RTMT.
All three options report the same Replicate_State
object indications:
' 0: Replication did not start.
1: Replicates have been created, but their count isincorrect.
2: Replication is good.
3: Replication is bad in the cluster.
4: Replication setup did notsucceed.
To verify database replication, use the command-line interface (CLI), Cisco Unified Reporting,
or the CiscoUnifiedReal-Time Monitoring Tool (RTMT).
All three tools display the so-called Replicate_State object, which represents the state of
replication as anumerical value. The following list shows the possible values for the
ReplicateState:
0: This value indicates that replication did not start. Either nosubsequent nodes
(subscribers) exist, or the Cisco Database Layer Monitor service is not running and has not
been running since the subscriber was installed.
1: This value indicates that replicates have been created, but their count isincorrect.
2: This value indicates that replication is good.
3: This valueindicates that replication is bad in the cluster.
4: This value indicates thatthereplication setupdid not succeed.
CiscoUnified Communications Manager Troubleshooting 2-69
Cisco Unified Reporting
This is an example of how Csco Unified Reporting can be used to check the replication state.
2-70
. _j ,
-
UnHlW CM DilBbatc Sl-I.j.
;-r;;;^:rf,':3,f"
::
..:..' . ..,-..,.. ..,,.==,,
. < -..
....:=,.., "'xa"

'U-M [<=>,.,,
' '" "*"" ' ~" '"'' -'"-
w- , ., .
--_ _ *" ' ':- " -'
T;
-__j_
,.'..',' ""*' '"" ''""
-:
j;";"": -:-''
'^y^~:~,.
.........
".: :":;"' """<-
States:
0
1
2
3
4
Use Cisco Unitied Reporting as follows:
Step 1 From the Navigation drop-down menu, choose Cisco Unified Reporting.
When Cisco Unified Reponing displays, click the System Reports link.
Step 2
Step 3
Step 4
Generate and view the Unified CM Database Status report, which provides
debugging infonnation for database replication and shows the replication counters
for all servers in the cluster. All servers should have areplicate state of2. AI! ser\ers
should ha\ ethe same number ofreplicates created519 in the example.
It you see any servers whose replicate states are not equal to 2, inspect the
Replication Server List on this report (not expanded in the example). It shows which
servers arcconnected and communicating with each node. Each server should show
itself as local (in its list) and the other servers as active connected. Ifyou see any
servers as dropped, it usually means that there is acommunication problem between
the nodes.
Step 5 Optionally, generate and view ihe Unified CM Database Status report, which
provides asnapshot ofthe health ofthe Cisco Unified Communications Manager
database.
Cisco Unified RTMT Database Summary
Thisexample shows how CiscoUnified RTMT canbe used to checkthereplication state.
Cisco Unified RTMT Database
To verifydatabase replication by usingCisco Unified RTMT, perform thesesteps:
Step 1 Open the Cisco Unified RTMT.
Step 2 Click the CallManager tab.
Step 3 Click Database Summary. The Replication Status pane displays. The table shows
thereplication status (Replicate_State) andthereplicates thatarecreated. Normally,
the status of 2 is reported. All serversshouldhave the samenumberof replicates.
Cisco Unified RTMT Performance Counters
This example showshowto displaythe ReplicateState performance counter.
'5 i* i L.*-.|n-l *,(.!,*
4- * - - [ t D
[.-lUf n4 hfU?l
Cxiifd i..*ûa A
r>f ., t ,., "p*r A***-***"*' P*t*t.kw.W**rtH
^L"" -in i -
'""HHMiîî^M }"^^^^^^^W
*"l " - -T-J- ='-' J I * "
^^^B^^^B
-^-^-^-^-^^-^H
îJl - ' d ,,! .],= -. .. jiiîî^HLîî^H l^Hi^H
*M"3 " " ' "- = ,1 .1 1 ii 1 -.
^H^H
*! "'
^^^^^^
*-L3:r , -.>!,,.- H.^li^.^H ^.^H^.^H
frl'E^L, ,F.-! ,&.' l-w . j-ie-j !'!^^^^ U" IPHH
r*d- - "- " *- r=.<r H ".Mil- t**T -i..a.l.- :*n.
PwlirvntMm
Youcan also view the Replicate_Stateperformance monitoringcounter:
Step 1
Step 2
In Cisco Unified RTM1. choose System> Performance> Open Performance
Monitoring.
Double-click the publisher database server (first node) to expandthe Performance
Monitors,
Step 3 ClickNumber of Replicates CreatedandStateof Replication.
Step 4 Double-clickReplicateState and Number of Replicates Created. Thecounters
appear on the right side of the Performance window as graphs.
Ihe example -*hows the counters for a subscriber and indicates that the replication goes well,
because the Replicate_Siateis constantly equal to discrete 2.
2-72 Troubleshooting Cisco Unified Communicalions (TVOICE) v8.0
Cisco Unified Communications Manager CLI
This example shows how to use the CLI to diagnose replication problems.
Manager CLI
Databases suspected not in synchronization
admin =ijt ils dBreplicati
statue
utils dbreplicatioQ status
jZ =0J Mtil. dbr.plic.ttoa wMttt> t* *** l pro8*
The final output will ba in file
Cm/traCe/dbl/Bdi/HBpllctioi.Status.2010_01_27_16_39_53.out
PI* o 'tils vi ctwloa . ^
see the output .
If all the servers have agood Cisco Unified RTMT status, but you suspect that the databases
are not synchronized, you can run the Cisco Unified Communications Manager utils
dbreplication status CLI command. You should execute this command only on the publisher
of a cluster.
You can run this status command for all servers in the cluster by using the utils dbreplication
status all command or for one particular subscriber by using utils dbreplicat.ua status
<hostname> command. The status report will tell you ifany tables are suspect.
The example shows that the command runs in the background and that all the status
information goesto a logfile named
cm/trace/dbl'sdi/ReplicationStatus.2010_0i_27_16J9_53.out.
Use the utils dbreplication runtimestate command to check the generation process of the
replication log file.
>2010 Cisco Systems. Inc.
2-74
Cisco Unified Communication:
16-02
tils dbreplication runtimestate
DH and Replication Services: ALL RUNNING
Cluster Replication State: Replication status
and started at: 2010-01-27-
ssir^rtSS-ir090" m w*ua -^- <
REPL. REPLICATION SETUP
SERVER-NAME ip ADDRESS
LOOP? (RTMT) t details
cucKSOl-pub
(2) PUB Setup Completed
cucbSOi-tub
7es (2)
REPLICATION
STATUS
= ted
:ted
REPL. DBvetS
QUEUE TABLES
atch
atch
Using the utils dbreplication runtimestate command repetitively shows the progress of the
started replication status. '
The example shous that 240 of 519 tables have been processed so far and that the replication
status is still running. '
The 'able should list all members of the Cisco Unified Communicalions Manager cluster with
he.r I addresses and network connectivity state (delay in milliseconds). The Remote
rocedurc tails (RPQ should say "Yes" to indicate that the servers have initialized the'
database connectivity.
Manager CLI (Cont.) x
adminûtilB dbreplication runt inestate
DB and Replication services: all ruhNiHQ
n Stat Clustei Replication State: Beplici
16-02
Replication ttu Oamtaml COWlStSO SI* Cabl* oMtkad Ottt ot 519
Errora or Klraatclwa Wars Pouadi
n 'til* vi aetliralog
e/tr*c*/disl/dl/iipnctionat.ta*.1810l)i_iT^l_l_53.ouf 6o * U* HfcUa
DB Version: ccm8_0 l_X0O00_17
Number of replicated tablem: 519
(Output omitted)
nd started at; 2010-01-27-
Take the repetitive utils dbreplication runtimestate command output on the same server on
which you executed the command utils dbreplication status until you see "Replication status
command COMPLETED" along with the number of replicas (tables) processed.
The example shows the replication status reports errors and mismatches in the database. You
can find all replication status details in the log file.
2010 Cisco Systems, Inc Cisco Unified Communicalions Manager Troubleshooting
2-76
anaqer
sdmin:file view activelog
jsn/trace/dbl ,'sdi ,'Repl nations tatus. 2010 CI 27 16 39 53.
ID STATE STATUS QUEUE CONNECTION CHANGED
g cijcmaO 1 pub 0 1 10000 17
0 J 10000 17
Snapact Replication Sunoary
27 16:13:33
For title.
ecmd bteaplat _eueaaDl _puh_ccB_0_ 1_1Q0 00__17__i__15 S_tjrpegroupve EBiooItup
replication ia auapact tor aodafali
g_cucaSOl_aub_ce8_0_1_10 00 0_17
Foe table i ccBdbtiplate_cuOB801jiub_ccBg_0_l_lllD!)O_a7_l_33_proo*ceO(i9
replication ia auapact for node Oil t
B__cueIIOi_aub_ec8_0_l_i0D0IJ_17
For table: ccsidl)taiplte_cuc01j>ul)_cOB813_l_10DOO_17_l_3<l_proO*sao<le*rvic*
replication ia auapact tot nodaii) i
gcucoSQl aub ccb8 0_i 10000 17
Display the log file as seen in this example. The log file spans over tens of CLI screens, but it is
not necessary to browse them all. Already its beginning indicates the replications suspects. The
long suspect replication summary list gives more details on which replicas (tables) are
incorrect.
If there are suspect tables, perform a replication repair to synchronize the data from the
publisher ser\er to the affected subscriber servers.
Resolving Database Replication Issues with
This topic explains the procedure to resolve and re-create the database relationship in the Cisco
Unified Communications Manager cluster.
Resolving Database Replication Issues
If suspects exist, attempt to repair thereplication:
Run utils dbreplication repair.
? Verify the replication status if the failure has been resolved.
If not, and if any subscribers showa status of 4 or
had a status of 0 for more than four hours, reset the
replication:
i Stop the replication between thesubscriber and publisher.
2 Reset the replication ofthe failed subscriber.
3 Verify the replication status if thefailure has been resolved.
Ifsuspects are reported, attempt to repair the replication between the publisher and the
subscriber:
Step 1 At the CLI, run the utils dbreplication repair command.
Step 2 When the repair is completed, verify the replication status to see ifthe failure has
been resolved.
Ifthe repair did not help and suspects remain, consider the following. Ifany subscribers still
show areplication status of4, or had the status of0for more than four hours, you might need to
reset the replication:
Step 1 Stop the replication between the subscriber and the publisher.
Step2 Reset thereplication of thefailed subscriber.
Step 3 Verify the replication status ifthe failure isresolved.
Cisco UnifiedCommunications Manager Troubleshooting
2-77
Repairing Database Replication
This example shows how to repair the database replication.
2-78
jj %,*. r*~fenri(,*v
.t'iJiiil II |U
<- Be aware the repair process affects the cluster performance.
' Run at the publisheronly.
- Wait until "COMPLETED" is displayed.
*Then check whether tables are synchronized
admini-jtils dbreplication repair ill
utll9 dbreplication repair:
Replication Repair is no. running in the background
Use command 'utils dbreplication runtimestate' to check its progress
Output will be in file
cm/ tracs/dbl/sdi/ReplicationflepBir.1993 02 28 17 30 25.out
PleH us* -fila viaw activalog
/Crc./dbl/wii/RBpUe,tion8.p,iE.1,9^()a_aB_17__30_2s_ollt - CtwBaDa to 1 Eh.
You can make the replication repair on all subscriber servers (using the all parameter) or on
just one subscriber server by using the utils dbreplication repair |m>dename||all command.
The repair is triggered from the publisher.
The repair process will run in the background, and you can monitor its progress by using the
utils dbreplication runtimestate command that was used before.
All the details ofthe repair process are logged tothe log tile.
When running the repair, be aware that this process affects the performance ofthe cluster, and
it is recommended that you perform the database repair during maintenance hours ifpossible.
Wait until the progress displays COMPLETED. After running the replication repair, which can
take several minutes, you can run another status command to verify that all tables arc
synchronized. Ifthe tables are synchronized aflcr running the repair, you arc successful in
fixing the replication.
Troubleshooimg Cisco Unified Communications (TVOICE) vB 0
2010 Cisco Systems, int.
Resetting Database Replication
If the repair did not help, the replicationreset may be needed.
Kesettint
If repair did not help, reset the replication.
First, check that the connectivity to all nodes is maintained.
dull*" rrt "v Pi4r=is*it.
10.1 S.I irm
10.1.5.3 t"M
If Cisco Unified RTMT shows for a subscriber server the status "bad" (database replication
suspect), checkthat the hosts, rhosts, sqlhosts, and servicesfiles havethe appropriate
information.
Generate and view the Cisco Unified Communications Manager Cluster Overview report in
Cisco Unified Reporting. Verify that the subscriber servers have the same version, that
connectivity is good, and that the time delay is within tolerances.
lesetting Database
Subscriber
adminmtils dbreplicat ion atop
>,lU><l>t>IIOI=<>>l<t<tl><l<ltl
This command will delete Che market file!
is stopped
It will also stop any replication setup c
) bo that automatic replication setup
rrently executing
(May take considerable time to show any c mmand output)
Deleted the marker file, auto replication setup is stopped
Commanded Out of Service
A Cisco DB Replicator [NOTRUNNING]
A Cisco DB Replicator [STARTED)
Completed replication process cleanup
(Output omitted)
If the conditions are acceptable, do the following to reset the replication.
Step 1 Start u ith the subscriber servers. At the failed subscriber servers, use the utils
dbreplication stop command to stopthe automatic setupof database replication. Do
this for all subscriber servers that have the replication state value of 4.
2-80 Troubleshooting Cisco Unified Communications(TVOICEl v8 0
2Q10Cisco Systems Inc
%
Publisher
idmin:utils dbreplication stop cucmBOl-sub
This command will delate the marker tilela
is stopped
It will also stop any replication setup currently executing
that automatic replication setup
(May take considerable time to show any command output!
Deleted the marker file, auto replication setup is stopped
(Output omitted)
will stop PUB and SUBS: cucmBOl-sub
Stopping Sub: cucmBOl-sub
Stop replication sub Completed
Completed replication process cleanup
Please run the command 'utils dbreplication runtimestate' and make sure all
nodes are RPC reachable before a replication reset is executed
Step2 At thepublisher server, perform the utils dbreplication stop command for all or for
a particular subscriber, as shown in the example.
"Iheentire process of stoppingthe replication may takeconsiderable time. Leavethe
publisher intact until the CLI prompt reappears.
)2010 Cisco Systems, Inc Cisco Unified Communications Manager Troubleshooting 2-81
2-82
Publisher
admin;utils dbreplication resist cucmBOl-sub
Repairing of replication is in progress.
Background repair of replication will continue after that for 30
OK [10.1.5.31
minutes....
admin-.utils dbteplicatisn runtimeatate
DB and Replication Services: ALL RUNNING
Cluster Replication State: BBOABCAST SYKC Completed on 1 server*
27-18-05
U 3010-01-
Last Sync Result. S1THC COMPLETED 519 tables sync'ad out of
Sync Errors: HO KKSQHS
S19
DB Version: ecus 0 1 10000 n
Cumber cf replicated tables: 519
(Output omitted)
Step 3 At the publisher server, use the utils dbreplication reset command to reset and
restart thedatabase replication. Thecommand canbetriggered for a particular
subscriber by its hostname, as seen in the example, or for all subscriber servers that
need to be reset. The command causes subscriber servers in the cluster to have their
replication torn down and rebuilt.
Step4 Continue to take repetitive utils dbreplicaiton runtimestate command output until
the "Last Sync Result" shows SYNC COMPLETED.
Step5 Verify that the tables arc nowsynchronized by usingthe utils dbreplication status
command.
You can also use the command utils dbreplication clusterreset to reset the database
replication. This command will tear downand rebuild replication for the entirecluster. After
using thiscommand, youmust restart each subscriber server. However, youshould only use it
if you havealready triedutils dbreplication reset all, and it failed to restart replication on the
cluster. After all subscriber servershavebeenrestarted, youmust go to the publisher serverand
issue the command utils dbreplication reset all.
a Resetting the Cluster
If resetting the database replication didnot help, the reset of the cluster may be needed.
Resetting the Cluster
Subscribers
idmimutils dbrepli
... output omitted
Pubisher
admimutlls dbreplication stop
. . . output omitted . . ,
admimutils dbreplication clusterreset
This command will repair replication on all nodes in the cluster
Before running, eiecute dbreplication atop eserven on all subscribers
than axacuta dbreplication stop on tha publishar
liter clusterreset, "utils dbreplication reset all" should be executed
followed by the reboot of all subscribers
***** * ...................................................... ....
This command can take considerable amount of time, and will tear down
replication and build it bach again.
Are you sure you want to continue? (y/n):
Use the following procedure only if you have already tried to reset replication on the node, and
have been unsuccessful:
Step 1 Run the utils dbreplication stop command on the affected subscribers. You can
perform this command on all subscribers at the same time.
Step 2 Wait until Step 1 completes, then, run the utils dbreplication stop command on the
affected publisher server.
Step 3 Run utils dbreplication clusterreset from the affected publisher server. When you
run the command, the log name gets listed in the log file. Watch the
' var/!og'active/cm/trace/dbl/sdi file to monitor the process status.
Step 4 From the affected publisher, run utils dbreplication reset all.
Step 5 Stop and restart all the services on all the subscriber servers. Alternatively, restart or
reboot all the systems (subscriber servers) in the cluster to get the service changes.
Do this only after utils dbreplication status shows Status 2.
Cisco UnifiedCommunications Manager Troublestiooling 2-83
Summary
The database in the cluster replicates information in a star
topology (one publisher, many subscribers). However, Cisco
Unified Communications Manager nodes also use a second
method to replicate run-time data in a mesh topology.
To verify database replication, use the CLI, Cisco Unified
Reporting, or Cisco Unified Communications Manager RTMT to
display the Replicate_State object.
If replication suspects are reported, first attempt to repair the
replication. If the repair did not help and any subscribers still show
the replication status of 4 or have had the status of 0 for more than
four hours, reset the replication.
This lesson has described the database replication issues that can occur in a Cisco Unified
Communications Manager cluster, how to diagnose them, and how to re-create the database
relationship.
References
E-or additional information, refer to ihese resources:
Cisco Systems, Inc. Troubleshooting Guide for Cisco Unified Communications Manager.
Release H.O(I). Cisco Unified Communications Manager System Issues. February 2010
Imp: www ei'-co.com en US partner''does, voice ip cnmnr'cucm Konhle-
,S 0 I ;b-,\ stem.html
Troubleshooting Cisco Urified Communications (TVOICE) vB.O 2010 Cisco Systems. Inc.
m*
Lesson 4
Troubleshooting LDAP
Integration Issues
Overview
CiscoUnifiedCommunications Managercan be configured to integrate withselect third-party
Lightweight Directory Access Protocol (LDAP) services.
Failure to replicate the LDAP is a problem that can occur in the Cisco Unified Communications
system. Knowing how to troubleshoot the LDAP integrations and replications will ensure a
consistent directory. This lesson describes issues that can occur with interactions between Cisco
Unified Communications Manager and the LDAP services.
Objectives
Upon completing this lesson, you will be able to explain how to troubleshoot LDAP
synchronization or LDAP authentication issues when using LDAP integration, This ability
includes being able to meet these objectives:
Review the LDAP integration options of Cisco Unified Communications Manager and
explain the typical issues that might occur when integrating with LDAP
Explain the procedure to identify and resolve synchronization issues with Cisco Unified
Communications Manager when it is integrated with Microsoft Active Directory
Explain the procedure to identify and resolve authentication issues with Cisco Unified
Communications Manager when it is integrated with Active Directory
LDAP Integration Options with Cisco Unified
This topic describes the LDAP integration options of Cisco Unified Communications Manager
and explains the typical issues that might occur when integrating with LDAP.
with Cisco Uimi&u i-ommumcai
End-user lookups
End-user authentication
> End-user provisioning via database synchronization
I
*>
IP Telephony Applications
y
n
IP Telephony Endpoinls
Corporate LDAP
Directory
r/.:roBotiAotve Directory
Microsoft ADAM and LDS
Netscape Direclory Serve
.Plaiel Directory Server
Su'i ONE Directo:y Surwu
Open.. DAP
'
IPTelephony End Users
LDAP directories typically store data thai docs not change often, such as employee
information, user privileges on the corporate network, and so on.
The information is stored in a database thai is optimized for a high number of read and search
requests and occasional write and update requests.
LDAP directories store all user information in a single, centralized repository that is available
to all applications. Applications can access the directory by using the LDAP, which provides a
standard method for reading and potentially modifying the information that is stored in the
directory.
Integrationbetween voice applications and a corporate LDAP directory is a common task for
many enterprise IT organizations. However, the exact scope of the integration varies from
company to company, and it cantranslateto one or morespecific and independent
requirements.
For example, a common requirement is to enable user lookups (sometimes called the "while
pages" service) from IP phones, so lhal users can dial a contact directly after looking up its
number in the directory.
Another requirement is to provision users automaticallyfrom the corporate directory into the
user database of unified communications applications. Automatic user provisioning from
corporate directory avoids having to add, remove, or modify core user information manually
each time a change occurs in the corporate directory.
Often, authentication using corporate directory credentials isalso required for end users and
administrators of theunified communications applications. Authentication enables theIT
department to deliver single-logon functionality and reduces the number of passwords that each
user needs to maintain acrossdifferent corporate applications.
You can satisfy each ofthese requirements by using aCisco Unified Communications system
that uses different mechanisms according tothe Cisco Unified Communications Manager
version that you use.
Cisco Unified IP Phones that are equipped with adisplay screen can search auser directory
when a user presses the Directories button onthe phone. The IPphones use HTTP tosend
requests toa web server. The responses from the web server must contain some specific XML
objects that thephone caninterpret anddisplay.
By default, Cisco Unified IP phones are configured toperform user lookups against the Cisco
Unified Communications Manager embedded database. However, you can change this
configuration so that the lookup isperformed onacorporate LDAP directory. In this case, the
phones send their HTTP requests to an external web server that operates as aproxy and
translates these requests into LDAP queries against the corporate directory. The LDAP
responses are then encapsulated inthe appropriate XML objects and sent back tothe phones via
HTTP.
Cisco Unified Communications Manager supports the following directories:
Microsoft ActiveDirectory(2003, and2008)
Microsoft Active Directory Application Mode (ADAM) 2003
Microsoft Active Directory Lightweight Directory Services (LDS) 2008
Netscape Directory Server 4.x
iPlanet Directory Server 5.1
Sun ONE Directory Server 5.2 and 6.x
OpenLDAP 2.3.39 and 2.4
Cisco Unified Communications Manager supports two types ofLDAP integration, which can
be enabled independently of each other:
LDAP synchronization: Allows user provisioning in which personal and organizational
data is managed inanLDAP directory and replicated totheCisco Unified Communications
Manager configuration database.
LDAP authentication: Allows user authentication against an LDAP directory: When using
LDAP authentication, passwords arc managed inLDAP.
Application users are not affected by LDAP integration. Application users are always
configured from Cisco Unified Communications Manager Administration, and their data is
always stored in the Cisco Unified Communications Manager configuration database.
2010 Cisco Systems, Inc. cisco Unjfjed Communicalions Manager Troubleshooting 2-87
Cisco Unified Communications Manager LDAP Integration
Options
The table shows how different user data is treated when using LDAP synchronization and
contrasts it to a scenario in which LDAP authentication is enabled.
Personal and organizational
settings
User ID Local
First, Mddle, and Last Name
Manager User IDand Department
Phone Number and Mail ID
Password Local
Manager Settings
PIN and Digest Credentials
Groups and Roles
Associated PCs Local
Confro&ed Devices
Extension MobilityPfofie and
CAPF Presence Group and
Mcbitty
LDAP {replicated
to local)
Local
LDAP (replicated
to local)
LDAP
Local
When you use LDAP synchronization, personal and organizational settings are configured and
stored inLDAP. With each synchronization, the data isreplicated to the Cisco Unified
Communications Manager database. However, as long as LDAP synchronization is enabled,
thisdatacannot bemodified inCisco Unified Communications Manager. User passwords and
Cisco Unified Communications Manager configuration settings arestillconfigured byusing
Cisco Unified Communications Manager Administration andstored in the Cisco Unitied
Communications Manager database only.
With LDAP authentication. Cisco Unified Communications Manager authenticates user
credentials against a corporate LDAP directory. When this feature isenabled, end-user
passwords are no longer stored in the Cisco Unified Communications Manager database (and
are not replicated to that database) but arc stored only inthe LDAP directory.
Personal user data is managed either in LDAP and replicated into the Cisco Unitied
Communications Manager database (ifLDAP synchronization isenabled) or iscontrolled
(managed and stored) only byCisco Unified Communications Manager.
Cisco Unified Communications Manager userdata(such as associated PCs or controlled
devices) are stored in the Cisco Unified Communications Manager database for each individual
user. Therefore, the usemame has tobeknown inthe Cisco Unified Communicalions Manager
database (to assign Cisco Unified Communications Manager user settings to the user) and in
the LDAP directory (to assign the password to the user). To avoid separate management ofuser
accounts in these two databases, it is recommended thatyoucombine LDAP authentication
with LDAP synchronization.
Troubleshooting CiscoUnified Communications (TVOICE) v80
LDAP Integration Considerations
Consider these facts when integrating withiLDAP.
Consider these LDAP integration facts:
Cisco DirSync service performs synchronization.
All synchronization agreements musl integratewiththe same LDAP family.
Asynchronization agreement for a domain will not synchronize users outside
of that domain nor within a cNId domain.
One LDAP user attribute is chosen to map intothe Cisco Unified
Communications Manager User IDfield.
Whenirtegrating with a Microsoft Active Directory forestcontaining multiple
trees, make sure that user IDmaps to an attribute that is unique across the
forest.
When synchronization is enabled, all existing Cisco Unified Communications
Manager end-user accounts are deactivated and purged fromthe Cisco
Unified Communications Manager database after 24 hours.
Authenticated users mustexist inthe LDAP directory (LDAP synchronization
is not mandatory but recommended).
Thesynchronization is performed bya feature service that iscalled Cisco DirSync, which is
enabled through the Serviceability web page.
All synchronization agreements must use the samesynchronization method. You cannot mix
synchronization agreements withActiveDirectory and any other LDAP server. CiscoUnified
Communications Manager uses the LDAP version 3.
A synchronization agreement specifies a so-called search base. A search base is an area of the
directory that should beconsidered for the synchronization. Asynchronization agreement is
achieved byspecifying a position in thedirectory treewhere Cisco Unified Communications
Manager begins itssearch (that means that it has access toall lower levels but not tohigher
levels).
Asynchronization agreement for a domain will not synchronize users outsideof that domain
nor within a child domain. The domain controllers have information only onusers within the
domain where they reside; therefore, multiple synchronization agreements are required to
import all of the users in a tree.
One LDAP user attribute (for example sAMAccountNamc, uid, mail, ortelphoneNumber) must
be mapped tothe User ID field ofauser in Cisco Unified Communications Manager and must
be unique across all users.
When synchronization orauthentication isenabled with anActive Directory forest that contains
multiple trees, multiple synchronization agreements areneeded. Inaddition, the
UserPrineipalName (UPN) attribute must be chosen as the attribute that is mapped tothe Cisco
Unified Communications Manager User ID because only this attribute isguaranteed by Active
Directory to be unique across the forest.
CiscoUnified Communicalions Manager Troubleshooting 2-89
2-90
At thebeginning of thesynchronization process, all existing Cisco Unified Communications
Manager end-user accounts aredeactivated. LDAP useraccounts that existintheCisco Unified
Communications Manager database (which are nowdeactivated) are reactivated, and their
settings ate updated if there are any changes.
This synchronization ensuresthat updatesare propagated. Inactive users that wereconfigured
in Cisco Unified Communications Manager but not present in LDAP will be deleted after 24
hours.
With LDAP authentication. Cisco Unified Communications Manager authenticates user
credentials against a corporate LDAPdirectory. When this feature is enabled, end-user
passwords are nolonger stored intheCisco Unified Communications Manager database (and
arc not replicated to that database) but are stored only in the LDAP directory.
Troubleshooting Cisco Unified Communicalions (TVOICE) v80 2010 Cisco Syslems, Inc
Major LDAP Integration Issues
This section lists the major issues when integrating with LDAP.
Major LDAP Integration Issues
The major issues when integrating with LDAP:
Network connectivity problems with domain controllers.
Domain controllers are down.
Service username or password issues.
Service user has incorrect permissions.
End users cannot connect to Cisco Unified Communications
Manager user pages after LDAPauthentication is enabled.
User search base points to incoiTect position in a tree.
Directory synchronization is set to manual or fortoo longa period.
Directory integration not operational due to misconfiguration or
Cisco DirSync not running.
Some end-user accounts were not synchronized from directory.
These are the major LDAP integrationproblems:
Most common problems are network connectivity issues with domain controllers or global
catalogs, incorrect addresses, port numbers, Domain NameSystem(DNS) resolution not
working as expected, or firewalls that block the LDAP traffic.
Domain controllers donot takeanyLDAP requests if they aredown, if services have
stoppedat the server, or if their database is corrupted.
Often, external systemswill use a special, dedicatedserviceaccount to accessthe LDAP
database. The service account username or password might bemistyped, or a special
character might be used in the password.
The service account onthe LDAP server might have incorrect permissions (read access
needed).
When LDAP authentication isenabled, end users might complain that they can no longer
connect toCisco Unified Communications Manager user pages. It islikely that they
configuted a different password at the LDAP server from theonethey used earlier at Cisco
Unified Communications Manager. The Cisco Unified Communications Manager
Credential Policy lets you configure a default password and some password policies. All
users that aresynchronized from Active Directory then usethis template for their
passwords. The same applies for the PIN.
Theusersearch base might point toan incorrect position ina tree, andtheusers that arc
supposed to synchronize are inaccessible.
Ifdirectory synchronization isset tomanual orisset for too long a period, LDAP server
updates do not reflect instantly at Cisco Unified Communications Manager.
CiscoUnified Communications ManagerTroubleshooting 2-91
Directory integration mightnot be operational because of misconfiguration, or the Cisco
DirSync servicemightnoi be running on CiscoUnifiedCommunications Manager.
If some end-user accounts are not synchronized from the directory, the user ID might not
be mapped uniquely. Users canbe stored in thechild domain if they ate not explicitly
included in the synchronization agreement.
2-92 Troubleshooting Cisco Unified Communications (TVOICE) v8.0 2010 Cisco Systems, Inc
LDAP Integration Fictitious Issues
This section lists the fictitious issues that users believe are aresult ofLDAP integration, but
they represent normal, expectedbehavior.
LDAP Integration Fictitious Issues
The following liststhe normal, expected behavior:
End users configured at Cisco Unified Communications Manager
disappear after LDAP synchronization isconfigured.
End users cannot be configuredat Cisco Unified
Communications Manager afterLDAP synchronization is
configured.
End user passwords are notconfigurable by administrator after
LDAP authentication is configured.
- End users cannot change their passwords byusing the Cisco
Unified Communications Manager userweb page.
End-userPIN attribute cannot be found and set at LDAP server.
Subscribers do not replicate with LDAP server.
These are themajor fictitious LDAP problems:
End users that are configured at Cisco Unified Communications Manager disappear after
LDAP synchronization isconfigured. This isnormal; all users that were not found in the
LDAP directory are marked inactive and purged after 24 hours by the garbage collector
process.
End users cannot be configured at Cisco Unified Communications Manager after LDAP
synchronization is configured. This is normal; the LDAP directory now manages all users.
End-user passwords are not configurable after LDAP authentication is configured. This is
expected; the LDAP directory now manages all user passwords, and Cisco Unified
Communications Manager never synchronizes any user passwords.
End users cannot change their passwords by using the Cisco Unified Communications
Manager User web page. When LDAP authentication is configured, the LDAP directory
manages all user passwords.
The end-user PIN attribute cannot be found at the LDAP server. This is normal; PIN is not
part ofthe LDAP server, and it remains administered at Cisco Unified Communications
Manager.
Subscribers do not replicate with the LDAP server. Only the publisher replicates with the
LDAP server. End users arc replicated to subscribers via the normal replication process
with the rest of the database from the publisher.
>2010Cisco Systems, inc.
Resolving Synchronization Issues in Cisco
Unified Communications Manager Using Active
Directory
This topic explains the procedure to identify and resolve synchronization issues with Cisco
Unified Communications Manager whenit is integrated with Active Directory.
2-94
10.1.5.4
LDAP server
*s
I
Cisco Unified
Communications
Manager Publisher DC port: 389
GC port 326
Idap Cisco com
cucm.cisco com
admin:utils network ping 10.1.5.4
PING 10.1.5.1 (10.1.5.4! 56(81) bytes ot
64 bvtes from 10.1.5.1: i emp eeq=0 ttl=13
e=5.
byte
byte
byte
10.1.5.
10.1.5 .
10.1.5.
eq^l ttl-128 tiae-1.13 o
eq.2 ttl=128 tin.e-0.2n
eq=3 ttl-128 time=0.580
--- 10.1.5.* ping statistics ---
4 paeketi trnnited, * roctivod, 0% packot lata,
rtt in/a/m/pdv . 0. 347/1.971/5.924/* .30* me,
time SOOln
pips 2
TCP
TCP
TCP
TCP
.0.0.0 :3389
10.1.5.4:319
10.1,5.1:1025
0.0.0.0:0
o.o.o.o io
10.1-5.4ilQ39
10 .1.5.4:1131
LISTENING
LISTENING
BSTABLISHEB
ESTABLISHED
Use ping from Cisco Unified Communicalions Manager to verify LDAP address reachability.
Pint; the IP address ofthe LDAP server from the Cisco Unified Communications Manager
command-line interlace (CLI) orfrom Cisco Unified Operating System Administration. Ifyou
use DNS use ping LDAPJioslname also to verify ifthe hostname ofthe LDAP server is
properly resolved. The figure shows how to ping from Cisco Unified Communications Manager
CLI byusing ihecommand utils network ping.
Check ifthe correct pori numbers arc used in the Cisco Unified Communications Manager
LDAP directory configuration. The domain controller uses the TCP port 389; the global catalog
uses the TCP port 3268. Verify that the ports are open at the LDAP server. Output from the
netstat command shows aport verification example on an Active Directory Domain Controller.
Troubleshooting Cisco Unified Communications (TVOICE) vB 0
Verify Services
This section describes how to verify that services arc running.
Verify Services
Publisher
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Verify that the Cisco DirSync feature service runs at the publisher. Navigate to Cisco Unified
Serviceability and choose Tools > Control Center - Feature Services, then select the
publisher from the drop-down menu.
On the LDAP server, verify that the directory services arc running properly. The figure shows
an Active Directory example. Use Event Viewer with Directory Service view to verify that
Active Directory services do not report any errors or warnings that could indicate problems.
Verify Service Account
Often a special service account is used to access the LDAP directory.
2-96
"ifv Se
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Make sure to create the service account in Active Directory to allow the Cisco Unified
Communications Manager synchronization agreement to conned and authenticate to it. This
account must be able io n.Wall user objects wilhin the desired search base and have a
password that is set never to expire. Alternatively, the administrator account is used, but any
other account with read access to all user objects within the desired search base suffices.
In this example, a separate service account with the username LDAP_Mgr was created in the
lab.com Active Directory domain and within the Users container. The password policy was
changed from the default when the user would have to change the password at the next login to
the fixed password that never expires. The password is not allowed to contain any special
characters.
The sen ice account, when created in Active Directory, belongs to domain users and
authenticated users groups. These groups have read access permission to all user objects by
default.
Troubleshooting Cisco Unified Communicalions (TVOICEl v8 0 2010 Cisco Systems. Inc
Verify LDAP Directory Configuration
In Cisco Unified Communications Manager verify that LDAP synchronization is enabled.
Verify LDAP Directory Configuration
To start synchronization, mark the check box in the LDAP System Information section in Cisco
Unitied Communications Manager.
theôllowin^'^ CommU1,iCatins Mana8er' verifV the LDAP Directory configuration, namely,
Check that the LDAP service account, called "LDAP Manager Distinguished Name," uses
the same credentials that are configured on the LDAP server. Be sure to use one of these
c reucn1131 s.
Complete canonical name, for instance, cn=LDAP_Mgr,cn-Users,dc-lab,dc=com
UPN, for instance, ldap_mgr@lab.com
' fDAÛr Sear,? BaSC iS the Path that deflncs from where the synchronization pulls users
from the Active D.rectory. Cheek that the LDAP User Search Base points to the correa
" aZZ n1^'!017 SrChrnifti0n SchedU,e SCtS h0W often ,he users Pulled from the
Active Directory. You can also set it to perform the synchronization just once. If you mark
this check box, you must trigger the synchronization manually in the future To pull user
changes to the Cisco Unified Communications Manager automatically set the
^synchronization period. Make sure that ^synchronization is not set to too long avalue
like once amonth, for instance. Active Directory does not send automatic incremental '
updates so if user information ,s changed in Active Directory, in the worst case it could be
reflected ,n Cisco Unified Communications Manager after one month. This per od also has
cZlTZTc V3lUe' """I; "6h0WS'If "" chan*CS arC slWsed toê seen in
manually Commu"lcatlon Mer instantly, perform the complete synchronization
' configuration USC ^ ^^*^^" hostmme and Port in the LD^ directory
32010 Ctsco Systems, Inc.
Troubleshooting LDAP Synchronization
The fieurc shows most common errors that are made when configuring LDAP synchronization.
2-S
LDAP Directory Configuration
|Q-
Cisco DirSync Trace
2010-02-03 12:02:09,253 ERROR [DSLDAPSyncImpl(954cc920-b54f-elda
f61b548998e9)] ldapplugeble.DSLDAPSyncIii.pl (DSLDAPSyncImpl.Java:
LDAPSyn=(9S4c=920-b54f-elda-2ef<!-61b54S9!>8e<n [LDAPFullSync]
coB.sun.jndi.ldap.LdapReferralExcepticn: 1U3AP: e"Or =ê 10 -
ReEErr: DSID-031OO6E0, data 0. 1 access points
I H 'lab.con'
- rmmaioing cnOBB,(Ic-J.b.co"
MESSAGE [LEAP: error code 10 - 0000202B: RefErr: DSID-O310Q6EO,
-2ef
921)
poi
tat 1: 'lab.OEPB'
In the first example, LDAP Maiiater Distinguished Name has been misconfigured. Ihe last
component of the canonical string dc lab.com is incorrect. It should be dc-lab,dc-com
Because ofthis error, the Cisco Unified Communications Manager reports the login failure
immediately because an administrator tried to save this incorrect synchronization agreement.
Similarly, if the password would be incorrect, the same login failure would appear on the page.
Troubleshooting the LDAP User Search Base is abit more complex because its failure is not
caught dunne the configuration process. You have to activate tracing. Navigate to the Cisco
Unified Communications Manager Serviceability page. For the publisher, choose Trace >
Configuration >Directory Services >DirSync. Set the DirSync traces to Debug.
Then attempt to perform the complete synchronization from the LDAP Directory configuration
page The second example shows apart of the tracing output that indicates the LDAP User
Search Base problem. Like the previous case, the last component of the canonical string
dcMab.com is incorrect.
Cisco DirSync Service Parameters
Cisco DirSync can betuned byusing service parameters.
isco Uirbync Service P
System >Service Parameters >DirSync
BOirSynt (AttitfF) r
it.i.s; u<tiv<>-
CkHUimdt fimill.l OarwtUn Ht elf to r.>r>)
c =lJ-Î=-.BJ;ai-*
-' -'---' "- II *
0ua<i*4 faJot
Choose System >Service Parameters in Cisco Unified Communications Manager
Administration. Then choose aserver in the Server drop-down menu. Choose the Cisco
DirSync service in the Service drop-down menu. The Service Parameter Configuration window
that displays allows configuration ofthe DirSync service parameters:
Maximum Number ofAgreements specifies the maximum numbers ofagreements that can
be configured from the GUI.
Maximum Number of Hosts specifies the maximum number of hosts that can be configured
for failover purpose.
Retry Delay on Host Failure (sees) specifies the delay that is used in retry logic in case of
an LDAP connection failure.
Retry Delay on HostList Failure (mins) specifies the delay that is used in retry logic in case
of an LDAP connection failure. Unlike Retry Delay on Host Failure, the delay is applied
when the retry starts overagain on theentire host list.
LDAP Connection Timeout (sees) specifies the timeout period (in seconds) that is used to
establish the LDAP connection. The LDAP service provider aborts the connection attempt
ifaconnection cannot be established in the specified timeout period.
Delayed Sync Start Time (mins) specifies the delay that is applied before starting a
synchronization process when the Cisco DirSync application starts.
Setting Up Default Password and PIN for Synchronized End
Users
The Cisco Unified Communications Manager docs not synchronize ActiveDirectory
passwords.
tor Synchronized End Users
Microsoft Active Directorypasswords never synchronized.
- Set the default password and PIN.
I.r*tf*fHt*l PolUT DtfU tnloi
End-User Password i :'''*-'-"
End-User PIN
Cisco Unified Communications Manager has noknowledge of the Active Direciory encryption
mechanism. In Cisco Unified Communications Manager, adefault Credential Policy
mechanism is used. At installation, Cisco Unified Communications Manager assigns a static
Default Credential Policy to user groups. It docs not provide default credentials. The system
docs not support empty (null) credentials. You must configure end-user default credentials
immediately after installation, oruser logins will fail.
You can configure Credential Policy from the Cisco Unified Communications Manager
Administration pages at User Management >Credential Policy Default. Here, you can
configure cither the default end-user password or the PIN.
2-100 Troubleshooting Cisco Unified Communications (TVOICE] v8 0
Some End Users Not Synchronized
This section describes the issue when some endusers arenotsynchronized from theLDAP
directory.
Some End Users Not Synchronic
n=-=. *^*c dent
-J=*
I
*-* -wad
The UserPrineipalName(UPN)
attribute is guaranteed by Microsoft
Active Direciory to be unique across
the forest.
Asynchronization agreement for a
domain willnot synchronize users
within a child domain.
Separate agreements are required.
I I I
'mnttn hhjwii BtUri
wftiQ w***& w&t
I I I
aBUwi ="k*m OltWI
Ifsome end users arc not synchronized from the directory, consider the following:
Asynchronization agreement for a domain will notsynchronize users outside of that
domain nor within achild domain. Separate synchronization agreements are required to
import all users in this example. Although User Search Base 1specifies the root
dc=vse,dc=info, it will not import users that exist in the child domain
dc=amer,dc=vse.dc-lab.
When synchronization is enabled with an Active Directory forest that contains multiple
trees, multiple synchronization agreements are needed (two in this example). The
UserPrineipalName (UPN) attribute is guaranteed by Active Directory to be unique across
the forest, and the UPN must be chosen as the attribute that ismapped to the Cisco Unified
Communications ManagerUser IDinthis case.
) 2010 Cisco Syslems. Inc
Resolving Authentication Issues in Cisco Unified
Communications Manager Using Active Directory
This topic explains the procedure to identify and resolve authentication issues with Cisco
Unified Communications Manager when it is integrated with Active Directory.
's A t.
Most synchronization resolutions applyalso for
authentication.
Verify LDAP Authentication configuration.
Ifenabled, whether credentials are correct
Whether user search base and LDAP server address and
port are correct
nforEiHllJirn-
You canenable the LDAP authentication function independently of the LDAP synchronization
function. However, if you enable authentication alone, the user IDs inCisco Unified
Communications Manager will match the user IDs that are defined inthe corporate directory.
To avoid the high poteniial for error, it is recommended that you combine LDAP authentication
and LDAP synchronization.
The following statements describe Cisco Unified Communications Manager behavior when
LDAP authentication is enabled:
End-user passwords are authenticated against the corporate direciory.
End-user passwords arc managed in LDAP, not in Cisco Unified Communications
Manager,
End-user passwords arc only stored in LDAP; they are not replicated to Cisco Unified
End-user PINs andother Cisco Unified Communications Manager usersettings are
configured and stored in Cisco Unified Communications Manager only.
As a best practice, adedicated service account should be used in the LDAP directory that is
exclusively used by Cisco Unified Communications Manager for interacting with LDAP.
Verify that the account has the same credentials in Active Directory and Cisco Unified
Verify also that the LDAP User Search Base that is configured at the Cisco Unified
Communications Manager points tothe correct Active Directory path.
2-102 Troubleshooting Cisco Unified Communications (TVOICE) v80
Make sure that the IP address or hostname and port numbers are correct. When you enable
LDAP authentication with Active Directory, an Active Directory global catalog server at port
3268 should be used for faster response times.
2010 Cisco Systems. Inc. Cisco Unified Communications Manager Troubleshooting 2-103
Troubleshooting LDAP Authentication
Troubleshooting LDAP authentication is not as easy as that for synchronization.
2-104
riI*&&*4.A&l*t*lttA4< ,
ri TJ*?tW^
End users cannol connect to Cisco Unified CommunicationsManager user
pages after LDAP authentication is enabled
Use a sniffer to get the conversation traffic between Cisco Unified
Communications Manager and the LDAPserver
Cisco Unifred
Corrmunicalion
Manager
Search1 ^doe
Base dc=awid. dc=mto
$
i*
o
~, #.=*, *=*
B P
On<tfim
/ l\
Users might complain that they cannot connect to CiscoUnified Communications Manager
user pages or other Cisco UnifiedCommunications applications that authenticatethrough Cisco
Unified Communications Manager after ihey enable LDAP authentication. To remedy this
problem, use a packet sniffer to trace ihe conversation traffic between Cisco Unified
Communications Manager and the LDAP server.
The figure illustrates the authentication process for the first user, whose UPN is
jdoei'iiavvid.info;
1. The user authenticates to Cisco Unified Communications Manager via HTTPS with its
username (which corresponds to the UPN) and password.
2. Cisco Unified Communications Manager performs an LDAPquery against an Active
Directory global catalog server by using the username that is specified in the UPN
(anything before the (a sign) and deriving ihe LDAP search base from the UPN suffix
(anything after the hi sign). In this case, the username is jdoe and the LDAP search base is
"de-avvid. de-info."
3. Active Direciory identifies the correct Distinguished Name that corresponds to the
username in the tree that is specified by the LDAP query, in this case, "cn=jdoe. ouÛsers.
dc-avvid. dc-info."
4. Active Directory responds via LDAP to Cisco Unified Communications Manager with the
complete Distinguished Name for this user.
5. Cisco Unified Communications Manager aticmpts an LDAP bind with the Distinguished
Name provided and the password that was initially entered by the user, and the
authentication process then continues as in the standard case.
*
Summary
Summary
Cisco Unified Communications Manager supports LDAP
synchronization and authentication.
; When resolving synchronization issues, start verifying
network connectivity, check servicesand service account
verify the Cisco Unified Communications Manager LDAP'
systemand directory configuration.
Most ofLDAP synchronization resolutions apply also for
authentication. In addition, you can usea packet sniffer to
inspect the conversation traffic between Cisco Unified
Communications Manager andthe LDAP server
This lesson explained how to troubleshoot LDAP synchronization or LDAP authentication
issues when integrating Cisco Unified Communications Manager with LDAP.
References
Cisco Systems. Inc. Cisco Unified Communications System 8.x SRND April 2010
Imp. u.asco.com en US docs\oiee _ip_.comm/eucm/srnd/8x/uc8x.html
' *7 SyHS?' ?L- CiSC Umfied Cm<ôns Manager Administration Guide
Release 8.0(1), February 2010.
http: wwu.cisco.com enUS.docs/voiccjp comm/cucm/admin/8 0 1/ccmcfV
nccm-8()l-cm.hnnl a
LDAP directory documentation of individual supported vendors
'2010 CiscoSystems, Inc.
Cisco Unified Communications Manager Troubleshoofing 2-105
2-106
9m
Module Summary
This topic summarizes the key points that were discussed in this module.
lodule Summary
Endpoint registration issues are some of the most common
issues found inCiscoUnified Communications Manager. When
MGCP gateways fail toregister, itis usually duetogateway
configuration errors, connectivity errors, or Cisco Unified
Communications Manager configuration errors.
If CiscoUnified Communications Manageris unavailable on the
network, ensurethatthe appropriate services are not stopped,
deactivated, unresponsive, or needing to be restarted.
For database replication errors in Cisco Unified
Communications Manager, use the CLI torecreate the Cisco
UnifiedCommunications Manager publisher-to-subscriber
relationship.
The directory in Cisco Unified Communications Manager default
is inthe database. However, LDAP synchronization and
authentication can be enabledwith select third-party LDAP
services. Issues will typically arise fromconfiguration or
connectivity problems.
In this module, you learned how to isolate and troubleshoot reported issues that relate to Cisco
References
Cisco Systems, Inc. Sample ofDebug MGCP Packets, April 2008.
http: cisco.com en'US/products'sw,voiccsW''ps556/
pruducts_tccli_notcUQ186a0080174804.shliiil
Cisco Systems, Inc. Troubleshooting Guidefor Cisco Unified Communications Manager,
Release8.0(1), DeviceIssues. February2010
http: uww .ciseo.com en 'US 'parmer'does/voice ..ip .comm/cucm/trouble''
S 0 1 tbde\ice.html
Cisco Systems. Inc. Troubleshooting Guidefor Cisco Unified Communications Manager,
Release 8.0(1). Cisco Unified Communications Manager System Issues. February 2010
http- www.cisco.com en US partticr/docs.voice_ip_comm/cucm/troiiblc/
S_0_1 tbs; stem,html
Cisco Systems, Inc. Cisco Unified Communications System 8.x SRND, April 2010.
http: wv\w.cisco.conven.US/'docs'voice ip comm/cuem/smd/8\/uc8x.html
CiscoUnified Communicalions Manager Troubleshooting 2-107
Cisco Systems, Inc. Cisco Unified Communications Manager Administration Guide
Re/ease 8.0(1), February 2010.
Imp: uww.cisco.com en US docs voice ip comm/cucm admin X(i I
cemefg hccm-S01-cni.html
LDAP directory documentation ofindividual supported vendors
2010Cisco Systems lr
Module Self-Check
Ql) Which area would you check first ifyou discovered that your Cisco IP phone was not
able to obtain inline power? (Source: Troubleshooting Common Gateway and Endpoint
Registration Issues)
A) Windows Network Monitor
B) Cisco Unified Communications Manager
C) Cisco Unity Connection
D) Cisco switch configuration
Q2) Which four statements describe how touse Cisco Unified Communications Manager to
troubleshoot endpoint connectivity? (Choose four.) (Source: Troubleshooting Common
Gateway and Endpoint Registration Issues)
A) Use the pingcommand from a server that is in theCisco Unified
Communications Manager group toverify connectivity to a CiscoUnified IP
phone.
B) Use theCisco Unified Reporting tool to inspect the state of theconnectivity to
the Cisco Unified IP phone.
C) Verify thatboth theCiscoCallManager andtheCiscoTFTP services arc
activated and started.
D) Verify that the DHCP server is configured at the Cisco switch thatis facing the
Cisco Unified IP phone.
E) Makesure that thecorrectTFTPoption 150has been offered to the Cisco
Unified IP phone.
^ F) Use the Cisco Unified RTMT to verify that the Cisco Unified IP phone points
to the correct TFTP option 150.
G) Restart the CiscoUnifiedCommunications Managerpublisherserver to restore
t endpoint connectivity.
*" II) Locate the phone that has anissue and recreate its configuration file from
Cisco Unified CM Administration.
~** Q3) Which five network and device configurations would you display ona Cisco Unified IP
^* phone when resolving endpoint connectivity issues? (Choose five.) (Source:
Troubleshooting Common Gatewayand Endpoint Registration Issues)
^"^ A) phone MAC address (host name)
*r B) PC port status
C) domain name
D) VLAN ID
^ E) DHCP server status
F) IP address, subnet mask, default gateway
G) TFTP server address
H) inline power level
^ I) Unified CM configuration
J) Messages URL
^^ 2010 Cisco Systems. Inc. Cisco Unified Communications Manager Troubleshooting 2-109
0-4) Put the following steps, describing MGCP gateway registration, in the correct order.
(Source: Troubleshooting Common Gateway and Endpoint Registration Issues)
A) TCP connection is opened.
B) RQNT is sent by the Cisco Unified Communications Manager and ACK issent
back in response.
C) RSIP is sent by the gateway andACK is sent back inresponse.
D) The gateway boots.
E) AUEP issent byiheCisco Unified Communications Manager and ACK issent
back in response.
F) '['he IP stack is initialized through static or DlICP settings.
1. Step 1
2. Step 2
3. Step 3
4. Step 4
Step 5
s
6. Step 6
Q5) When troubleshooting Cisco MGCP gateways, the command _
is a good place tostart. This command displays thestatus of theregistration of the
Cisco IOS gateway to theprimary Cisco Unified Communications Manager server.
This command also displays any configured backups, aswell as other configurable
settings. (Source: Troubleshooting Common Gateway and Endpoint Registration
Issues)
06) Which statement best describes the most probable cause ofan H.323 gateway showing
a registration status of-unknown"? (Source: Troubleshooting Common Gateway and
Endpoint Registration Issues)
A) Network connectivity between the gateway and the subscriber is preventing
registration.
Hi 11.323 settings are incorrect.
C) The"unknown" status indicates that themodel of gateway is not a default for
theversion of Cisco Unified Communications Manager that is installed.
D) Thegateway is not configured withthe correctdial peers.
L) H.323 gateways will never register with Cisco UnifiedCommunications
Manager due to the peer-to-peer nature of the protocol.
Q7) Which four of the following are the most likelysources of CiscoUnified
Communications Manager not responding to endpoint requests? (Choose four.)
(Source: Troubleshooting Cisco Unified Communications Manager Availability Issues)
A) There couldbean MGCP timeout that is set to a largevalue.
B) The Cisco UnifiedCommunications Manager server could have crashed.
t I The Cisco Unified Communications Manager server could have fallen into
standby mode.
D) Cisco I FTP service performs internal cleaning.
L) CiscoCallManager servicecouldfreeze or couldfail to start completely.
I I The memory garbage collector has started.
G) Aprocess inCisco Unified Communications Manager could beconsuming too
much of the CPU cycles, or the server could be heavily overused.
Ill Aprocess witha memory leakcouldcause a shortage of systemmemory.
2-110 Troubleshooting CiscoUnified Communicalions (TVOICE) v8.0 2010CiscoSystems. Inc
B**-
Q8) Which two CLI commands are used to view the status of Cisco Tomcat and start it if it
has stopped? (Choose two.) (Source: Troubleshooting Cisco Unified Communications
Manager Availability Issues)
A) show service list
B) utils service start Cisco Tomcat
C) utils start Tomcat
D) utils service list
E) utils service Cisco Tomcat
Q9) Which three of the following arepotential causes of slow server response? (Choose
three.) (Source: Troubleshooting Cisco Unified Communications Manager Availability
Issues)
A) firewall or access list ineffectin thepath
B) incorrect dial plan design at the Cisco Unified Communications Manager or
gateways
C) Cisco Unified IPphones using outdated, incompatible firmware
D) high CPU usage, a malfunctioning process, ora memory leak
E) Cisco CallManager service failure
F) speedand duplex mismatches between the serverand the CiscoUnifiedIP
phone
G) speed and duplex mismatches between the server and the switch
Q10) Cisco Unified Communications Manager nodes replicate all information including run-
lime data ina mesh topology (that is, every node updates every other node). (Source:
Troubleshooting Database Replication Issues)
A) true
B) false
Ql1) The ReplicatcState object represents the state ofdatabase replication as anumerical
value. Match the values with their meaning. (Source: Troubleshooting Database
Replication Issues)
A) 0
B) 1
C) 2
D) 3
E) 4
1. This value indicates that the replication setup did not succeed.
2. This valueindicates that replication is badin the cluster.
3. This value indicates thatreplicates have been created, but theircount is
incorrect.
4. This value indicates that replication isgood.
5. This value indicates that replication did not start. Either no subsequent
nodes (subscribers) exist, or theCisco Database Layer Monitor service is
not running and has not beenrunning since the subscriber was installed.
2010 Cisco Systems. Inc cjsco Unified communications Manager Troubleshooting 2-111
Q12) If database repair didnot help andsuspects remain, youmight needtoreset the
replication. Which twoconditions should beconsidered when resetting thereplication.'
(Choose two.) (Source: Troubleshooting Database Replication Issues)
A) if any subscribers still show a replication status of 0
B) if any subscribers still showa replication statusof 4
C) if the publisher still showsa replication statusof 4
D) if the status of 0 is seen for more than 2 hours
E) if the status of 0 is seen for more than 4 hours
Q13) With which ihree LDAP services canyou synchronize Cisco Unified Communications
Manager Release 8.x? (Choose three.) (Source: Troubleshooting LDAP Integration
Issues)
A) Microsoft Active Directory
B) DC-Directory
C) MS SQL 2000
D) Netscape Directory Server 4.x
h) Informix Directory
F) OpenLDAP 2.3.39 and 2.4
G) ADS
Q14) Which two pori numbers can be used when integrating Cisco Unified Communications
Manager with Active Directory? (Choose two.) (Source: Troubleshooting LDAP
Integration Issues)
A) 369
B) 16387
C) 389
D) 3168
E) 3268
Q15) Asynchronization agreement for adomain will not synchronize users outside that
domain or within a child domain. (Source: Troubleshooting LDAP IntegrationIssues)
A) true
B) false
Q16) Which three statements describe Cisco Unified Communications Manager behavior
when LDAP authentication is enabled? (Choose three.)(Source: Troubleshooting
LDAP Integration Issues)
A) End-user passwords areauthenticated against the Cisco Unified
Communications Manager database andPINs arcauthenticated against the
corporate directory.
B) End-user passwords are authenticated against the Cisco Unified
Communications Manager database.
C) End-user passwords areauthenticated against thecorporate directory.
D| End-user passwords are only stored inCisco Unified Communications
Manager and there isno need for them to bereplicated from the corporate
directory.
L) End-user passwords are only siored in the corporate directory; they arc not
replicated toCisco Unified Communications Manager.
F) End-user PINs and other Cisco Unified Communications Manager user settings
arc configured and storedinthe corporate directory.
G) End-user PINs and other Cisco Unified Communications Manager user settings
arcconfigured and stored inCisco Unified Communications Manager.
2-112 Troubleshooting Cisco Unified Communicalions (TVOICE) v8.0
on D
Q2>
A. C. F. H
Q3)
A. D. KG. I
Q4) 1-D.2-F. 3-A. 4-C. 5-E, 6-B
05) sho\ ccm-manager
06) V.
Q7) B.E.G.H
08)
B.I)
Q9) B. 0, G
010) B
QUI
,\-5.B-3,C4,D-2, L-l
012) B.F.
Q13) A. D, F
QI4) C.h
01?)
A
016) C. E. G
)2010CiscoSystems, Inc. CiscoUnified Communicalions ManagerTroubleshooting 2-113
Troubleshooting CiscoUnified Communications (TVOICE) v80 2010 CiscoSystems.
Module 3
Troubleshooting Call Setup
Issues
mm Overview
This module discusses four possibletroubleshooting areas inwhichyoumay have call setup
issues. The module looks at troubleshootingon-premises single-site calls, on-net multisite calls,
including intercluster calls, and off-netcalls. The module also discusses thedifferent topology
attributes for completing a call and the different tools and commands bothon gateways and on
the Cisco Unified Communications Manager.
Module Objectives
Uponcompleting this module, youwill be able to diagnose a call setupissueand resolve the
issues as youdiscover or reveal them, given a trouble call for which the source of theproblem
is unknown. This ability includes beingable to meet these objectives:
Describe commonly experiencedcall setup issues including call setup failure, caller ID
issues, inefficient routing, and redirecting number issues and the most likely causes of these
issues
Explain the common callingissuesthat can occur in a single-site Cisco Unified
Communications Managerdeployment and identifythe most likelycausesof these issues
Explain the common calling issues that can occur in a multisite Cisco Unified
Communications Manager deployment and identify the most likely causes of these issues
Explain the common callingissuesthat can occur with off-netcalls and identify the most
likelv causes of these issues
Troubleshooting Cisco Unified Communications (TVOICE) v8.0 2010 Cisco Systems Inc
Lesson 1
Examining Call Setup Issues
and Causes
Overview
When a call is set up in a Cisco Unified Communications system, configuration errors can
cause problems in many areas. Some ofthese areas include router configurations, Cisco Unified
Communications Manager configurations, quality ofservice (QoS) settings, ISDN settings, and
carrier issues. This lesson provides anoverview of some of theissues thatcanarise with call
setup ina Cisco Unified Communications system.
Objectives
Upon completing this lesson, you will be able to describe commonly experienced call setup
issues including call setup failure, caller ID issues, inefficient routing, redirecting number
issues, and the most likely causes ofthese issues. This ability includes being able to meet these
objectives:
Define the major issue types that cause call setup failure intheCisco Unified
Describe the common reasons that calls can fail to set up in asingle-site environment
List major causes of why calls thatareplaced within a Cisco Unified Communications
Manager cluster can fail
Describe the common causes ofcall setup failure between clusters, including the call
control discovery
Call Setup Issues
This topic defines the major issue types that cause call setup failure in the Cisco Unified
Communications system.
lypicai oaii setup issues
Typical call setup issues include:
' Fast-busy tone dunng and at the end of dialing
- Missing or incorrect caller ID
Nonngback
- Dead air
One-way calling
< Inefficient call routing
' Unexpected second dial tone
Call semp issues can take many forms, including the following:
The call fails to complete. This problem ischaracterized by a tast-busy tone during orafter
dialing.
The caller ID is missing or incorrect.
When placing acall, public switched telephone network (PSTN) orIP phone callers do not
hear aringback. Other issues can occur when transferring acall from an IP phone; such
issuescancause a PSTN caller who is beingtransferred not to hear ringing duringthe
transfer.
Dead air issues can occur. Both parties who areinvolved ina call hear nothing (ncithei a
fast-busy tone norvoice) when thecall is answered,
One-way voice issues can cause voice tobeheard in only one direction.
Inefficient call routing can occur when mink orWAN link saturation or hairpinning occurs.
Second-dial tone issues include the lack of a second dial tone when it isdesired or a second
dial tone beinn heard after ihe wrong digit.
Troubleshooting Cisco Unifed Communications (TVOICE) vB.O
) 2010 Cisco Systems Inc
Single-Site Call Setup Failure
eT,mroTmenrCribeS ^ ^^"^^ ^ " to SCt "P in *-"gle-sitc
Typical Single-Site
Common Cisco Unified
causes:
Incorrect digit manipulation
Unknown or unregistered
target
Common endpoint causes:
CoSsettmgsdefined inCisco
Unified Communications
Manager
Common call setup problcms can occur when call setup fails within asingle site
or CT, ports, and the Cisco Unified clmun.catls Managef ndP""S' V<"CC-mai'
include the cla 0 erv^TcoS ^ttm fon *"H " **" ' ^ ThCSC Ktti"ES
Mg,r;atoCrmUmCa,'0nS Ma"agCr "" "" " " *ftil <*' s."6le s.te for the
Atranslate patten, is configured with incorrect digit manipulation
The des,iat, mlgt no, be registered , the Cisco Unified Communications Manager
Calling privileges might be misconfigured.
^rcî::dZ,,rdpmms w,ih,n asmgie si,eôf coS seigs
2010CiscoSystems, Inc.
Troubleshooting Call Setup Issues 3.5
Intracluster Call Setup Failure
This topic lists the major causes of why calls that are placed within aCisco Unified
Communications Manager cluster can tail
3-6
SU<
<y
Cisco Unified
Communications
Manager
;v
^
Common endpoint causes:
- Codecmismatch between endpoints
defined by regions
- CoS settingsdefinedin Cisco
Unified CommunicationsManager
/
V
_L:
HQ
Branch
Common Cisco Unified Communications Manager causes:
. Incorrect digit manipulation on Cisco Unified Communicalions Manager
- Unknown or unregistered target
. Lack of MTP resources
WAN link. These problems are the most common call setup problems.
The following endpoint settings can cause call setup to fail:
^cXmt==Cr^ôêndpoint
device pool.
. CoS setfin.s on the endpomts: In Csco Unified Communications Manager, you manage
these settings by creating and applying partitions and CSSs.
The Cisco Unified Communications Manager route plan can cause the call io fail for these
reasons:
. Incorrect diait manipulate that is used in atranslation pattern.
. The destination might not be registered to the Cisco Unified Communications Manager.
AMcdta Termination Point (MTP) ,s required for the call, but no MTP ,s available.
Troubleshooting Cisco Unified Commun
ications (TVOICE) v8 0
lultisite Intracluster Call Setup fssyes (Cont.
Cisco Unified
Communications
Manager
Common gateway causes:
Misconfigured gateway or issues on PSTN
connections can prevent calls.
Improper QoS configuration drops or delays
signaling.
* Lack of gaieway DSP resources prevents the
call when a codec mismatch makes transcoding
between endpoints necessary.
Common WAN causes:
WAN link provider is not complying wiih SLA.
> WAN provider is having problems.
Locations of RSVP-based CAC prevent ltie call
The settings on the WAN link that could cause the call setup issue are as follows:
The carrier may be having problems or is not complying with the service level agreement
(SLA).
QoS issues on the WAN link may be causing the call setup messages to be dropped or
severely delayed. A compressed Real-Time Transport Protocol (cRTP) mismatch on the
two ends of a point-to-point WAN link would cause call failure. This cRTP mismatch
would occur when only one side of the point-to-point connection has cRTP configured.
Locations-based or Resource Reservation Protocol (RSVP)-based Call Admission Control
(CAC) mechanisms have prevented the call and automated alternate routing (AAR) is not
configured or is not configured properly.
The settings on the gateway that could cause the call setup issue are as follows:
Various PSTN and gaieway issues, if the two sites are connected by using PSTN as a
backup to the primary IP WAN. These issues wilt be covered in a separate lesson.
Class of restriction (COR) on the router may be restricting the call from being set up to an
analog device.
The lack of a digital signal processor (DSP) resource prevents two endpoints that are using
dissimilar codecs from connecting a call.
'2010 Cisco Systems, Inc.
Troubleshooting Call Setup Issues 3-7
Intercluster Call Setup Failure
3-8
This topic describes the common causes of call setup failure between clusters, includingthe cal
control discovery.
/ ^y -^y
Cisco Unified
Communications
Manager
&&4 &
1&J
mLW
IP WAN
BR-1
1
Cluster Ay
Common Intercluster
Local-end CoS
Re moIe=-end CoS
ICT settings
SIP Irunk settings
cRTP mismatch
QoS issues with signaling
across WAN
Galekeeper CAC mechanism
Cisco Unified Border Element
Call Control Discovery
Cluster B
This figure describes common call setup failure causes when call setup fails between Cisco
Unified Communications Manager clusters.
Calls between two Cisco Unified Communications Manager clusters can fail for these reasons:
The local-end CSS docs not include the route pattern partition. If these settings are
incorrect, they can prevent call setup.
The remote-end trunk CSS does not include the route partition of the endpoint and the call
is prevented from being delivered.
The intercluster trunk (ICT) settings might be configured incorrectly.
If a Session Initiation Protocol (SIP) trunk is used, the settings on both ends might be
configured incorrectly.
A cR'I P mismatch might exist on the WAN link between the two sites.
QoS issues (includes cRTP) or lack of QoS across the WAN link that connects the two siics
can cause drops, latency, or jitter in ihe call setup messages.
The gatekeeper CAC mechanism might prevent the call because of bandwidth saturation.
Issues of Cisco Unified Border Element might prevent the ititer-clustercall if Cisco Unified
Border Element is used to interconnect the clusters.
Call Control Discovery (CCD), if used, might be misconfigured as follows:
It could be advertising or requesting a service at a cluster that might not be
associated with an\ trunk.
CCD-related route partitions might not be included in the calling device CSS.
There may be Service Advertisement Framework (SAF) issues, such as neighbors
are not formed, SAF clients are not registered, and so on.
The service parameter CCD MaVimum Number of Learned Patterns might besettoo
low,
The Learned Patterns blocking filter might be configured to block some learned
patterns, and so on.
CCD issues are discussed in a separate lesson in detail.
)2010 Cisco Systems, Inc. Troubleshooting Call Seiup Issues 3-9
Summary
IH(\'
Call setup issues might be caused by many different causes
that are specific to the topology.
When calls fail to set up properly within a single site, the
cause is usually a CoS setting, a problem with translation
patterns, invalid destination, or a codec mismatch.
When call setup fails with an intracluster call across a WAN
link, the cause could be CoS settings, translation patterns,
invalid number, codec mismatches, CAC mechanisms, a lack
of DSP resources, or a WAN issue.
Calls between clusters can fail due to WAN issues, CAC
mechanisms, CCD issues, QoS issues, compression
mismatches, remote-end CoS, local-end CoS, or incorrect
trunk settings.
In this lesion, you learned to describe commonly experienced call setup issues including call
setup failure, caller ID issues, inefficient routing, redirecting number issues, and the most likely
causes of these issues.
3-10 Troubleshooting Cisco Unified Communications (TVOICE) v8.0 ) 2010 Cisco Systems. Inc.
Lesson 21
_ Troubleshooting On-Premises
_ Single-Site Calling Issues
* Overview
^^:zz^^:^t^hTcn two dw'th'-thc<*-
translation patterns. CoS in C, souX, CoZ T^ ?** f Servioc <CoS> setti"B* -
partitions and calling search snace^S^ 7 T Managcr takeS the form of
way eaNmg, forward ^" al LTil f T""""^ '" **"' *** *"
problems, you need to know how Cisco V^r ^ând *oub'ôot these
analyzes digits. ^ ComKKs Manager collects and
Objectives
Describe typical reasons for call setup to fail for .premises ca,ls
Revrew how Csco Unified Communications Manager coHects and analyzes digit.
' X^' "C'SC0 Un'fled C"M-ager uses partittons and CSSs ,
Troubleshoot common causes of single-site call setup failures
" S;""* ***causes f one-way cal, setup as aspecial case of smgle-site ca setup
Exp, the pss,b,e causes and recommended actions in the even, of aca-forward,g
On-Premises Call Setup Issues
Thts topic describes typical reasons for call setup to fail for on-premiscs calls.
3-12
0 n - P f 6 *' 'ni s fc t* *<' ^
Csco CRS
i
Most common on-premises issues
CoS misconfigured
incorrect digit manipulation
Unknown or unregistered target
- Voice-mail orCTI ports not registered
= Ports busy
Single
site
This lesson will discuss only the calls that take placemen devices at the same geographical
site and ,n the same Csco Unified Communications Manager cluster.
These issues arc the most common while callmg on-premiscs:
. CoS is miseonfieurcd in Cisco Unified Communications Manager.
. Digu manipulation is misconfigured: for example, by using translation patten..
The target is unknown or unregistered, like an IP phone.
voice-mail or computer telephony integration (CTI) ports arc not registered with Csco
All \oice-mail portsare busy.
Troubleshooting
Cisco Unified Communications (TVOICE) v8 0
2010 Cisco Syslems, Ire
Digit Collection in Cisco Unified
This topic reviews how Cisco Unified Communications Manager collects and analyzes digits.
lanager Caii-Routing Logic
Cisco Unified
Communications
Manager uses closest-
match logic to select
the best pattern.
When multiple
matching patterns are
present, the best
pattern is the pattern
that meels this criteria:
Matches the dialed
string
User A
dials1200
UserB
dials1212
Matches the fewest
strings otherthan Usei c
the dialed string d,aisi234
1XXX
12XX
121X
1234
Reachability of internal destinations is provided by assigning directory numbers to all
endpoints (such as IP phones, fax machines, and analog phones) and applications (such as
voice-mail systems, auto-attendants, and conferencing systems).
When anumber is dialed, Cisco Unified Communications Manager uses closest-match logic to
select which pattern to match from among all the patterns in its call-routing table.
In practice, when multiple, potentially matching patterns are present, the destination pattern is
chosen based on this criteria:
It matches the dialed string.
Among all the potentiallv matching patterns, it matches the fewest strings other than the
dialed string. For example, consider the case that is shown in the figure, in which the call-
routing table includes the patterns IXXX, I2XX, and 1234.
When user Adials the string 1200, Cisco Unified Communications Manager compares it with
the patterns in its call-routing table. In this case, there are two potentially matching patterns,
IXXX and 12XX. Both of them match the dialed string, but IXXX matches atotal of 1000
strings (from 1000 to 1999) while 12XX matches only 100 strings (from 1200 to 1299).
Therefore. 12XX is selected as thedestination of thiscall.
When user Bdials the string 1212, there are three potentially matching patterns, IXXX, 12XX,
and 121X As previously mentioned, IXXX matches 1000 strings and 12XX matches 100
strings. However, 121X matches only 10 strings; therefore, it is selected as the destination ot
the call.
Troubleshooting CallSetup Issues 3-13
I" prfdialSthermg l234'there are three potentially matching patterns, IXXX PXX
and 1234. As previously mentioned, IXXX matches 1000 strings and I XX matches 100
Digit Collection and Digit Analysis Overview
This section reviews low digits are collected and analyzed.
Digit Co flection and Digit Analysis
Device jW^JMîiilâMiwH*,]
HUH
SCCP Digit by digit
IP phone
SIP
En bloc
KPML
SIP dial rules
Gateway MGCP, SIP, H.323
En bloc
Overlap sending and receiving
Trunk SIP, H.323
En bloc
Overlap sending and receiving
This figure describes how digitanalysis is performed for different devices, based on their
addressing methods.
Cisco Unified Communications Manager docs not always receive dialed digits one by one.
Skinny Client Control Protocol (SCCP) phones always send digit bydigit. Session Initiation
Protocol (SIP) phones can use en-bloc dialing to send the entire dialed string atonce orKeypad
Markup Language (KPML) to send digit bydigit. If digits are received en-bloc, theentire
received dial string ischecked at once against the call-routing table.
The table shows the supported addressing methods in Cisco Unified Communications Manager
for different devices.
In SIP, en-bloc dialing can be used where the entire dialed string is sent in a single SIP INVITE
message, or KPML can beused which allows digits tobesent onebyone. SIP dial rules are
dial rules that are processed inside the SIP phone. Thus, a SIP phone can detect invalid
numbers and play a reorder tone without sending any signaling messages toCisco Unified
Communications Manager. Ifdialed digits match an entry ofa SIP dial rule, the dialed string is
sent in asingle INVITE message to Cisco Unified Communications Manager, IfCisco Unified
Communications Manager requires more digits, KPML can now be used to send the remaining
digits from the SIP phone to Cisco Unified Communications Manager one by one.
Trunks and ISDN PRI gateways can be configured for overlap sending and receiving, which
allows digits tobesent or received one byoneover anISDN PRI.
Digit-by-Digit Analysis
3-16
Cisco Unitied Communications Manager analyzes incoming dialed digits one by one as shown
in the figure.
Diuit An,
1000
^.J ?'aled Dlglts
^*~'~7r -j <None>l
-a-
1001
Call Setup J0._
x.
I
List Potential Matches
List Current Match
Route Patterns
1XXX
10XX
If an endpoint is sending dialed digits onebyone, Cisco Unified Communicalions Manager
immediately starts digit analysis when it receives the first digit. Infact, digit analysis starts
even one step before: when a phone indicates anoff-hook state toCisco Unified
Communications Manager. Cisco Unified Communications Manager looks up a null string
dialed number, which matches all available call-routing tables at this point.
By each additional digit that isreceived. Cisco Unified Communications Manager can reduce
the list of potential matches (that is, theentries of the call-routing tables that match the digits
received sofar). When a single entry is matched (like the directory number 1001), the so-called
-current match" is used and the call is sent to the corresponding device.
Troubleshooting CiscoUnified Communications (TVOICE] v8.0
Digit Collection Example
The figure illustrates the digit collection ofCisco Unified Communications Manager.
Digit Collection Exampl
User dial string:
1111
Cisco Unified
actions:
- No other patterns could
match, extend call
1111
121X
1J23JXX
1;
\3\0-Am
13!
- ". "I.
iii%i>-!k ""
Si'- .-="a"".- ^ "-..-
... Vs
Digit collection isstopped as soon as an entry in the call-routing table is matched in its
complete length and no other (longer) potential matches exist. In the example, auser dialed
llll, CiscoUnifiedCommunications Managerinterprets the numberdigit by digit. After
analyzing the first two digits, only one potential match is left (the first entry), because all other
entries inthe call-routing table require a difTerent digit than l at the second position. Cisco
Unified Communications Manager continues collecting digitsuntil it received four digits
(llll); now the first entry isfully matched and used toroute thecall.
Troubleshooting Call Setup Issues
Partitions and CSSs
This topic describes how aCisco Unified C'ommunications Manager uses partitions and CSSs
to manage calls.
Partitions and CSSs Review
- Apartition isa group of numbers with the samereachability.
Any dial able pattern (directory number, route pattern,
translation pattern, voice-mail port, Meet-Me conference
number, etc.) can be part of a partition.
ACSS is a list of partitions and includes the partitions that the
CSS can access:
Adevice can call only those numbers that are located in
the partitions that are part of its CSS.
ACSScan be assignedtoanyentity (including phones,
phone lines, gateways, and applications) that can
generate a call-routing request.
An IPphone can have a CSS configured at each line and
at the device; an effective CSS is a combination of line
CSS and device CSS.
Calling pri\ ileges are an imponant dial plan component. Calling privileges are used to
implement CoS so that not all users can access all call-routing table entries; depending on the
callingdevice or line, somedestinations are permitted, whereas othersare not. Partitions and
CSSs arc two major configuration elements of call-privilege implementation.
Apartition isa group of dialable patterns thai have similar accessibility.
Partitions arc assigned tocall-routing targets: any call-routing table entry, including voice-mail
ports, directory numbers, route patterns, translation patterns, and Meet-Mc conference numbers.
ACSSdefines whichpartitions are accessible to a particular device.
Adevice can call only those call-routing table entries that arc in partitions that are part of
the CSS of the device.
CSSs are assigned tocall-routing request sources, such as phone lines, gateways, trunks,
voice-mail ports, and applications.
On most call-routing request sources such astrunks, gateways, ortranslation patterns, you
can configure only one CSS.
However, on IP phones, you can apply a CSS once per line and once at the device level.
Troubleshooting Cisco UnifiedCommunications (TVOICE) v8 0
If both Ime and device are configured, then the CSS of the line from which the call is.placed is
considered first. The effectively used CSS is composed of the partitions that are listed in the
line CSS followed by the partitions ofthe device CSS:
The device CSS is usually used to provide call-routing information, such as which gateway
to select for all public switched telephone network (PSTN) calls.
Aline CSS can be used to block the route patterns that are prohibited by certain classes of
service, independent of the used PSTN gateway
>2010 Cisco Systems, Inc. Troubleshooting Call Setup Issues 3-19
Partition and CSS Considerations
Some special situations that arc related to partition and CSS exist in Cisco Unified
3-20
Partition and CSS Considerations
Thefollowing are important considerations:
Entities that are in partition <None> are always accessible
regardless ofwhether the calling entity has a CSS.
' ,Eh?S'inai h*are CS5 <None> ass'9ned can access only entities
that are in partition <None>.
*ACSS is an ordered list of partitions
Multiple identical entities can exist in the call-routing table but
must be in different partitions.
If no single best match exists, the call-routing table entry with
the partition that is listed first in the calling-device CSS is used.
Resulting route-selection priorities:
* Best match
- Partition order (if multiple, equally qualified matches exist)
By default, all entities that you can configure with apartition are in partition <None> also
ca ed the null partition. Members of partition <None> are always accessible by sources of a
call-routing request, regardless of the call-routing source CSS.
Similarly, all entities that you can configure with aCSS arc assigned CSS <Nonc> by default.
Entities that do not have an assigned CSS in other words, those entities that use CSS
<None> can access only those call-routing targets that are in partition <None>.
As by default, when no partitions or CSSs are assigned and all entities are associated with the
null partition and CSS <~None>. all calls are possible for all calling sources.
ACSS is an ordered list of partitions; the partition that is listed first has higher priority than a
partition that listed later. When Cisco Unified Communications Manager performs acall-
routing lookup, ,t considers all accessible entities ail targets that reside in apartition that is
listed mthe CSS of the calling phone and all targets that do not have an applied partition bv
best-match logic. y
If no single best match is found, then Cisco Unified Communications Manager uses the call-
routing tabic entry whose partition is listed first in the calling-device CSS.
In summary. Cisco Unified Communications Manager selects the entry of the call-routing table
according to this order:
1. Cisco Unified Communications Manager searches for the best match.
2. If multiple, equally qualified matches exist, then the order of the partition in the calling-
deuce CSS is ihe tie breaker, and the match is found in the earliest listed partition.
Device and Line CSS
This section describes the capability ofIP phones to be configured with adevice CSS and a line
CSSand howthese two interact witheach other.
Device and Line CSS
IPphones can have a CSS
configured at each line and
at the device.
CSS of the line from which
the call is placed is
considered first
Device CSS is then added
- Effective CSS consists of:
: Line CSS
;; Device CSS
Line CSS
Resulting CSS
Phii mH
ii "i 1 *
"A 1 \-
-Jlj
Id- I- *02
CoS can be implemented in different ways. When using the traditional approach ofCoS
implementation in Cisco Unified Communications Manager, external route patterns are placed
into partitions. CSSs are configured per CoS and applied to the respective phones. No CSSs are
applied to lines, and, therefore, the phone CSS applies to all lines. It may sound reasonable not
to use the separate line CSS because typically aphone should have the same privileges on all its
lines.
The traditional approach has no problems or disadvantages as long as itis used in asingle-site
environment inwhich all devices should use the same PSTN gateway for external route
patterns.
The traditional approach can result in many partitions and CSSs when applied to large multisite
deployments with centralized call processing. This configuration might be required (if local
route groups arc not used to simplify) because the device CSS is used to determine both the
path selection (which PSTN gateway to use for external calls) and the CoS.
It is possible to significantly decrease the total number ofpartitions and CSSs needed by
dividing these two functions between the line CSS and the device CSS, in what is called the
line device approach.
Based on how the line CSS and the device CSS for each given IP phone are combined, follow
these rules to implement the line deviceapproach:
Use the device CSS toprovide call-routing information (for example, which gateway to
select for all PSTN calls).
Use the line CSS toblock theroute patterns thatarenotallowed bya certain CoS
(independent of the usedPSTNgateway).
Create an unrestricted CSS for each site and assign it to the device CSS of the phone This
CSS should contain apartition that features route patterns that route the calls to the
appropriate local gateway per site.
Create CSSs that contain partitions that feature blocked route patterns for those types of
calls ihat arc not permitted by the CoS of auser and assign them to the lines of the phone of
a user.
On most sources of acall-routing request such as trunks, gateways, or translation patterns only
one CSS can be configured. On IP phones, however, aCSS can be applied per line and once at
the device le\el. It both are configured, the CSS of the line from which the call is placed is
considered lust. In other words, the effectively used CSS is composed of the partitions that are
listed inthe line CSS followed by the partitions of the device CSS.
On CTI ports, the line and the device CSS are placed in reverse order; the partitions of the
device CSS arc placed before the partitions of the line CSS.
3-22 Troubleshooting Cisco Unified Communicalions (TVOICE) v80
Time-of-Day Routing Review
This section reviews time-of-day (ToD) routing in Cisco Unified Communications Manager.
ToD Routing Review
Time and date information can be applied to partitions.
- CSSs that includesuch a partition have access to the
partition only if the current date and time match the time and
date informationapplied to the partition.
Based on time, different routing is allowed:
- Identical route pattern is putinto multiple partitions.
At least one partition has time information applied.
- If this partition is listed first inCSSs, it will take
precedence overthe other partition during thetime
applied to the partition.
Iftime does not match, the second partition of CSS is
used (the first oneis ignored because of invalid time).
ToD routing can be implemented in Cisco Unified Communications Manager by applying time
and date attributes to partitions by using time schedules and time periods. Time periods define
time ranges or dates and are grouped into time schedules. Time schedules are then assigned to
partitions.
ACSS that includes apartition, which is associated with atime schedule only, has access to the
partition if the current date and time match the time and date information that is specified in the
time schedule that is associated with the partition. Ifthe configured time schedule does not fall
into the current date and time, the partition is logically removed from the CSS.
ToD routing can be used toroute calls differently based on time in this way:
m Identical route patterns are created and put into different partitions.
At least one of these partitions has a time schedule applied.
Ifthe partition with ihe time schedule is listed first in CSSs, itwill take precedence over
otherpartitions while it is associated with the partition.
Ifthe current time does not match the configured time schedule, the partition that has the
time schedule assigned isignored and the next partition becomes the partition with highest
priority.
ToD Routing Example
3-24
This section presents anexample of ToD rout tin
Blocking International Calls on Weekends and January 1
W
CSS.
Weekend
Standard
nr] Dials
' 90114369918900009
Current Time: 20:00
Current Day: Sat
Route Pattern 9 011'
Pamtior- Standard
Route to PSTN
Partition Standard (no timeschedule)
FirsI parlilion (Weekend) is ignored
because no lime match. Pattern not
Firsl partition (Weekend) is active
matched and listed first Pattern
blocked.
Route Patlern: 9011!
Partition Weekend
Block This Pattern!
Partition Weekend Time Schedule: TS1
TimeSchedule TS1: Time Period' TP1, TP2
TP1. 00:00-2400 Sat-Sun
TP2' 00.00-24:00 Jan 1
ToD routing is configured by using and binding together several configuration elements.
Atime period specifics atime range that is defined by start and end times and arepetition
interval (days of the week or aspecified calendar date). One or more time periods are assigned
to atime schedule. The same time period can be assigned to multiple time schedules,
Atime schedule is agroup of time periods. Time schedules are applied to partitions and
therefore, make the partition inactive in aCSS when the applied time schedule does not match
the current date and lime.
The figure shows an example of how international calls can be blocked during the weekend and
on January I.
In this example. ToD routing is implemented by first creating aroute pattern, which allows
international calls. The route pattern is put into the standard partition, which has no time
schedule applied.
Asecond, identical route pattern is created, which is placed into the Weekend partition.
Atime period is configured for Saturday to Sunday 0:00 to 24:00. Another time period is
configured with aspecified date: January 1. These two time periods arc put into atime
schedule, and the time schedule is assigned to the Weekend partition.
Phones arc assigned with aCSS thai contains the Weekend partition first, followed by the
Standard partition.
Route patterns and translation patterns can be configured with the parameter Block This Pattern
to deny the call ifthe pattern was selected by the call-routing logic (best match, earlier listed
partition).
Phone users arc able to dial international calls at any time during aweekday (Monday through
Friday) because the Weekend partition is not active (based on the Standard partition).
<e 2010 Cisco Systems Inc
Now, where the route pattern that is in the Weekend partition is configured to block the call,
international calls are not possible whenever the Weekend partition is active (as it is listed
before the Standard partition in the CSS of the phones). Weekend partition is active on
weekends and on January 1.
2010 Cisco Systems, Inc Troubleshooting Call Setup Issues 3-25
Troubleshooting Single-Site Call Setup Failure
This topic describes hov. to troubleshoot common causes of single-site call setup failures.
3-26
Problem report:
An internal caller dials another four-digit internal IP phone and hears
a reorder tone
Consider the possibilities:
Atranslation-pattern issue exists.
The partition of the destination is not in the CSS of the caller.
The dialed number is currently unregistered.
The dialed number is an invalid number in the Cisco Unified
Communications Manager configuration.
M.3
W
HO-CIPC1
2001 ""-^F^5!i
If you experience a problem when making a call between two devices within the same cluster at
the same site, the two most likely causes are CoS settings and translation patterns.
CoS issues can manifest in several ways: one-way calling issues, forwarding issues, or call-
setup problems.
An internal caller dials another four-digit internal IP phone and hears a reorder tone.
An on-premises call can fail to set up for several possible reasons:
A translation pattern issue exists because of these reasons:
The CSS of the translation pattern does not have the partition of the transformed
target-dialed number.
A loop has formed because of the use of multiple translation patterns.
The transformation mask on the translalion pattern results in an invalid dialed
number.
The partition of the destination is not in the CSS of the caller.
The dialed number is nol registered and no Calling Forward on Unregistered (CFUR) target
is configured.
The dialed number is an invalid number in the Cisco Unified Communications Manager
configuration.
You can take these actions to resolve on-premiscs call-setup failures:
Use Cisco Unitied Communications Manager Dialed Number Analyzer to determine if a
trai^lation partem is being invoked and what changes are made if a translation pattern is
present. Fix any issues thai are discovered.
Troubleshooting Cisco UnifiedCommunications (TVOICE) v8.0
Use the Cisco UnifiedCommunications Manager Dialed Number Analyzer or the Cisco
Unified Communications Manager administrative web interface to determine whether
partitions and CSSare configured and whetherthosesettingsare blocking thecall. Such
behavior might be done byintention. Fixanysettings that are preventing thecall, assuming
that the call blocking is unintentional.
Use the Cisco Unified Communications Manager administrative web interface to search for a
directory number, and verify that the directory number is valid, or registered, or both.
2010 Cisco Systems. Inc. Troubleshooting Call Setup Issues 3-27
SCCP Call Setup Flow
This seciion demonstrates a call flow for a call setup between two li' phones using SCCP.
vziiA +~Si<?.!*<* Jm&&*i sf ^v ? i'SsuÂw *-i3i^jJji-s,i.'tt.i>ii'i^pSf 'K'fo*.>i~5.*
Calling IP phone Cisco Unified Communications Manager
V
I&!1
'-^l-
Mr, a;T.J:
'.;,- ;',...;
H..="\ i-.lBL>B : .v.
St C:. "e
h;=:- ^-lUve.-i
S'^.:T(.".- ':.iJls..,:.ir;ii ;;-.
._.
! ;k--...lb,i!i>,.,'
Called IP phone
The figure shows the exchange of messages between the callingCisco IPphone andCisco
The callingphonegoes ofthookand sendseachdialeddigit ina separate KeypadBuiton
message to Cisco Unified Communications Manager.
CiscoUnified Communications Manager collectsthe dialeddigits and matches thedirectory
number of the called Cisco IP phone.
TroubleshootingCisco Unified Communications ITVOICE) v8 0
mr
Hv
Hw
r*
iCCP Call Setup Flov* (Cont
Calling IP phone
Called IP phone
'-
CaliSlste
P tJallSlite
Calilnfo
SetRinger (InsroeRnsg)
Calllnfo
DisplayPiornpiStatus
Seler.tSoft.Keys
Ca'linto
0&!N>f*
D'-ilridNumber
S's'lTong (Alertioa;
CrlMSlrlSP
3etecT.S0UK.eys
pispiavPrtiTpSStaius (Rmqoul)
ClearNcitify
Se-tRmgor (RingOff) |
Cisco Unified Communications Manager notifies the called Cisco IP phone about the inbound
call and sets the ringer.
Cisco Unified Communications Manager sets the alerting tone at the calling IP phone and
updates thedisplay of thephone.
When the called IP phone goes offhook, Cisco Unified Communications Manager sets the
ringer of the phoneto off.
3-30
Calling IP prion
Cisco Unified Communicalions Manager Called IP phone
"W SH.,jr-p /inii.lus^ lU/.
(M-iNS;afc
1 SI:i}>T0r,e
CaMinSrj
!
".n ,>rftV
-I\r.\,
tii-|(5,.=S(,i;i<y
! V 'i-S'-i - i "!
;f.".'nK.;r<-lv(H:ii. nr.,,-1
; -
'- < =.!. " =!
i'imi i S:a;;M(j-jidTr,.n.,i!
2-way voice palh
The called IP phone line button lamp is activated; the display of the phone is updated.
Cisco Unified Communications Manager stops the lones at the calling and called IP phones and
starts with the media setup using OpeiiReceivcChannel and StarlMediaTransmission messages.
1he call setup is completed when the two-way voice path is established.
i 2010 Cisco Systems Inc
MP
**%
Tracing CSS Problems
This section explains the trace output with a focus onCSS troubleshooting. This section relates
to the scenario shown inthe previous figure, inwhich the IPphone HQ-CIPC2 with the
directory number2002placesa call to directory number 2001.
Tracing CSS Problems
Tracing to resolve CSS problems;
Enable Cisco CallManager service tracing.
Use CiscoUnified Communications Manager RTMT to display
SDI traces.
Place a test call between the affected phones and record the time
for debug-output reference.
Search for the recorded time inthe trace-output file; locate initial
events.
The trace output inthe next figure was simplified for clarity.
Route partitions inheriterror-handling capabilities fromthe CiscoUnified Communications
Manager software. Aconsole and system diagnostic interface (SDI) file trace areprovided for
logging information and error messages. These messages arepart of the digit-analysis
component of thetraces. You must know how the partitions and CSSs areconfigured and
whichdevicesare ineachpartitionand its associated CSSto determine the sourceof the
problem.
Enable detailed tracing ofCisco CallManager service totroubleshoot common causes ofsingle-
site call senip failures inCiscoUnified Serviceability.
Youcandisplaythe generated SDI trace filesby usingthe Cisco Unified Communications
Manager Real-Time Monitoring Tool (RTMT).
Startthetroubleshooting procedure by placing a test call between the twodevices thatare
reporting theproblem. Tofind thecall reference in thetrace files, record thetime of thistest
call. You must also time synchronize Cisco Unified Communications Manager because all trace
file timestamps will use the server time.
Display the trace file and locate the initial events that relate to the test call.
>2010Cisco Systems, Inc
NewCall soflkey button pushed
03/02/2009 14 48 53.871 CCW|Stationlnit. (0OQO0O3) SoftKeyEvent
softKeyEvert=2(NewCall| linelnstance-0 P\
caliReference=0 |<CUD .StandAloneCluster>NIDMP-1 5.2><CT::1,100,37,1.887><IP
1 ' ><DEV ' '. . . ><LVt .StateJTransitioiVxMASK ,0020>
Phone line goes off hook
TCP handler assigned to phone whon
registering Identifies trace output related
lo s smgle phone
03/02/2009 14 48.53 871 CCM|SlationD. (0000003) restart0_StationOffHook- INFO
Cl-Oon line=l, SPKMode=0, alwaysPrimeLme=0. fid=8S88,
offHookTngger=0 |<CLID ,StandAloneCluster><NID;:10.1.5.2><CT1.100,37.1.887><l
P 10 1 100 58><DEV::-0 . "' ,'><LVL;;Significant><MASK ;0020>
03/02/2009 14.46 53 871 CCW|StaIionD (0000003) reslartO_StationOffHook - INFO'
Cl=0online =1, SPKMode=0 New call |
<CLID SlandAloneCluslerxNID' 10 1 5 2><CT. 1,100,37,1.887><IP.. 10.1.100.58><D
EV ... " ><LVL SignificantxMASK. 0020>
When you openthe tracelog file inCiscoUnified Communications Manager RTMT, locatethe
initial caller phone event, as shown in ihe figure.
The caller at HQ-CIPC2 with IP address 10.1.100.58presses the NewCall softkey button.
Thetraceoutput alsoshowsTCP handlernumber that wasassigned to the IPphone duringits
registration. Thisnumber remains the same until the IPphone unregisters. This number also
uniquely identifies all traceoutput that is relatedto this single IP phone.
Phone line 1goes off hook, and the newcall is initiated.
3-32 Troubleshooting Cisco Unified Communications (TVOICE) v8.i
i 2010 Cisco Systems. Inc
Tracinq CSS Problem^
03/02/2009 1448:53 874 CCM|D Q:i /Vvtlvwp: wait_DaReq:
daReq.partitionSearchSpace(0aa4ac59-0b28-e72c-c4a0-ba7226e79975:),
filteredPartitionSearchSpaceStrinotBR managers pt:HQ managers pt.PSTN ptl,
partitionSearchSpaceString(BR_managersj)IHQ_mana9ers pt:PSTN_pt)|<CLID::StandAlon
eClusterxNIO.:10.1.5.2><CT::1.100.37.1^87><IP.:10.1.100'58xDEV::HQ-
CIPC2><LVL..Delailed><MASK::0800> \*"---^^
IList ofaccessible partitions thatthe phone ofthecaller can reach I
03/02/2009 14:48:53.875 CCM|5^rTi^s/.: match(pi="2'.fqcn="4085552002", cn="2002".
plv="5'. pss="BR_managers_pt:HQjTianagers_pt:PSTN_pt",
Tod Filtered Pss-"BR managers pt:HQ managers pt:PSTN pi".
dd-"",dac="0")|<CLID :StandAloneCluster><NID::10.1.5.2>';CT::1,100,37,1.887><IP:: 10.1.10
0 58><DEV..HQ-CIPC2><LVL: Detailed><MASK::0800>
Di;iit collection and ana=vsis continue
03/02/2009 14 48 53.875 CCM|Digitanalysis:
-;:.>-: -.!! y -".;-:-r:il.';-nn"<.EKi-t|<CLID::StandAloneCluster><NID::10.1.5.2><CT::1,1
00.37 1.887><IP 10 1.100 58><OEV..HQ-CIPC2><LVL::Detailed><MASK::0800>
The caller starts number dialing, which triggers thedigit-collection and digit-analysis processes
on Cisco Unified Communications Manager.
Thedigit-analysis logs display theCSSas partition search spaceand the list of partitions
that are members of this CSS:
BR_managcrs_pt
I lQ_managers_pt
PSTNj>t
When considering the trace output, look primarily tofilteredPartitionSeachSpaceString or
TodFilleredPss rather than parlitionSearchSpace orpss,because these strings contain the
partitions thathave been used for thefinal CoSprocessing.
Digit collection anddigit analysis continue until theuserdials enough digitsfor a potential
match to exist.
03/02/2009 14 48 56 436 CCMpigil Analysis, wait DaReq: M'.i.;i> j I :;;:<< v Hi.
.' ''.:-- |<CLID:StandA,oneCluster><NID"10.1 5.2><CT.:1,100,37,1 892><IP::10.1.100 5
8><DEV"HQ-CIPC2><LVL Arbitrary><MASK;:0800>
Caller and calling information reviewed, includes CSS and partitions
03/02/2009 14.48 56 436 CCM] . . . ..(pF,'2".fqcn="4085552002'\ cn=',2002",
plv="5", pss="BR_managers_pt HQ managers pl.PSTNpt".
TodFilteredPss-"BR managers pt'HQ managers pt.PSTN pi",
dd="2001".dac="0")|<CLID;SlandAloneCliJSlerxNID::10.1.5.2><CT..1,100,37,1.892><IP'10
1.100 58><DEV' HQ-CIPC2><LVL ,Delailed><MASK:0800>
03/02/2009 14 48 56 436 CCM|. I .;
... :.: I > :<CLID..StandAloneCluster><NID:10.1.52><CT.
1,100.37.1 892xlP. 10 1 100 58xDEV HQ-CIPC2xLVL. De1ailed><MASK::0800>
After the user completes dialing, the caller and calling information is reviewed in the trace log
file. This file shows the calling and called numbers and the CSS wiih all its partition members.
The completion of dialing also completes the digit-collectionand digit-analysis processes,
which can be seen in the trace file as "Nol'otentialMatchesExist."
TroubleshootingCisco Unified Communications(TVOICE) v8 0
Tracing CSS Problems (Cont.
03/02/2009 14:48 56 436 CCM|SlationD (0000003)
v i: '..-: |<CLID StandAloneCluster><NID::10.1.5.2><CT.:1,100,37,1.892><IP::10.1.100.58
xDEV :HQ-CIPC2><LVL State Tra nsi lion ><MASK; 0020>
Reorder tone (orAnnunciator error message, ifenabled) starts playing
03/02/2009 14:48:56.438 CCMjStationD: (0000003) StartTone tone=37(ReorderTone).
direction=0.|<CLID::StandAloneCluster><NID::10.1.5.2><CT::1,100,37,1 892><IP::10 1.1005
8><DEV:HQ-CIPC2><LVL::StateTransition><MASK::0020>
- engine
03/02/2009 144900 130CCM|Stationlnit: (0000003) ~i^\'m\E:s^! -oftKsy!:->.<;!;< ^tii-dC ,'!}
callReference=28396204.|<CLID::StandAloneClusterxNJD::10.1.5.2xCT::1,100,37,1.893x1
P::10.1.100.58><DEV::HQ-CIPC2><LVL::State Transit ion><MASK::0020>
03/02/2009 1449:00.131 CCM|StationD: (0000003) StojjTone |<CLID::StandAloneCluster>
<NID.:10.1.5.2><CT;;1,100,37,1 893><IP::10 1.100 58><DEV::HQ-CIPC2><LVL::State
Transition><MASK:.0020>
At this point, the call should be routed and the ringbaek tone should be played tothe caller. But
because permission problems exist, the processwill divert fromnormal.
Cisco Unified Communications Manager removes the dial tone and then immediately plays the
reorder tone to the caller to indicate the error state.
Alternately, if Annunciator isenabled, the respective error message will play instead. This
action will show upin thetrace file toreplace theplaying of thetone.
The caller hangs upline 1,and Cisco Unified Communications Manager removes the reorder
tone.
3-35
One-Way Calling
3-36
fhis topic explains the possible causes of one-way call scnip as a special case of single-site call
setup failure.
Problem report:
* Phone Acan initiate calls to phone B, and the audio works in both
directions; however, phone B cannot initiate calls to phone A.
The phone ACSS includes the partition of the phone Bline, but the
phone BCSS does not include the partition of the phone Aline.
OK
Inthefigure. IPphone Acancall IPphone B, but IPphone Bcannot call IPphone A.
This issue withpartitions andCSS typically occurs when the CSS of phone Aincludes the
partition of the phone Bline, but the CSS of phone Bdoes not include the partition of the
phone A line.
Use CiscoUnified Communications ManagerDialed NumberAnalyzer or theCisco Unified
Communications Manager administrative webinterface toverify andfixtheCSS andpartition
settings on the IP phones.
o Results Summary
Calling Party Information
o Calling Party = 3001
o Partition = Internal^
O Device CSS = Internal.ess
o Line CSS =
O AAR Group Name =
O AAR CSS =
Dialed Digits = 9011457876001
Natch Result = BlockThisPattern
Route Block Cause = Unallocated Number
Called Party Number =
Matched Pattern Information
O Pattern =
O Partition =
TroubleshootingCisco Unified Communications (TVOICE) v8.0
2010 Cisco Syslems, Inc
Call-Forwarding Issues
This topic explains the possible causes and recommended actions in the event of a call-
forwarding failure.
Call-Forwarding Issues
Problem report:
Phone A cannot forward calls to another phone.
The CSS on the IP phone does not include the partitionof the
destination directory number.
* The specified destination is invalid.
The specified destination is currently unregistered.
2001 3001
Cisco Unified
Communicalions Manager
Headquarters
This figure describes a scenario in whichone IPphonecannot forward a call to anotheron-
premises IP phone.
An on-premises call can fail to forward because of these reasons:
The CSS onthe IPphone does nothave the partition of thedestination directory number
included.
The specified destination is invalid.
The specified destination is not registered.
To resolve on-premiscs call-forwarding failure, use Cisco Unified Communications Manager
Dialed Number Analyzer or the Cisco Unified Communications Manager administrative web
interface toverify andfixanyCSS andpartition settings.
Use the Cisco Unified Communications Manager administrative web interface toverify that the
specified directory numberis validand registered.
Troubleshooting Call Selup Issues
Destination Unregistered
This section shows how to verify if the call-forwarding destination is currently unregistered.
.Ik,, <w Za " .Ail-.^ SStuJ~~$i <raJ
Device > Phones
Dialed Number Analyzer _
PSTN-ReBlnclHc! ess
includes partition of 30C1
Device > Phones > tine 3001
The first figure showsthe list of configured phones and highlights the rowshowing the
unregistered phone.
The second figure shows how tocheck if the destination isa valid number. The Cisco Unified
Communications Manager Dialed Number Analyzer can be used to simulate calling between
two phones, for example 2001 and300i as seeninthisCisco Unified Communications
Manager Dialed Number Analyzer example. Theoutput shows iheresult of simulated call as
valid (RouteThisPattem). but it also shows that the destination is nor registered(UnRegistered}.
There is an optionin CiscoUnified Communications Managerto forward a call to another
number if the destination is currently unregistered. 'Ihe bottomfigure shows Directory Number
Configuration pagewiththe Call Forward Unregistered (CFUR)option.
Inthis particular example, the CFUR number is not set, sothe call fails when the phone is not
registered. But even if the CFUR number isset, the call could possibly fail if itsassociated CSS
restricts the call forwarding.
Forwarding to Voice-Mail Issues
This topic explains the possible causes and recommended actions in the event that acall that is
supposed to be forwarded to voice mail is not forwarded.
Forwarding to Voice-Mai! Issue?
Problem Report:
. Phone Bdoes not answer. The call issupposed tobeforwarded Io voice
mail but. instead, a busy signal is heard.
Consider the Possibilities:
The specified destination CFNA isinvalid orunspecified.
- The voice-mail server or ports are currently not registered
Allvoice-mail ports are currently in use.
B Nol answering
Cisco Unified
Headquarters
This section describes a scenario inwhich calls tovoice mail are failing.
Problem Report
Phone Acalls phone Band phone Bdoes not answer. The call is supposed to be forwarded to
voice mail but. instead, a busy signal is heard.
Consider the Possibilities
These arc possible reasons why the calls may fail to forward to voice mail:
The CSS on the Cisco IP phone configuration is lacking the partition ofthe voice-mail pilot
number.
The specified destination Call Forward No Answer (CFNA) is invalid or unspecified.
The voice-mail server or ports arecurrently notregistered.
All voice-mail ports are currently in use.
Gather Facts and Formulate an Action Plan
Take these actions to troubleshoot the reason why the calls are not forwarding to voice mail;
' !hmlChr 'rrv10I7he -CiSC 'P PhnC- VCnfy lhat aCSS JS SCt fr lhe CFNA condition and
that the CFNA value is the valid pilot number of the Cisco Unity Connection voice-mail
rune Hon.
Using ihe administrative web interface of Cisco Unified Communications Manager verify
that the status ot the voice-mail ports is"registered."
Using the administratis eweb interface of Cisco Unity Connection, verify the current usa,e
of the \oice-mail ports.
Forwarding to Voice*
Device > Phones > Line
Q
-V4ICB HH Pllfrt IltfoTTlttiO
Mall pikrl Humhr ^9g0
Does VMPilotcontain voice-mail ports partition7
Advanced Features > Voice Mail > Voice Mail Ports
Voice-mailports not registered!
When acall has been sent toavoice-mail system, the voice-mail system can request that the
call be transferred to another directory number, toaPSTN destination (for example, the cell
phone of auser), or to an assistant. In all these scenarios, the voice-mail port is the entity that
requests the call that is being routed by Cisco Unified Communications Manager.
The top figure shows how to verify the forward setting on an IP phone line appearance. The
bottom figure shows how to verify the status ofthe defined voice-mail ports from Cisco
Unified Communications Manager. In this example, the ports are not registered correctly. This
incorrect registry could be areason why the calls forwarded to voice mail are failing.
2010 Cisco Syslems. Inc
Advanced Features >Voice Mail > Voice Mail Ports
CiscoUnified Communications Manager RTMT
Bothvoice-mail ports busy!
|0ll*rtiBH

*>I^.'i..^-'&,= '.,Le.i
il "1
HlKfMliUWlt
I >-* -ii- ML'.Itjii.i
1 fr^Crs -|MC,a,,Ifc,r.
1
O-'i. :"<;>,..=,
1
*ZJ3 -'.- ^*.M.ni^r*>*
j *>_* 'i-- ;^^
^ 1
I 1
* Z^ "H ; C-l-4i-3j HobJ"!.
* ' "=- E.'sns " <-' tr. ~ V
lina ii-a
*> ~> ,r - m-^J ' EeVrici
* -*:.).; -.^...(
^
=v . .*M" "
"* - , ,E ...J,,,.,,,
"* Ci-rtt+w-.n
P1-..1. =:',-,,
'fhis figure shows how to verify ifall voice-mail ports arc busy.
First, check ifthe \oiec-maii ports arc registered. The upper figure shows two registered voice-
mail ports.
In Cisco Unified Communications Manager RTMf, you can display the activity ofthe two
voice-mail pons. Display thechart for the performance counter Cisco Hunt Lists >
CallsInProgress. as shown on the bottom figure. The display shows that the two ports are active
at the moment, which means that no further voice-mail forwarding, other than the active two,
can take place.
Summary
Summary
*Most common on-premises issuesinclude CoS, incorrect digit
manipulation, unknown orunregistered target, and all ports busy.
CiscoUnified Communications Manageranalyzes digits one at a
time until only one explicit match orpotential match exists. If
multiple matches exist, theone with thefewest number of
potential matches is used.
Partitions aregroups of called numbers with identical reachability
characteristics. CSSs are listsof partitions that the ownerof the
CSS has access to. Timeschedules and time periodsare used to
activateor deactivatepartitions within a CSS, dependingon time
ordate infonnation.
Call setupfailure within a single siteisoften duetoCoSsettings,
an invalid directory number, oran unregistered directory number.
One-way calling is almost always an issueof CoSsettings.
Summary (Cont.)
Call-forwarding issues are usually due to CoSsettings, an
invalid directory number, or an unregistered directory number.
When calls fail to reach voice mail, many issues can be the
cause, including the CoS setting, unconfigured voice-mail
ports or pilot numbers, all ports being in use, orCFNA
settings having been entered incorrectly.
In this lesson, you learned to explain the common calling issues that can occur in asingle-site
Cisco Unified Communications Manager deployment and identify the most likely causes of
these issues.
3-43
References
Foradditional information, refer tothese resources;
Cisco Systems, Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0/1), February 2010.
http; www cisco com en US does \oicc ip commcuem/admiiyN 0 I ccincfii kxm-81)
cm.html
Lesson 3
Troubleshooting On-Net
Multisite Calling Issues
i*. Overview
This lesson discusses the issues that can arise in an environment with multiple sites and
multiple clusters. The issues that are commonly encountered can be with the intercluster trunks
'mm (ICTs). overlapping dial plans, and issues with calls between clusters. These issues collectively
areall a part of theoverall Cisco Unified Communications Manager dial plan.
^ Objectives
Upon completing this lesson, you will beable toexplain the common calling issues that can
occur in a multisite Cisco Unified Communications Manager deployment and identify the most
likely causes of these issues. This ability includes being able tomeet these objectives:
Describe the issues that are related to call setup that are common to multisite deployments
Explain the issues that arerelated tocall setup and anoverlapping dial plan and describe
how to solve these issues
Explain the issues of intercluster communications and CCD-type calls and describe how to
solve these issues
Review gatekeeper functions, describe thetypes of RAS messages, andtroubleshoot
discovery, registration, and call admission issues
ReviewCisco Unified Border Element general functions and the Cisco Unified Border
[lenient function inthe CCDprocessandtroubleshoot CiscoUnifiedBorder Element in
the CCD process
Explain how to recognize codec-related issues andhowtoavoidcodec mismatches thatcan
cause remote calls to drop immediately
Multisite Dial Plan Issues
This topic describes the issues that are related to call setup that are common to multiple site
deployments.
3-46
HQ-1
Cisco Unified
CoiTTunicadoni
Manager Ousts
BRANCH-1
IP WAN
ntersite possible causes
* Overlapping dial plan
- Toll bypass sellings
- Local end CoS
Remote end CoS
ntercluster Irunk sellings
QoS issues with signaling
across WAN
CAC mechanism prevents
the call
Issues with CCD
|J
IBR-2
HQ-2
Cisco Unified
Communicalions
Manager Cluster
BRANCH-2
Because the distributed call-processing model is two or more campus clusters that arc
interconnected with WAN andpublic switch telephone network (PSTN) links, all of the
problcms that canoccur ina campus deployment apply tothismodel as well. Inaddition, some
unique problems canoccur inthis call-processing model.
The different types of settings that youshould consider when call setup fails include the
settings on the endpoints. thesettings on the Cisco Unified Communications Manager route
plan, and the settings on the WAN link.
These settings ontheendpoints could cause thecall setup to fail:
1 he endpoints are not registered.
Coder-decoder (codec) mismatch between the two endpoints: InCisco Unified
Communications Manager, thismismatch is managed by theregion setting of thedevice
pool of the endpoint.
CoSsettings onthe endpoints: In Cisco Unified Communications Manager, you manage
this by creating and applying partitions and callingsearch spaces(CSSs),
The Cisco Unified Communications Manager routeplancancause thecall to fail because of
these reasons:
Aconfigured transformation pattern causes incorrect digitmanipulation.
AMedia Termination Point(MTP) is required for thiscall, but no MTPis currently
available.
The destination might not be currently registered to the Cisco Unified Communications
Manager.
Call Control Discovery (CCD) issues might occur when patterns are not correctly
advertised or imported.
These settings on the WAN link could cause the call-setup issue:
. The carrier might be having problcms or is not complying with the service level agreement
(SLA).
Qualitv of service (QoS) issues on the WAN link might be causing the call-setup messages
to be dropped or severely delayed. Acompressed Real-Time Transport Protocol (cRTP)
mismatch on the two ends of apoint-to-point WAN link could cause the calls to fail. This
problem would occur when only one side of the point-to-point connection has cRTP
configured.
The locations or an RSP-based Call Admission Control (CAC) mechanism has prevented
the call and automated alternate routing (AAR) is not configured or is configured
improperly.
>2010 Cisco Sys.ems, Inc Troubleshooting Cat! Setup Issues 3^7
Overlapping Dial Plan
This topic explains the issues that are related to call setup and an overlapping dial plan and
describes how to solve these issues.
3-48
Wail for
Catling 911 interdigit timeout!
-
What makesthis overlapping?
Calling 911
Route Call
*v t^J
Route Pattern
911
||9 1[2-9]XX[2-9]XXX
One of the most common configuration problems in an intercluster Cisco Unified
Commumcations Manager configuration is an overlapping dial plan. Dial plans overlap when
two or more patterns match aset of dialed digits. This situation most commonly occurs when
an mternal extension matches the starting digits ofalonger route pattern. This situation will
cause the t isco Unified Communications Manager to wad for additional digits until an explicit
match ,s made or until the interdigit timeout (15 seconds by default) occurs. Many users who
have to wait for the mterdigit timeout will give up before the timeout occurs and perceive the
call as failing. Ii the user had waited for the interdigit timeout to expire, the call would have
worked. To resolve this issue, use either site or access codes to make the patterns unique
Note
When two phones have the same exact extension in the same partition, this scenario is
known as a shared lineappearance.
In the first example in tins figure, the Cisco Unified Communications Manager digit analysis
process finds nvo possible matches when auser dials 911. The first patten, is an exact match
but the second pattern could match if the caller dials additional digits before the interdigit
timeout occurs. If the caller waits for the length of the interdigit timeout, digit analysis will
choose the pattern of91 I and route the call as expected.
Potential matches are patterns that could match ifadditional digits are received. The call is
extended it one or more matches exist and one ofthese conditions ismet:
No more potential matches exist.
Potential matches exist: however, the matching pattern is marked "urgent."
Potential matches exist and the interdigit timeout has expired.
Note If multiple matches exist, digit analysis will select the pattern withthe fewest number of
potential matches.
Whena potential matchexists, and the matching patternsare not marked as urgent, the user
will have towaitfor theinterdigit timeout to expire before thecall is routed. During thistime,
theuser might thinkthatthecall has failed andhangup before theinterdigit timer expires. To
avoid thisissue, if possible, do not allow thedial planstooverlap. If youconfigure thedial plan
with the rules of the North American Numbering Plan (NANP) oranyother numbering plan
that is being implementedin mind, you can avoid most of these issues.
In the secondexample inthis figure, the NANPstatesthat thedigit dialedafter a "1" cannotbe
a "1" or a "0."Inaddition, the first digit of theoffice code (the first digit ofa seven-digit
number) cannot be a "I" or a "0." Thecorrect routepatternof 9.1[2-9]xx[2-9]xxxxxx
eliminates the conflict with 911.
2010 Cisco Systems. Inc. Troubleshooting Call Setup Issues
3-49
Intercluster Call Setup
3-50
This topic explains the issues with 11.323 and Session Initiation Protocol (SIP) trunks as well as
CCDtypes of calls that can occur when call setup to another cluster is failing and describes
how to solve these issues.
ClusterA
Route pallern does
not have correcl
partition
Phones use CS&*"!
lhal denies use of
the route partem
Second choice requires
digit manipulation
Cluster B
Inbound trunk
uses CSS that denies
reaching phones.
, " UnifiedCM= Cisco UnifiedCommunications Manager
lClusterC
fhis section discusses the issues that can arise in an environment with multiple clusicrs.
Commonly encountered issues can involve theconfiguration of ICTs or the use of thePSTN as
an intcrcluster-conncctivity backup.
if calls between the clusters and over the ICT arc not setting up correctly, many of the possible-
causes relate to permissions or digit manipulation.
Problems withthe CSSsettingon theoriginating phonecouldexist. This CSSmust include ihe
route partition that is associated with theroute pattern that is to bedialed.
Problems with digit manipulation could existwhen progressing thecall over ICT. Such
problems can involve digit-discard, prefix, or transformation mask settings, or the dialed
number might be an invalid number in the originatingcluster.
IPconnectivity issuescouldexist between theCisco Unified Communications Manager servers
of thetwo clusters. These problems canrelate to various IPandvoice QoS issues, or theCAC
mechanism that is usedbetween the twoclusters might prevent ihe call (as can occur with
11,323 gatekeeper-based bandwidth management). Or, the carrier of the IP WAN could be
having transport problems.
The remote cluster inbound trunk has an incorrect or incorrectly configured CSS that denies the
reaching of the remote IP phone.
Theremote endpouu has the incorrect partition or is not registered totheremote Cisco Unified
Generally, to help troubleshoot call-setup problcms ina mullicluster environment, use the
combination of all the troubleshooting recommendations that were provided thus far.
<* Intercluster On-Net Call Anatomy
This sectionexploresthe process of an intercluster on-net calling.
Intercluster On-Net Cai
Calling direction
Local Cisco Unified
Trunk or Route List
Called-Party Trans forma lions
/
RP 512XXXX
Partition "' ,;. . '
DDI Called-Party Transformation
Pref.l
5122001 dialed
7
Iniercluster CSS *=.-
Remote Cisco Unified
Communicalions
, Manager
X
Trunk Inbound Calls
Significant Digits. 4*
Trunk CSS: Phones
Trunk Incoming
Called-Party
Number
Selling:
f rj-iix. iitr.rj
D=dii:, CF-S
Partition1 P Hones
Site Code: 512
" Translalion pattern used alternatively
The figure analyzes the configuration elements and their mutual relationship when setting up
the call from one cluster to another cluster via a trunk. The figure also demonstrates the points
of possible misconfigurations that need to be revisited during the call-setup troubleshooting.
The trunk can be any supported trunk, because the outlined process is independent of the type
of the trunk. Having a call setup via a gatekeeper-controlled trunk or via a Cisco Unified
Border Element involves these devices also, but they will be discussed separately later in this
lesson.
The two clusters that are shown use overlapping directory numbers 2001 and, therefore, site
codes were added. Because the call setup will proceed from the left cluster to the right cluster,
only the target cluster site code 512 is shown.
When the calling phone places a call to 5122001, the call-routing engine on the local Cisco
Unified Communications Manager checks the call-routing destinations that the phone can
access based on its CSS. In this case, Intercluster CSS lists the RemoteSite partition, and the
phone can access the route pattern 512.XXXX. The route pattern can perform various digit
manipulations by using discard digits instruction (DDI), a called-party transformation mask, or
by adding a prefix and points to either a route list or an individual trunk.
At the trunk or route list, additional called-party transformations are possible. The tmnk
performs the called-party transformations via the called-party transformation CSS. Route list
called-party transformations use DDI, a called-party transformation mask, or a prefix to modify
the number.
The call setup proceeds along the trunk and reaches the remote cluster. For the inbound call,
further digit manipulations are possible. Digit manipulation can be performed at the trunk itself
as significant digits or by employing translation patterns as an alternative. Digit manipulation
can also use Incoming Called Party Number settings that allow adding a prefix, stripping digits,
or configuring more complex called-number manipulation by using CSS. This method will be
discussed in greater detail in the next lesson.
>2010Cisco Systems, Inc. Troubleshooting Call Setup Issues 3-51
The inbound trunk can have the CSS set (Inbound Calls section). To reach the target phone, the
trunk CSS must have anaccess tothepartition thatis used at thephone; otherwise, ihecall is
rejected and the caller hears the reorder tone.
Onthe next pages, a Cisco Unified Communications Manager trace ofa successful call setup
via a SIP trunk is presented.
Troubleshooting Cisco Unified Communicalions (TVOICE) v8 0 2010 Cisco Systems Inc
*" Tracing Call Setup via a SIP Trunk
Based on the intercluster on-net call analyses, this section describes the Cisco Unified
Communications Manager intercluster on-net call trace output.
Tracing Call Setup via SSP Trunl
Alltracing output shown from local Cisco Unified Communications
Manager perspective
Caller took phone off-hook and dialed a number
Stationlnit. (0000019) SoftKeyEvent .-HhK.^fvei'i-^mw.Cdin linelnstance=0
callReference-0 |1,100.49.1.3312*10.1.2 17*SEP0024C4454AD8
StationD. (0000019) DialedNumber dialed Nurnber=5122001 linelnstance=1
callReference=20476773.|1,100,49,1.3314*10.1.2.17ASEP0024C4454AD8
StalionD- (0000019) (1.100.9,24) Calllnfo call.ngPartyName="cjll!r-.;Pjity-->oo'
eg pnVoiceMa ilbox= alle mateCa llingParty=
calledPartyName="calledParty=5122001 cdpnVotceMailbox=
ofiginalCalledPartyName="originalCalledParty=5122001
originaICdpnVoiceMa ilbox= origina ICdpn Red irectReason=0
laslRedirectingPartyName=''iaslRedirectingParty=5122001
Ia st Re dire cti ngVoiceMai Ibox= Ia st Redirecti ng Re ason=0
<,i! T!-:'Ou!L{)(j!vs'!linelnstance=1 call Reference-2047 6 773. version:
8572001311,100.49,1.3314*10.1.2.17*SEP0024C4454AD8
All the tracing that is shown on this and the following pages is shown from the calling side, the
local Cisco Unified Communications Manager perspective.
At first, the caller takes the phone off-hook and dials the number 5122001. From the local
Cisco Unified Communications Manager, it is the outbound call, and you can see the summary
of calling and called numbers.
) 2010 Cisco Systems, Inc. Troubleshooting Call Setup Issues 3-53
3-54
Outgoing trunk >' .-... kselected
,VSIP/SIPCdpc(1.66.7)/ci=2047i&?r4*?ccbld=a/scbld=0/LocalizeOutpulsedNumber
SIPCdpc on device -" , CSS = ,useDevicePoolCgpnCss =1
AltemateCgpn(globai)=2001 cgpn-2001, cdpn=5122001,
mUpdatelnstruction=0|1,100.49,1.3314*10 1 2 17*SEP0024C4454AD8
Outgoing SIP INVITE senl
/.'SIP/SIPTcp/wait_SdlSPISignal Outgoing SIP TCP message io 10 2 1.1 on port 5060
INVITE srp5122001@10 2 1 1:5060 SIP/2 0
Via SIP/2 0/TCP 10 1.1 1 5060,branch=z9hG4bK172dee923
From =... ' ->,tag=ae40d64d-5de5-4f28-b7f9-2614e6ea9d44-20476774
To <sip5122001@10 2 1 1>
truncated ..
CSeq 101 INVITE
Contact <sip2001@10 1 1 1 5060,transport=lcp>
P-Asserted-ldentity <sip 2001@10 1 1 1>
Remote-Party-ID <sip2001@10 1 1 1>.party=calling.screen-yes;pnvacy=off
(runea (erf
The local Cisco Unified Communications Manager selects the outbound trunk, which is the SIP
trunk named SIP-trunk.
The local Cisco Unified Communications Manager formulates the SIP INVITE and sends it on
to the trunk.
"racing Caii Setup via SIP Trunl
Incoming SIP ringing received
//SIP/SIPTcpAvait_SdlReadRsp'
SIP/2.0 .< r<,.-:,-.
Via. SIP/2.0/TCP 10.1.1 1:5060,branch,z9hG4bK172dee923
From: <siP2001@i0 1.1. l>;tag=ae40d64d-5deS4f28-b7f9-2614e6Ba9d44-20476774
tnâtlf0,@102 11>;la9^72fb06df-6cd3-4b89-92e1ê847500b57d-20583272
Contact <sip.5122001@io 21.1 5060;! ransport=tcp>
Supported: X-cisco-srtp-fatback
Supported. Geolocation
P-Asserted-ldenlrty: <sp:2001@lQ 2 1 1>
Remote-Party-ID: <sp 2001@10.2.1 1>;party=called;screen=yes priw,cy=off
Content-Length: 0
Ringbaektone at caller phone
StabonD. (0000019) StartTone tone=36(AlertingTone)
direction^ |1,100,56,1 19*10.2.1 1""
'TCP messagefrom 10.2.1.1 onport 5060.
The remote Cisco Unified Communications Manager returns the SIP 180 Ringing to confirm
that the dest.nat.on ,s reachable. The local Cisco Unified Communications Manager sets the
ringbaek tone at the calling phone.
2010CiscoSystems. Inc.
;IMi
//SIP/SIPTcp/wail_SdlReadRsp \
SIP/2 0 - : = -- > - '<-
Via SIP/2 0/TCP 10 111.5060.branch=z9hG4bK172dee923 ._
From <sip2001@10 111>;ta9=ae40d64d-5de5-lf28-b7f9-2614e6ea9d44-20476774
To <s.p5122001@10.2 11>.lag=72fb06df-6cd3^b89-92e1-de847500b57d-20583272
truncated
v=0
o=CiscoSystemsCCM-SIP2000 1 IN IP4 102 11
s=SIP Call _,_,
C=IN IP4 10 2 2 '7 - Called-phone IPaddress
t=0 0
m=aud,o4000RTP/AVP9 0 8 116 18 101
a=rtpmap 9G722/8000 Called side offers codecs
a=ptime 20
a=rtpmapOPCMU>8000
a=plime 20
a=rtpmap 8 PCMA'8000
a=ptime 20
a=rtpmap H6 1LBC/8OOO
a=ptime 20
a=maxptime 60
. truncated
J I-TCPmessagefrom 102.1.1 on port 5060'
The remote Cisco Unified Communications Manager sends the SIP IS3 Session Progress to
notify the local Cisco Unified Communications Manager about these particulars:
The called-phone IP address, to set up the media to, is 10.2.2.17.
The list of codecs that the called phone supports.
3-56 Troubleshooting
CiscoUnified Communications (TVOICE) v80
Tracing Call Setup via SIP Trunk (Cont.)
SIP/SIPTcp/wa.LSdIReadRsp i-:^.,-ay;f.TCP message from 10 2.1.1 on por!5060-
Via. SIP/2 07TCP 10.1 1.1:5O60;branch=z9hG4bK172dee923
From:<sip:2001@10.1.1.1>:tag-aed0d64d-5de5-4f28-b7f9-2614e6ea9d44-20476774
To <sip:5122001@10.2.1 1>,tag=72fb06df-6cd3-4b89-92e1-de847500b57d-20583272
.. truncated ...
v=a
oKîscoSystemsCCW-SIP2000 1 INIP4 10 2 1 1
s=SIPCall
c=IN IP4 10.2 217 SameIPaddress andoffered codecsfrom thecalled side
m=audo4000RTP/AVP9 0 8 116 18 101
a=rtpmap 9 G722/8000
a=ptme.20
a=rtpmap.O PCMU/8000
a=ptjrrte.20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:116iLBC/8000
a=ptme20
a=maxpf,me:60
... truncated .
The remote Cisco Unitied Communications Manager sends the SIP 200 OK with the same
Session Description Protocol (SDP) information as in the previous SIP 183 Session Progress.
3-58
StationD (0000019, OpenReceiveChannel conferenceID=20476773
passThruParty!D=16777237 millisecondPacketSize=20
compressionType=6(Media_Payload_G722_64k)RFC2833PayloadType=101
qualifier^ > sourcelpAddr^lpAddr type 0
ipAddrft<0aO2O211OOOMOO0OO0O0OOOOO00O0O0(10.2 2.17).mylP:lpAddr.type.0
ip,4Addr.0xQa010211(10 12 17) |1.100,61,1 !n*n'
//SIP/SIPTcp<wail_SdiSPISignal -'; -. :' TCPmessage to 10 211on port 5060
.. sip5122001@10.2 115060, Iransport =tcp SIP/2 0
Via SIP/20/TCP10 1 1 1 5O60,branch=z9hG4bK234c5O57
From <sip2001@10 111>;tag-a&40d64d-5de5-4f28-b7f9-2614e6ea9d44-20476774
To <sip5122001@10.2 11>,tag=72fb06df-6cd3-4b89-92e1-de847500b57d-20583272
v = 0
o=OscoSystemsCCM-SIP 2000 1 IN IP4 1011.1
s-SIPCall
t=0 0
m=audo .
a=rtpmap.9
a=ptime.20
a=recvor\ly
truncated
RTP/AVP 9 101
\._/8000
The local Cisco Unified Communications Manager opens the media for receiving at the calling
phone {10 1"> 17) 0.722 is preselected for open receive channel, but no remote Real-Time
Transport Protocol (RTP) ports are known yet. They come in the next SIP signaling messages.
The local Cisco Unified Communications Manager formulates the SIP ACK and sends it to the
remote Cisco Unified Communications Manager with this information:
IP address ofthe calling phone, ioset the media to. is 10.1.2.17.
The calling phone RTP pon is 26848 and the list of codecs that the phone supports (only
G.722at the beginning of the list is shown).
Tracing Call Setup via SIP Trunk (Cont.
//SIP/SIPTcp/wa^SdlReadRsp' u c,,, sih TCP message from 10 211on port 35639
* r sip.2001@10.1 1-1.5060;transport=tcpSIP/2 0 P
Via SIP/2 0/TCP 10 21.1 5060,branch=z9hG4bK511665c95
From <sip:5122O01@l0.2.1.1>;tag=72fbO6df-6cd3-4b89-92e1-de847500b57d-2058327?
To. <8.p:2001@10 1-1.1>.ta9=ae40d64d-5de5^f28-b7f9-2614e6ea9d4420476774
v=0
o=OscoSystemsCCM-SIP2000 2 IN IP4 102 11
s=SIPCall
c=lN IP4 10.2 2 17 Called-phone IPaddress
m=audio 23402 RTP/AVP 9101 Called-phone RTP port
a=SP20 G722'8000 " Caned-s,6e selected codec
a=rtpmap:101 telepfione-event/8000
a=fmlp-l01 0-15
.^SIP rce.NVITnlftd CmmTati0nS Mm- returns the SIP INVITE (more precisely,
it is SIP rc-INVITE) that contains the callcd-phone media information:
The IPaddress of the called phone is 10.2.2.17.
The RTP port of the called phone is23402.
The selected codec is G.722.
Note
2010CiscoSystems, Inc.
in his call setup, the target phone selects the codec. The selection is based on the list of the
caHing phone and its own lists of the feasible codecs (SIP delayed offer). Another option is a
SIP early offer, mwhich the codec negotiation is different from that shown in the figure
3-60
//SIP-SIPTcp'wart.SdISPISignal.. . : .- TCP message.o 10 2.1 1on port 35639
index 4
Sip/2 0 .'".
via SIP/2 0/TCP 102 1 V5060.branch=z9hG4bK511665c95 --,.,
Rom <,p 5 22001@10 211>.tag=721b06df-6cd3-4b89-92e1-de847500b57d-20583272
truncated .
Starting media transmission from the calling :.>.'
tothe called 10.2.2.17/23402 using G.722
SlatonD (0000019) slartMediaTransmission conferenceID=20476773
DassThruParlylD=16777237 re motel pAddress=lpAddr.lype 0
!pSdr0rfa02021100CCO0CCCC0O0^
m(IISecondPackelS.ze=20 compresSType=6(Media_Pay.oad_G722. 64k)
RFC2833PayioadType-101 qualif-erOut^?. mylP: lpAddr.type:0
,pV4Addr0x0a01021H " " )|1.100.56,1.22*10 2.1 1"
The local Cisco Unified Communications Manager sends the SIP 100 Trying to confirm the
receipt of the SIP rc-lNVnti.
The local Cisco Unified Communications Manager has all the necessary media ^"tanon ,o
stan the media transmission from the calling phone 10.1.2.17lo the target phone 10.2.2.17. Ihe
remote RTP port is 23402, and the selected codec is G.722.
Tracing Call Setup via SIP Trunk (Conl
//SIP/SIPTcp/wait_SdlSPISignal. s' =!;: <-." $\>.
SIP/2 0"' /
Via: SIP/2.0/TCP 10.2.1.1:5060;branch=z9hG4bK511665c95
From <sip.5122001@10.2 1.1^;lag=72fb06df-6cd3-lb89-92e1-de847500b57d-20583272
To <sip:2001@l0.1.1 1>;tag=ae40d64d-5de5-4f28-b7f9-2614e6ea9d44-20476774
.. truncated ...
v=Q
o-CiscoSystemsCCM-SIP2000 2 IN IP4 10.1.1.1
s^SIP Call
c=INIP4 10 1 2.17 *.
1=0 0
m=audio 26848 RTP/AVP 9101 Calling-phone RTP port
TCP message to 10.2.1.1 on port 35639
a=rtpmap 9 G722/8000
a=ptime.20
Calling-phone IP address
Calling-side codec
a=rtpmap 101 telephone-event/8000
a=fmtp:1010-15
The local Cisco Unified Communications Manager confirms the media establishment by
originating the SIP 200 OK. SIP 200 OK isalso used to confirm the calling-phonc media
parameters to the remote Cisco Unified Communications Manager (note that the calling-phone
RTP pons are unchanged):
IP address ofthe callingphone,to set the mediato, is 10.1.2.17.
The calling-phone RTP port is 26848, and the confirmedcodec is G.722.
J2010Cisco Systems, Inc.
//S1P/SIPTcp/waS_SdlReadRsp : .. ' . i'PTCP messagefrom 10.2.1.1 on port 35639
sp2001@10 1 1 1.5060,transpOi-l=tcp SIP/2.0
Via. SIP/2 0/TCP 10.2.1 1.5060.branch=z9hG4bK67241de04
From <sip.5122001@102 1i>,tag=72fbu6df-6cd3-4b89-92e1-de847500b57d-20583272
To <sip2O0i@10 1.1 I>,tao,=ae40d64d-5de5-4f28-b7f9-2614e6ea9d44-20476774
Date: Wed, 03 Mar 2010 17-17.51 GMT
Call-ID ac4f5bOO-b8e199b7-2-101010a@10 1 1 1
Max-Forwards 70
CSeq 101 ACK
Allow-Events presence, kprnl
Content-Length 0
Calling-side setup now compleled
TheremoteCiscoUnified Communications Manager returns SIP ACKto acknowledge the
used calling-side media parameters. This completes the call setup at the local end ot the SIP
trunk.
Tracing Call Setup via SIP Tn
* Tracing output now shows remote Cisco Unified Communications
Manager actions before call setup completion.
Inbound call on SIP trunk SIP irk, significant digits selected
//SIP/SIPD(i,65,6)/ccbld^lWscbid=0/reslartO_SIPSetijplnd:mTsp.deviceName[SIP-
'rl ]|1,100.56.1 24A10 1.1.1"
20:14-55 717 |//SIP/SIPD{1,65,6)/ccbld=10/scbld-07restart0_SIPSetuplnd:
mTsp.deuceName( 5^->*].IransformDn2001J1.100,56,1.24AtO.1.1.1**
20:14:55.717 |//SIP/SIPCdpc(0,0,0)/ci=[Vccbld=O/scbld=0/globalize. Performing
stripAndPrependDigits Prefix data = , Strip Data = o|***n*
Starting media transmission from the called 10 2 2.17
lo the calling 10.1.2.17/26848 using G.722
StationD (0000001) startMediaTraremission conferencelD=205B3275
passThnjPartylD=16777221remotelpAddress=lpAddr.type:0
ipAddr OxOa01 tJ2110ixro0000000000000000000(10.1.2.17)
milliSecondPacketSize=20compressType=6(Media_Payload_G722_64k)
RFC2833PayloadType=101qualif>erOul=?. mylP: lpAddr.type:0
ipv4Addr0xOa020211('O 2? '?) |1,100,56.1 27*10.1.1.1"
This figure shows that the two selected events before the call setup iscompleted at the remote,
called-side Cisco Unified Communications Manager.
Initially, when theSIPINVITE wassent by thecalling Cisco Unified Communications
Manager, the called Cisco Unified Communications Manager received this SIP message onits
SIP trunkthat is namedSIP-trk. Although the number5122001 was dialed, the SIP trunkhas
thecalled-number digit manipulation using four significant digits configured. Thus, the
transformed number is 2001.
When the called Cisco Unified Communications Manager learns about themedia parameters,
itssets upthe media and starts thetransmission from the called phone with the IPaddress
10.2.2.17 to the calling phone with the IPaddress 10.1.2.17. The calling-phone RTP port is
26848, and the codec is G.722.
ICT and H.323 Trunk Issues
This section describes the typical issues of Cisco Unified Communications Manager ICT and
11,323 trunks.
3-64
4-. .n.ir^. *fft..XZtJtJLx&rr<ma ~ &&**&
and H.323
ICT
Gatekeeper issues:
Registration
Incorrect call routing
Bandwidth managementCAC
RP Roule pattern TP Translation pattern. RL Route list, CSS Calling search space
Call setup that uses nongatekecpcr-controlled ICT is like SIP trunk call setup; however, another
II,323-like signaling protocol is used. All issuesthat arisefromincorrect CSS, partitions,
incorrect digit manipulation, or codec mismatch wouldapplyto any type of trunk. On 11.323
trunks, in addition, common issues might include fast-start or slow-start problems on a trunk
that is inbound or outbound.
Agatekeeper-controlled trunk involves the gatekeeper inall call-setup, midcall, and call-
disconnect signaling (known alsoas Registration, Admission, andStatus [RAS]) procedures.
This scenario raises a completely newsei of potential issueswhensettingup thecall. Theseare
the most typical gatekeeper issues:
Registration: Cisco Unified Communications Manager cannot register tothegatekeeper,
andtheentire gatekeeper-controlled trunk is practically notoperational.
Misconfigured call routing: This issue could cause call-setup blocking or cause thecall io
progress to an incorrect endpoint.
Bandwidth management: The CACmechanism at the gatekeeper might blockthe call
cither because of misconfiguration or because of exhausted WAN resources.
Call-setup troubleshooting, when gatekeeper-controlled trunk isused, iscovered indetail later
in this lesson.
TroubleshootingCisco Unified Communications(TVOICE) v8.0
Cisco Unified Border Element Call-Setup Issues
This section describes thecommon, general call-setup issues whenCiscoUnified Border
Element is used.
Cisco Unified Border
Setup issues
Digit manipulation
m trunk, gateway, or IP
Cisco Unified Border Element issues:
Misconfiguration
MTP not allocated for H.323-SIP
CSS and partition issues
Incorrect digit manipulation
CAC failure
RP Route pattern, TP Translation pallern. RL: Route list, CSS- Calling search space
WhenCiscoUnifiedBorder Element is usedto interconnect two CiscoUnified
Communications Manager clusters, all call setup, midcall, and call disconnect signaling
traverses it. Configure theCiscoUnified Border Element at CiscoUnifiedCommunications
Manager by using a trunk orgateway. The troubleshooting from the Cisco Unitied
Communications Manager perspective isthe same as for atrunk oragateway. However,
because the Cisco Unified Border Element manages all intercluster signaling and, optionally,
media, troubleshooting procedures must include theCisco Unified Border Element aswell.
The most typical issues, when Cisco Unified Border Element isused, would include these:
There ismisconfiguration at the Cisco Unified Border Element orconfiguration does not
match what issetat Cisco Unified Communications Manager.
MTP is not allocated for H.323-S1P when fast start and early offer is required.
CSS and partition issues exist.
Incorrect digit manipulation occurs in Cisco Unified Communications Manager
configuration at the trunk, gateway, route pattern or route list, orat the Cisco Unified
Border Element itself.
CAC failure. The Resource Reservation Protocol (RSVP) CAC mechanism blocks the cal
because of either misconfiguration or exhausted WANresources.
Call-setup troubleshooting, when Cisco Unified Border Element is used, iscovered in detail
later in this lesson.
CCD Operation Review
This section reviews ihe concept of CCD, the newfeature of CiscoUnitiedCommunications
Manager Version 8.x.
CCD Operation Rtsi
San Jose Unified CM' Routing Table
New York Unified CM' Routing Table
10 2.1 1 SIP
.442077112/4 10.2.1 1 H.323
10.1 1.1
10.1.1.1 SIP
.1949222/4 10111 H 323
10211
Advertising 8212XXXX
8442XXXX
8212XXXX
^S| 10 211
SAF Network
U^
Advertising 6408XXXX,
8949XXXX 8949XXXX
Irvine
SAF-enabled H.323 trunk
SAF-enabled SIP Irunk
NewVortt
8442XXXX
London
' Unified CM - Cisco Unified Communicalions Manager
TheCCD feature leverages the Service Advertisement Framework (SAF) network service, a
proprietary Cisco service, to facilitate dynamic provisioning ofintercall agent information. By-
adopting the SAF network service, the CCD feature allows Cisco Unified Communications
Manager toadvertise itself along with other key attributes, such as directory number patterns
that arc configured in Cisco Unified Communications Manager Administration. This allows
other call control entities that alsouseSAF network tousetheadvertised information to
dynamically configure and adapt their muting behaviors. Likewise, ail entities that use SAF
advertise the director)' number patterns that they own along with other key infonnation, so
other remote call-control entities can learn the information andadapt therouting behavior of
the call.
With theCCD feature, each local Cisco Unified Communications Manager cluster canperform
these tasks:
Establish an authenticated connection with the SAF forwarder.
Advertise totheSAF network thedirectory number pattern, signaling protocol, IPversion 4
|IP\4) address, or hostname of the server that arcconfigured inCisco Unified
Communications Manager Administration for the local cluster.
Register with the SAF network to learn routes coming from other remote call-control
entities that also use the SAF-relatcd network.
Use theinformation that is learned from theadvertisements todynamically addpatterns to
the Cisco Unified Communications Manager cluster master routing table, which allows
Cisco Unified Communications Manager to route and setup calls tothese destinations by
using the associated IP address and signaling protocol information.
When connectivity to aremote call-control entity islost, the SAF network notifies Cisco
Unified Communications Manager to mark the learned information as IP unreachable. 'I he
call then noes through the PSTN.
3-66 Troubleshooting Cisco Unified Communications (TVOICE) v8.0
toTnHmatsAFr *^Tf"* ^ *" ^^^S if ' S'<* ^tivity
to its pnmary SAF forwarder for any reason, another backup SAF router can be selected to
advertise and hstrn for infnrm.f.n^ c seicciea to
advertise andlisten for information.
\
Lolnh, > T^ UnifiCd C0m"icata Manager clusters, each having two
5ttf^ . CS" C1T? maimam hSted direCt0ry nUmber Patterns-directory number
patterns that belong to the local call-control entity. The San Jose Cisco Unified
^Sten"a8n rt^ ^ dirCCtory nUmber pattems ^SXXXX and
SS; ClSC UmfiCd Commuons Manager in New York has 82I2XXXX and
It^S^sSt; f hSted TPattCmS Md PS failVer Prcfkes for ** Pêms,
network ^^ 0miatl0n l *' remote CaI1-COntrol enti** that use the SAF
Each Cisco Unified Communications Manager also listens for advertisements from remote call
The Cisco IOS router notifies the local Cisco Unified Communications Manager when remote
call-control entities advertise their hosted directory number patterns. The SAF WdeT
a, 2 crCqUeStS fr0m thC 'Cal CiSC Unif'ed C-municationS Manages so that
the dutlef CmmUn,Cat,0nS M^ " ^-rtise the hosted directory number patterns for
SAF-enabled trunks that are assigned to the CCD advertising service, H323 or SIP manage
l"t^rSotr"' em,tieS ^ ^ ^ SAF ^^2lT^
patterns "*reqUCStmg SmicC manage outSoinS ^to learned
CCD U|>eratioi
San Jose Unified CM" Routing Table
NewYork Unified CM* Routing Table
8212XXXX '1212444 M 10.2.1.1 SIP
B442XXXX +442077112(4 102.1.1 H323
10.1.1.1
B408XXXX -.1408555 M 101.1 1 SIP
8949XXXX +1949222M 10.1 1-1 H.323
1021 1
This figure shows the call to the hosted directory number pattern that leverages he established
SAF nmvork When the pattern is learned and the SAF network is reachable, callers can place-
calls between the clusters by using the prescribed SAF-enabled trunks.
When connectivity to aremote call-control entity is lost, the SAF network titles Cisco
Unified Communications Manager to mark the learned in.ormation as IP unreachable. 1he call
then goes through the PSTN tnot shown in the figure). The same call would go to
+442077112001 in this case.
Typical CCD Issues
44^1^^.
This section describes the typical issues that areexperienced with CCD processes.
ileal CCD Issues
San Jose Unified CM" Routing Table
New York Unified CM* Routing Table
EEÊ e&ssi
B442XXXX 442077112/4 105.1.1 H.323
10.1.1.1
S9SjSf-*-~-
Si Jose
8949XXXX
trvine
Misconfigured SAF trunks
Patterns not advertised
Digit or pattern manipulation issues
Maximum number of learned
patternsexceeded
' Unified CM = Cisco Unified Communications Manager
=1ft.1.1j1 SiP
10.2.1.1
CCD is acomplex mechanism, and itcan have several points offailure. However, there are also
many potential points ofmisconfiguration. Both ofthem lead into unavailable patterns and
unsuccessful calls.
Aconfiguration page in Cisco Unified Communications Manager allows the administrator to
block certain routes thatarebased onLearned Pattern Prefix, Learned Pattern, Remote Call
Control Identity, and Remote IP.
You can configure the system to invalidate all patterns that were learned from unreachable
remote call-control systems. Ifthe SAF connectivity islost, then CCD would invalidate all
learned hosted patterns, and PSTN would beused asabackup toreach these targets.
These are typical CCD issues:
Misconfigured SAF trunks are not exchanging information with SAF forwarders.
CSS and partition issues exist on phones, trunks, and other call-routing components.
Patterns arc not advertised because of a failure or misconfiguration.
Digit or pattern manipulation issues produce incorrect called numbers orrouting plans.
Maximum number of learned patterns is exceeded; nomore patterns arcincluded inthe
master routing table.
CCD troubleshooting is discussed indetail laterina separate module.
Troubleshooting Gatekeepers
3-70
This topic reviews gatekeeper functions, describes the types of RAS messages, and explains
how totroubleshoot discovery, registration, and call admission issues.
Cisco
Unified
Communications
Manager
2125552001
New York
RAS Cisco
. Unified
Communications
Manager
\
A
4085552001
San Jose
Agatekeeper is an 11.323 entity on the network that provides services to 11.323 terminals,
gateways, and multipoint control units. Gatekeepers have mandatory and optional
responsibilities.
The mandatory responsibilities are these tasks, which occur simply because the device is in the
network and has been configured:
Address translation: Calls originating within an H.323 network can use analias toaddress
the destination terminal. Calls originating outside the H.323 network and received by a
gateway can use an E.I 64 telephone number to address the destination terminal. The
gatekeeper must be able to translate the alias orthe F.. 164 telephone number into the
network address for the destination terminal. The destination endpoint can be reached by
using the network address on the 11.323 network. The translation is done by using a
translation table that is updated with registration messages.
Admission control: The gatekeeper can control the admission ofthe endpoints into the
H.323 network. It uses these RAS messages to achieve this: Admission Request (ARQ),
Admission Confirmation (ACT), and Admission Reject (ARJ),
Bandwidth control: The gatekeeper manages endpoint bandwidth requirements. When
registering with agatekeeper, an endpoint will specify its preferred codec. During 11,245
negotiation, adifferent codec might berequired. These RAS messages are used tocontrol
this codec negotiation: Bandwidth Request (BRQ), Bandwidth Confirmation (BCF) and
Bandwidth Reject (BRJ).
Zone management: Agatekeeper is required to provide address translation, admission
control, and bandwidth control for terminals, gateways, and multipoint control units that
are located within its zone of control.
mm
fett^^
Thegatekeeper can provide these optional responsibilities:
Call authorization: With thisoption, thegatekeeper canrestrict access to certain terminals
or gateways basedon policiessuchas (ime-of-day.
Call management: With this option, thegatekeeper maintains active call information and
uses it to indicate busy endpoints or redirect calls.
Bandwidth management: With this option, the gatekeeper can reject admission when the
required bandwidth is not available.
Call control signaling: With thisoption, the gatekeeper canroutecall-signaling messages
between H.323 endpoints byusing theGatekeeper-Routed Call Signaling (GKRCS) model.
Alternatively, it allows endpoints tosend H.225 call-signaling messages directly toeach
other,
Endpoint-to-gatckeepcr signaling is H.225 RASsignaling. This signaling is User Datagram
Protocol (UDP)-based. Signaling messages between gateways are H.225call control,setup, or
signaling messages.
H.225 call control signaling is usedtoset upconnections between H.323 endpoints. The ITU
11.225 recommendationspecifies the use and support of Q.931 signaling messages.
This figure shows how gatekeeper signals thecall setupina multizone gatekeeper network. It
shows the sequence of RAS signaling events.
This shows the sequence of basic gateway-to-gatekeeper signaling that occurs between zones:
1. The phone inNew York dials the phone number 408 555-2001 forSan Jose.
2. The Cisco Unified Communications Manager in New York sends the gatekeeper an ARQ,
asking permission to call the phone in San Jose.
3. The gatekeeper does a lookup, finds that the phone inSan Jose isserviced through the
Cisco Unified Communications Manager, and returns an ACF with the IP address of the
remote Cisco Unified Communications Manager.
4. The Cisco Unified Communications Manager in New York sends an H.225 call-setup
message totheCisco Unified Communications Manager in SanJose.
5. The CiscoUnifiedCommunications Managerin SanJose sends an H.225call-proceeding
message to theCisco Unified Communications Manager inNewYork.
6. The Cisco Unified Communications Manager in San Jose sends the gatekeeper an ARQ,
asking permissionto answer the call.
7. The gatekeeper returns anACF with theIPaddress of the Cisco Unified Communications
Manager in New York.
8. The Cisco Unified Communications Manager in New York and the Cisco Unified
Communications Manager inSanJoseinitiate anJI.245 capability exchange andopen
logical channels.
9. The Cisco UnifiedCommunications Manager in San Jose sets up a call to the phone at 408
555-2001.
10. The Cisco Unified Communications Manager in San Jose sends a call to connect to the
Cisco Unified Communications Manager in New York.
11. Dual RTP streams flow between the IP phones.
2010Cisco Systems, Inc Troubleshooting Call SetupIssues
Gatekeeper H.225 RAS Messages
This sectionlists and explains the 11.225 RASmessages.
Gatekeeper H.225 RAS Messag*
RASprotocol is made of these messages:
- Discovery (GRQ, GCF, GRJ)
- Registration (RRQ, RCF, RRJ)
Unregistration (URQ. UCF, URJ)
* Admission (ARQ. ACF, ARJ)
- Location Request (LRQ, LCF, LRJ)
* Disengage (DRQ, DCF, DRJ)
Bandwidth Change (BRQ, BCF, BRJ)
* Resource Availability (RAI, RAC)
s Request in Progress (RIP)
Status Queries (IRQ, IRR, IACK, INACK)
The figure shows the RAS messages that arcsent toand from the gatekeeper. There are several
types of RAS messages:
Discovery:
Gatekeeper Request (GRQ)
Gatekeeper Confirmation (GCF)
Gatekeeper Rejection (GRJ)
Registration:
Registration Request (RRQ)
Registration Confirmation (RCF)
Registration Rejection (RRJ)
Unregistration:
Unregistration Request (URQ)
Unregistration Confirmation (UCF)
Unregistration Rejection (URJ)
Admission:
Admission Request (ARQ)
Admission Confirmation (ACF)
Admission Rejection (ARJ)
Location request:
I ocaiion Request (LRQ)
ttr
m*
^
-h^^^
Location Confirmation (LCF)
Location Rejection (LRJ)
Disengage:
Disengage Request(DRQ)
Disengage Confirmation(DCF)
Disengage Rejection (DRJ)
Bandwidth change:
Bandwidth Change Request (BRQ)
BandwidthChange Confirmation(BCF)
Bandwidth Change Rejection (BRJ)
Resource availability:
Resource Availability Indicator (RAI)
Resource Availability Confirmation (RAC)
Request in progress:
Request in Progress (RIP)
Status queries:
Info Request (IRQ)
Info Request Response (IRR)
Info Request ACK (IACK)
Info Request NAK(INACK)
)2010 Cisco Systems, Inc.
Gatekeeper Call-Routing Flowchart
Thefigure shows thegatekeeper How routing fiow chart.
KtJCfiS
1 Tech p-efix rratch'>
H
2 Zone pref.i rrakh"'
Targei roie = matched zone
3 Is large! zone local'
4 tech prefix found ir, Step 1'<
N
5 Is target address registered^
In
6 Is defa It tech prefix set"?
I
Hup-off lech prefix>
Is 'arq-rejecl-UFiknown-prefix" Sep
In
Targel zone - local /one
Find local gateway with tech prefix
Selecl local gaieway wiih tech prefi)
J N
Depending on the configuration ofthe gatekeeper that is summarized in the figure, the
gatekeeper can originate a RAS message io a remote zone gatekeeper (LRQ) orreturn a
response (ACF or ARJ) to a local endpoint.
mm
Common Gatekeeper Issues
The most common 11.323 gatekeeper issuesare describedin this section.
Common Gatekeeper Issues
Common gatekeeper issues:
Configuration errors
Registration issues
Duplicate ID
Terminal excluded
- Security denial
- Invalid alias
Call admission issues
- ACF received but get a busy tone
ARJ is null, not enough bandwidth
ARJ received, called party not registered
Notech prefix or no E.164 address for the call
These are the common gatekeeper issues:
Configuration errors
Registration issues:
Duplicate ID
Terminal excluded
Security denial
Invalid alias
Call admission issues:
ACF received but get a busy tone
ARJ is null, not enough bandwidth
ARJ received, called party not registered
No tech prefix or no E.164 address for the call
Troubleshooting Cisco IOS H.323 Gatekeeper
Agatekeeper is an 11.323 network device that provides services such as address translation and
network access control for 11,323 endpoints. Gatekeepers areoptional inan 11.323 network, but
if agatekeeper ispresent, then endpoints must use its services. Gatekeepers communicate to
endpoints by usingRASsignaling, a derivative of Q.93 1.
uûmeTshootmq
H
I
H.323 Gatekeeper
RAS RAS'
Cisco Unified
n;vp Communications Managers*
(Endpoinls)
Useful show commands.
show gatekeeper gw-type-prefix
show gatekeeper status
show gatekeeper zone prefix
show gatekeeper calls
show gatekeeper endpoints
show gatekeeper zone status
Bandwidth management
Dial plan resolution
Discovery and registralion
Call failure
Useful debug commands
debug h225asn1
debug ras
debug gatekeeper main 5
Common gatekeeper issues include bandwidth management and dial plan resolution, which arc
the two important functions ofgatekeeper. Gatekeeper clustering and alternate gatekeepers can
also be issues but are not directly related tocall-routing troubleshooting.
Lndpoint issues tend to fall into one of two categories. The first category involves the discovery
and registration process, You will often need to address these issues after initial deployment.
The second category involves call failure that is caused by call admission orcall routing. Call
admission issues require an understanding of the topology and coder-decoders, orcodecs, that
are involved in calls across gatekeeper-controlled WANlinks. The CACmechanisms that
prevent oversubscription of WAN links can result infailing calls, which might or might not be
a problem.
Various Cisco IOS commands can be useful when you troubleshoot gaiekeeper functions.
Theseare the most commonly usedgatekeeper showcommands:
show gatekeeper gw-type-prefix command displays information about the configured
prefix technology table.
KG-ISshow gatekeeper gw-type-prefix
GATEWAY TYPE FREFIX TABLL
Prefix- 5". I i. Kopoff zone GK-I.:
Staiicdily-ccnflyured gateways (not necessarily currently registered1
10.1.1.1:172c
Zone GK 1 rra =;ci gate*ay list
Prefix: 512' Hcpoff zone GK-2)
Statica-ly configured gateways (not necessanly currently registered
3-76 TroubleshootingCiscc Unified Communications (TVOICE] v8
10.2.1.1:1720
Zone GK-2 master gateway list:
10.2.1.1:33899 Trk-to-SJ_l
show gatekeeper statusdisplays overall gatekeeper status, including zone status.
HQ-lftshow gatekeeper status
Gatekeeper State: UP
Load Balancing: DISABLED
Flow Control: DISABLED
License Status: AVAILABLE
Zone Name: GK-1
Zone Name: GK-2
Accounting: DISABLED
Endpoint Throttling: DISABLED
Security: DISABLED
Maximum Remote Bandwidth: unlimited
Current Remote Bandwidth: 0 kbps
Current Remote Bandwidth (w/ Alt GKs): 0 kbps
Hunt Scheme: Random
showgatekeeper zone prefixdisplays the zone prefix table.
show gatekeeper calls displays the status ofeach outgoing call ofwhich the gatekeeper is
aware.
showgatekeeperendpointsdisplays the status of registered endpoints.
HQ-l#show gatekeeper endpoints
GATEKEEPER ENDPOINT REGISTRATION
CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags
10.1.1.1 33374 10.1.1.1 32809 GK-1 VOIP-GW
H323-ID: Trk-tO-NY_l
Voice Capacity Max.= Avail.= Current.= 0
10.2.1.1 33899 10.2.1.1 32806 GK-2 VOIP-GW
H3 23-ID: Trk-to-SJ_l
Total number of active registrations = 2
show gatekeeper zone status displays the status ofzones that are related tothe gatekeeper.
HQ-l#show gatekeeper zone status
GATEKEEPER ZONES
GK name Domain Name RAS Address PORT FLAGS
5K-1 cisco.com 10.1.250.101 1719 LS
QCS ATTRIBUTES :
DSCP Option : default
BANDWIDTH INFORMATION (kbps) :
Maximum total bandwidth : unlimited
Current total bandwidth : 0.0
Maximum interzone bandwidth : 25
Current interzone bandwidth : 0.0
Maximum session bandwidth : unlimited
SUBNET ATTRIBUTES :
All Other Subnets : (Enabled)
PROXY USAGE CONFIGURATION :
Inbound Calls from all other zones :
to terminals in local zone GK-1 : use proxy
to gateways in local zone GK-1 : do not use proxy
to MCUs in local zone GK-1 : do not use proxy
Outbound Calls to all other zones :
from terminals in local zone GK-1 : use proxy
)2010 CiscoSystems. Inc. Troubleshooting CallSetup Issues
from gateways in local zone GK-1 : do not use proxy
from MCUs in local zone GK-1 : do not use proxy
GK-2 Cisco co- 10.1.250.101 1719 LS
CCS ATTRIBUTES :
TjSCP "prior. : uefauit
BANDWIDTH INFORMATION ' KDps)
Maxir.im total oaridwidLh : unii-.it.ed
Cj;: tr-.t total bandwidth : 0 .C
Maxima- interzone bandwidth : 2b
Maximum session bandwidth : jnlimited
SUBNET ATTRIBUTES :
AI I O'-.hei Subnets : 'Knabled)
FRCXY "JbAGE CONFIGURATION :
Trxour.d Calls froT ai- other zones :
to :e:-'.inals in local zone GK-2 : use proxy
tc gateways in local zone GK-2 : do not use proxy
ro MCUs in ".ccal zcr.e GK-2 ; do not use proxy
frorr. terminals in local zone GK-2 : use proxy
These are the most commonly used gatekeeper debug commands:
debug h225 asnl displays information about the actual contents of H.225 messages, the
RAS in this case.
debuy rasdisplays thetypes andaddressing of RAS messages that aresent andreceived.
This output shows thesuccessful registration, call setup, andcall disconnect.
HQ-lttdebug ras
H.3 23 RAS Messages debugging is on
Mat 3C 09: :4 :44 . 124: RecVJDP_IPSockData successfully icvd message of length
115 fiom 1C .1 .1 .1 :32 8 0 9
Mar 30 03:0-1:44.324. ARQ , seqfc 19; rrvd
Mar 3C 09:04:44.328: I?SOCK_RAS sendto: msg length 43 from
10.I.25 0.IC1:1~19 tc 10.1.1.1: 32809
Mar 30 03:04:44.123: RASLib: :RASSendACT' : ACF {seq# 19) sent to 10. 1.1. I
Mar 10 09:54:44.372: RecvUDP_IPSockData successfully revd message of length
:13 from 1j.2 .1 .1 :32306
Mar 3C 09:54:44.376: ARQ :seq# 21) revd
Mar 30 09:C4 :44.576: IFSOCK_RAS_sendto: msg length 24 from
1 0 .1 .2 30 .10 I :17 1 5 tC 1u .2 .1 .1 : 3 28 0 6
Mar 30 09:04:44.376 RASLib ::RASSendACF : ACF {seq# 21) sent to 10.2.1.1
Mar 30 09 .04 :46."'76 : KecvUDP_TPSockData successfully icvd message of lengri-
100 from 10.". .1.1 -.32809
Mar 30 03.04:46.776: KKQ :seqtf 20) leva
Mar 30 09:04:46.776: 1PSOCK__RAS_ seridto : msg length 64 from
10.1.250.101:1719 to :C . 1.1.1: 328 0 9
Mar 30 09:04:46.776: RASLib::RASSendRCF: RCF (seqft 20) sent to 10.1.1.1
Mar 30 09:04:51.392: RecvUDP_IPSockData successfully revd message ot ler.yrr
270 from 10.2.1 .1:32806
Mar 30 09:04:51.392; DRQ :seqff 22; icvd
Mar 30 09:04:51.332: I?SOCK_RAS_Sendto: msg length 3 from 10.1.250.10" "719
to 10./.1.1: 3 2 8C6
Mar 3C 09:04:51.392: RASI, ib: :KASSendDCF: DCF (Seqfl 22) sent to 10.2.1.]
Mar 30 09:04:^1.396: RecvUDP_IPSockData successfully icvd message of iengtr
273 f rcr. 10.1.1.1: 32309
Mar 30 09:04:51.356: DRQ iseqil 21: icvd
Mar 33 09:04:51.400: I?SOCK_RAS_sendto: msg length 3 from 10 . 1.250.i01 :1^19
to 10.1.1.1: 32305
Mar 30 03:04:51.400: RASLib::RASSendDCF: UCF (seqft 21) sent tc lQ.i.i.l
debug gatekeeper main5 is hidden debug command. It shows how the gatekeeper steps
through call-routing logic, as you can sec inthe next figure.
Troubleshooting Cisco IOS H.323
Gatekeeper (Cont.)
Giridabug gatekeeper main 5 ,_,., .ao.ru am
Mr 30 09:08,45.996: //xxx*/x:*^/Wg*-"SSrv arq.
iwjo-ij* ere crv-0x3, answarCsll-O
HîrM.0.4S.S. //aB*83FC70300/a0AB3Fn70300//rBB9rv_get_addrinfO:
(512200H Matched teeh-preti* 512
Mar 30 09:09 =45.996- ,,.,,, -bout Co check
//g0A83FD703 00/S0AB3FD703OO/OK/rsv_q. aelect_vlazone .
th. o,rce sida. BcC.oDBP-0x49J4PACB
H.r 30 09:08:45.996: vi.,on-- matched zone is
//8 0AB3PD703 00/B0A83F=r7O300/GK/raBarv_arq_Eiel8ct via.one.
GK-1, apd i iQvianaMlen.O
the destination aide, diit_ionep-0*49J4FfSBG
//8OAB3FD703O0/B0AB3FD7t33DO/GK/rasrv_alq_select vii.soe, matched ions is
GK-2, and lOULvianainelen-O
The figure shows the output of the debug gatekeeper main 5command. This is one of the
most useful commands for the gatekeeper troubleshooting commands because ,t shows all
stages ofthe gatekeeper decision tree.
This ,s the successful call setup. At the beginning, ARQ was received. It contained the called-
party number 5122001 that matched with the configured technology prefix.
The call is sourced in the zone GK-1 and it will proceed to the destination zone GK-2.
)2010 Cisco Systems. Inc.
M,
//eOA83FDirj3 0O/S0A83FD7O3OO/GK/g)! zon
IfI"tE ione= GK-2, caii direction
Mar SO 09,08:46.000:
'/SOAa3Prj703 00/BOA8 3FE7 0300/GK/gK
Mar 30 09 : 06 : IE . 014 : //xiumjij
arqp,0,4a07843C,i:rvsOJ:a003, anBwerCalJ=l
:d de9tiJtiori0dpoin'rirdL"eLTi8o3FD7C30c/GK/9k 9u seiect px: sout
It 30 09,08: 46.000: zones
get proxy uaage: local IOne= GK-1
1, epcype= 67586 be entry= 0
</GK/gk
The calhs progressed across the zones, where the local zone is GK-1 and the remote zone is
Troubleshooting CiscoUnified Commur
ications (TVOICE) v8 i
2010 Cisco Systems, inc
Troubleshooting Gatekeeper Registration
This section describes how to troubleshoot gatekeeper registration issues.
Troubleshooting Gatekeeper Re*
Problem report:
Manager not showing up as
registered tothe gatekeeper Hp323 Gatekeeper
Possible Causes:
- Network connectivity
Gatekeeper misconfiguration
Endpoint misconfiguration
Duplicate H.323 ID
Endpoint not authorized to register
RRJ
RRQ
Cisco Unified
Users report that they cannot place calls between Cisco Unified Communications Manager
clusters. One of the first things you do to troubleshoot this problem is check that the Cisco
Unified Communications Managers are registered to the gatekeeper. Use the command show
gatekeeper endpoints to check this information.
When an H.323 endpoint, like Cisco Unified Communications Manager, does not register, it
could be a problem with the registration process. These are the most common sources of
problems when dealing with endpoints not registering:
A network connectivity issue can prevent the communications between the endpoint and
the gatekeeper. Unicast network connectivity must exist for the registration process. Try the
ping or traccroute to verify the connectivity.
A configuration error on the gatekeeper can prevent the registration of the endpoint.
A configuration error on the Cisco Unified Communications Manager can prevent the
registration to the gatekeeper.
Registration errors can be caused by any of these reasons:
A duplicate 11.323 ID is already registered on the gatekeeper.
The IP address of the endpoint that is used for registration is not authorized to
register.
The zone that the endpoinl is attempting to register with is not configured on the
gatekeeper.
Verify Configuration That Can Affect the Registration
This secriondescribes which parts of the configuration to verify.
3-82
Device > Trunk
Device > Gatekeeper
,.f * <t*to& SJ-)*. raûi
ity OonlKjuu.l.on Uun 'V i AfJccl the
gatekeeper
loae local GK-1 Cisco.coa 10.1.350.101
zone local GK-2 ciaco.com
gw-typa-prafi* 511* bopoff QK-1 g Ipaddr 10.1.1.1 1720
gw-type-prefix 512* hopoff GK-2 gw ipaddr 10,2.1.1 1720
bandwidth interzone default 25
no shutdown
Verify that the configuration on the gatekeeper is correct and includes these configurations:
Verify that there is a local zone that matches the zone on the endpoint {casematters).
If the no zone subnet zonename default enable and zone subnet zonename suhnet-
address/bits-in-mask enable commands have been used to restrict which endpoints arc
allowed to register, ensure that the address allowed to register includes the endpoint (or
endpoints) in question.
Verity that the gatekeeper has the command no shutdown under the gatekeeper
configuration to enable the gatekeeper functions.
Verify that the configuration of the gatekeeper and trunks on the Cisco Unified
Communications Manager server are correct and include the following configurations:
The IP address of the gatekeeper is correct and reachable.
The trunk settings are correct and include the following:
The correct gatekeeper is selected.
Ihe terminal type is set to "gateway".
The technology prefix is set to match the gatekeeper.
The zone name matches the name of a local zone on the gatekeeper.
A local zone matches the zone on the endpoint (case matters).
This example shows the successful registration of two independent Cisco Unified
Communications Manager servers to the single gatekeeper:
GKft snrjw qatekeepe r e:iapoi nt s
GATEKEEPER ENDPOINT REGISTRATION
Troubleshooting Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Syslems, Inc
CallSignaiAddr Port RASSignalAddr Port Zone Name Type Flags
10.1.1.1 32990 10.1.1.1 32B02 GK-1 VOIP-GW
H323-ID: Trk-to-NY_l
10.2.1.1 32973 10.2.1.1 32807 GK-2 VOIP-GW
S323-ID: Trk-to-SJ_l
Total number of active registrations - 2
One of the most useful tools to troubleshoot gatekeeper discovery and registration issues is the
debug h225 asnl command. This command displays detailed messages that can assist in
narrowing the scope of possible causes. The common issues between gatekeepers and endpoints
are Registration Rejects (RRJs), where "REJECT REASON" can show the reason for the
rejection. The four rejections that are the most typical are the following:
RRJ: rejectReason duplicateAlias
RRJ: rejectReason terminal Excluded
RRJ: rejectReason securityDenial
RRJ: rejectReason invatidAlias
RRJ: rejectReason discoveryRequired
2010 Cisco Systems, Inc Troubleshooting Call Selup Issues 3-83
Troubleshooting Gatekeeper Call Routing
This section describes how to troubleshoot issues of call routing with a gatekeeper.
>/
$
Possible Causes:
Connectivity between clusters issues
Dial plan misconfiguration
Incorrect IP address returned
Called party is not registered
Insufficient bandwidth due to CAC
iunUilt.* hOuthiO ^-itckri I * f C it
HoutuHj
Problem report:
Cisco Unified Communications Manager registered but phones not
able to make intercluster calls
i-;
Users are reporting that calls between Cisco Unitied Communications Manager clusters are
failing. You use the command show gatekeeper endpoint to verify that the two Cisco Unified
Communications Manager servers that arc involved in the call are registered.
The call might be failing because of normal network operations. If the Cisco gatekeeper is used
as a CAC mechanism to prevent oversubscription of a WAN link, the fact that the call is failing
might be normal. You might consider the use of an automated second-choice path across
another WAN link, or PSTN, to avoid the call failing. If you still think that the call shouid be
ompleting successfully, here are some of the possible causes of the problem:
Connectivity issues between the endpoints (but not between the endpoints and the
gatekeeper, because they are successfully registered)
A misconfiguration on the Cisco Unified Communications Manager dial plan that
determines when to invoke the gatekeeper, or miscon figured gatekeeper call routing
A misconfiguration on the gatekeeper that returns the incorrect IP address in the ACF
message
Called party not currently registered
Lack of bandwidth because of CAC mechanisms
Troubleshooting Cisco Unified Communications (TVOICE) vB0 2010 Cisco Systems, Inc
H.323 Gatekeeper CAC
The H.323 gatekeeper is commonly used to perform CAC between multiple Cisco Unified
Communications Manager clusters.
H.323 Gatekeeper CAC
H.323 gatekeeper is optionally usedfor intercluster CAC.
H.323 gatekeeper sets the maximum bandwidth permitted
per zone or between zones.
* Call bandwidth is twice the bandwidth of the audio codec:
G.711: 128kb/s
G.729: 16kb/s
H-225 Trunk
ftprylBKbft J Max:100kb/s | ^RQM28kb/s
G729 Call ^ "^j G.711 Call
Not enough BWi
Common ARJ Reject Reasons
Theseare twocommon reject reasons that youwill see inan ARJmessage:
calledPartyNotRegistered: This isacommon reason for rejection from the local or
originating gatekeeper when the gatekeeper has no information on where the called number
needs to be terminated. There are two scenarios in which this problemcan occur:
The call terminates at a gateway and thegateway is notregistered with the E. 164
address or with a technology prefix. Toresolve this issue, make sure that the
gateway registers with a technology prefix tothe gatekeeper. Use the show
gatekeeper zone status command toconfirm that the endpoint isregistered.
The called party isaterminal in a remote zone and does not have aproxy that is
enabled inthe same gatekeeper zone inwhich it is registered. Bydefault, Cisco IOS
gatekeepers use a proxy for interzone terminal calls. Use the show gatekeeper zone
status command to view this information.
requestDenied: The reason for this rejection isthat the endpoint-requested bandwidth
exceeds the limit that isconfigured inthe gatekeeper. To resolve this rejection, increase the
bandwidth in the gatekeeper with the help ofthe bandwidth command under the gatekeeper
mode, or lower the bandwidth request fromthe endpoint.
Per-call bandwidth information exchange on H.323gatekeeper uses two types of RAS
messages:
Admission messages (ARQ, ACF, ARJ) areused at theinitial stage of call setup.
Bandwidth management (BRQ, BCF, BRJ) messages are used at later stages toadjust the
per-eall bandwidth.
Bandwidth limitations arc configured differently on different Cisco products and for different
features. Cisco IOS H.323 gatekeeper calculates the bandwidth per call as twice the bandwidth
of the audio codec. AG.729 call consumes 16 kb/s ofthe configured bandwidth and aG.71 1
call consumes 12H kb's ofthe configured bandwidth and similarly for other audio codecs.
Cisco IOS 11.323 gatekeeper CAC is implemented by using the bandwidth command.
Troubleshooting Cisco Unified Communications (TVOICEl v8 0 2010 Cisco Systems Inc
Troubleshooting Gatekeeper CAC
This section demonstrates how to troubleshoot gatekeeper CAC.
Troubleshooting Gatekeeper
gatekeeper
bandwidth interzone default 25
OK*show gatekeeper tone status
GATEKEEPER ZONES
GK name Domain Name RAS Address
PORT FLAGS
GK- 1 cisco.com 10.1.250.101 17 19 LS
QOS ATTRIBUTES s
DSCP Option s default
BANDWIDTH INFORMATION (Kbps) :
Current total bandwidth = lt> 0
Maximum interione bandwidth : 25
current interione bandwidth ; 16.0
... output truncated ...
The figure shows the gatekeeper that has been set up for limiting bandwidth for calls between
gatekeeper zones to 25 kb/s.
The second part ofthe figure shows an extract ofthe show gatekeeper zone status command
that displays an existing G.729 call between zones that consumes 16 kb/s. This is the complete
output of the command:
GKiishow gatekeeper zone status
GATEKEEPER ZONES
GK naive Domain Name
GK-1 cisco.com
QOS ATTRIBUTES :
BANDWIDTH INFORMATION (kbps) :
Current total bandwidth : 16.0
Maximum interzone bandwidth : 25
Total number of concurrent calls : 1
SUBNET ATTRIBUTES :
All Other Subnets : (Enabled!
Inbound Calls from all other zones :
to terminals in local zone GK-1 : use proxy
to gateways in local zone GK-1 : do not use proxy
from, terminals in local zone GK-1 : use proxy
GK-2 cisco.com
RAS Address PORT FLAGS
10 . 1.250.101 1719 LS
10.1.250.101 1719 LS
COS ATTRIBUTES :
BANDWIDTH INFORMATION 'kbps; :
Maxima- total bandwidth : unlimited
Curre"-. '.jta. candwid;. h : 16.0
"axi-um 1nterzone bar.awidt.h : 2 5
Current i.-.r.erzer.e bandwidth : 16.0
Max1. rrj~ session bandwidtn : unlimited
:'cta": r.UTiber ol concurrent calls : 1
SUBLET ATTRIBUTES :
All Other Subnets : '.Enabled)
Ir.bounc Cai-s from all other zones ;
to terminals n: local zone GK 2 : use proxy
"o gateways m local zone GK-2 : do not use proxy
O.itbo.jnd Calls to all ether zones :
trcrr, terminals in lo;:al zone GK 2 : use proxy
frcT. gateways in local zone GK-2 : do not use proxy
Liom MCUs :r. lecal zone GK-2 : do not use proxy
By using the show gatekeeper calls command, yoti can also show the call that has been
established:
GKtfshow gatekeeper calls
Totai r.umoer of active calls =- I.
GATEKEEPER CALL INFO
Loca.Oal-IE Age{sees! BW
' 32S45 11 ]6(Kbps!
Enapt.s': Alias E,l64Aadr
sre E?: Trk-to-NY__l 2001
Call SignalAddr Port RASSignalAddr Port
10.1.1.1 32990 1C.1.1 .1 32802
Endpl 3.; Alias E,J64Addr
dst LP: Trk-tO-SJ I 5122001
.:allSignalAddr Port RASSignalAddr Port
:--" -I .52973 10.2.1 .1 32007
3-88 Troubleshooting Cisco Unified Communications (TVOICEl v8 0 2010 Cisco Systems Inc
Troubleshooting Gatekeeper CAV
OK#d ebug h!!5 asnl
vali. e RasHeimage ::- attains ionReque
{
deBtlnatioolnfo
{
dialedDigits : -5122001"
}
srciafo
{
dialedDigits : -3001"
)
bandwidth 160
gatekeeparldentitiar ("GK-1"]
)
Bt :
val lie RasMesBage ::- admisa ioDRejec
{
TuquestEegHum 6
t :
re]ectReason requBstDeoiBd : NULL
>
During the existence ofthe G.729 call, and the bandwidth being limited to25 kb/s, only 9kb/s
is left for additional zone to zone calls.
This figure shows the call admission request for an attempted second G.729 call between the
zones. The admission request shows the called (destinationlnfo) and calling (srclnfo) numbers
along with the bandwidth of 16 kb/s that has been requested for this attempted call.
The admission request has been instantly followed by the admission reject message that denies
the call becauseof insufficient bandwidth (the reasonis shownas NULLin the output).
Troubleshooting Cisco Unified Border Element
This topic reviews Cisco Unified Border Element general functions and Cisco Unified Border
Element function in the CCD process. This topic also describes how to troubleshoot Cisco
Unified Border Element in the CCD process.
3-90
v 11.
j-iement Hinctio
Cisco Unified
Communications
Manager
$
H 225/H.245
or SIP
RTP
H 225/H 245
OF SIP
s
RSVP j!
Cisco Unified
Communications
H.225/H245 Mana^'
or SIP
I
RTP
Cisco Unified
Border Element
Cisco Unified
Border Element
SCCP or SIP
SCCP or SIP
y
Cisco Unified Border Elements areused tointerconnect Voll1 networks with each other.
Originally, to enable complete billing capabilities within VoIP networks, service providers used
Cisco Unified Border Elements. However, the functionality tointerconnect VoIP networks is
becoming more important for enterprise VoIP networks as well, because VoIP is becoming the
new standard for any telephony solution.
Cisco Unified Border Hlements can route acall from one VoIP dial peer toanother Voli' dial
peer. VoIP dial peers can also be managed by cither the SIP or H.323. As aresult, the ability to
interconnect VoIP diai peers also includes the ability tointerconnect VoIP networks that use
different signaling protocols or VoIP networks that use the same signaling protocols but that
face interoperability issues.
When interworking signaling protocols, aCisco Unified Border Riement supports these
combinations:
H.323-to-H.323: ACisco Unified Border Element fully supports fast-start with slow-start
interworking in all directions.
11.323-lo-SIP: H.323 fast-start to SIP early offer interworking is fully supported. An H.323
slow-start toa SIP delayed offer is another supported alternative.
SIP-to-SIP: Early offer and delayed offer arefully supported ona Cisco Unitied Border
Element in all directions.
Because aCisco Unified Border Element is asignaling proxy, it also processes all signaling
messages regarding the setup of themedia channels. This processing enables a Cisco Unified
Border Element to affect the flow ofmedia traffic. Two options exist: media flow-through and
media flow-around.
TroubleshootingCisco Unrfied Communications (TVOICE) v8 0
This figure shows aCisco Unified Border Element that is configured for media flow-through.
The Cisco Unified Border Element processes the signaling between the two Cisco Unified
Communications Manager clusters, and the source IP addresses ofthe IP phones are replaced
by the Cisco Unified Border Element IP address. Both endpoints might have the same IP
address, but because the Cisco Unified Border Element is involved, nointerworking issues
arise.
)2010 Cisco Systems, Inc Troubleshooting Call Setup Issues 3-91
Common Cisco Unified Border Element Issues
This section shows the issues that are commonly seen when Cisco Unified Border Element is
used to interconnect Cisco Unified Communications Manager clusters.
3-92
Lorn n; li>C<> Uri
Cisco Unified Communications Manager misconfigured
Cisco Unified Border Element misconfigured
Their configuration does not mutually match
MTP not allocated, DTMF or codec mismatch
Incorrect digit manipulation on Cisco Unified Communications
Manager or Cisco Unified Border Element
Call capacity exceeded. CAC failure
H 323
J
The common Cisco Unified Border Element issues arc the following:
Cisco Unified Communications Manager can be misconfigured in terms oftrunk or
gateway settings (depends what is used for Cisco Unified Border Element) or dial plan that
progresses the calls to Cisco Unified Border Element.
Cisco Unified Border Element can be misconfigured and not function as expected. Dial
peers often have many points where errors can occur,
Theconfiguration at theCisco Unified Border Element andtheconnected Cisco Unified
Communications Managers might not mutually match. You can experience issues such as
not allocating MTP when necessary orconfiguration ofincompatible dual tone
multifrequency (DTMF) and different codecs that are requested.
CSS and partition issues are also common. Trunks and gateways that realize connectivity to
the Cisco Unified Border Element could beconfigured with an incorrect CSS for inbound
calling orroute patterns with incorrect partitions for outbound calling.
Incorrect digit manipulation on Cisco Unified Communications Manager or Cisco Unified
Border Element might be present and producing unexpected call routing.
Cisco Unified Border Element can also be used for CAC functions. Ifthe capacity ofthe
network was reached, CAC returns a failure.
Troubleshooting Cisco Unified Communicalions (TVOICE) v80
*" Cisco Unified Border Element Troubleshooting Commands
This section provides information that you can use to troubleshoot Cisco Unified Border
Element configuration.
Cisco Unified Border Eleme
Troubleshootinq Commands
debug h225 asnl (H.323-H.323 and H.323-SIP scenarios)
debug h225 q931 (H.323-H.323 and H.323-SIP scenarios)
debug h225 events (H.323-H.323 and H.323-SIP scenarios)
debug h245 asnl {H.323-H.323 and H.323-SIP scenarios)
debug h245 events (H.323-H.323 and H.323-S1P scenarios)
debug cch323 all (H.323-H.323 and H.323-SIP scenarios)
debug voip ipipgw (H.323-H.323 and H.323-SIP scenarios)
debug voip ccapi inout (H.323-H.323, H.323-SIP, SIP-SIP scenarios)
debug ccsip messages (H.323-SIP, SIP-SIP scenarios)
show voip rtp connection
show call active voice brief
Cisco Unified Border Element can be implemented in H.323, or SIP, or H.323-SIP
environments. The commands to be used depend mainly on which signaling a Cisco Unified
Border Element uses.
Before you use any of the following debug commands, make sure to clear the log before a call
for debugging is made. The troubleshooting commands to use are sorted based on scenarios.
In H.323 to H.323 Cisco Unified Border Element scenarios use these troubleshooting
commands:
debug h225 asnl commanddecodes any H.225 messages into an ASN.l format.
debug h225 q931 command decodes H.225 Q.931 messages.
HQ-litdebug h225 q931
Q931 Message IE Decodes
Protocol Discriminator oxoa
CRV Length 2
CRV Value 0x0003
Message Type 0x05: SETUP
Bearer Capability: Length Of IE=3
Data B09GA2
Calling Party Number: Length Of IE-6
Data 008132303031
Called Party Number: Length Of IE=8
Data S035313232303031
User-User: Length Of IE=245
Data
052OBO0g00 03914A00050201805334401F0032003000 30 0 03100000000000
0000000000000000000000000000000000000000000000000000000000000
00OC0022COE5000012 Of436973636F43616C6C4D616E6167657200310001
S5AF17 21BB03 0003010A0102 0COOD51D800007000A01010106B8110000A71
C10A0102 0C3 2 0212 00 00000C6 013800A04 0001000A010101600B1D40FFFEO
00000000000000000
00000000000000000
300945
D55AF1
04010
;5334(
L721BI
)04C6(
O0A71D
030003
138011
3-93
14O0C1Q00A010IC1600A0O0A0IO1O16Q0B0I 0 00 10001 0001 0 01 OAOOl 001 4 0 14 0B5O00 0 12 0D8 2 04
0020040001030C03C00103
Q93 1 Message li; decodes
Protocol DiS"r ini nator : 1x08
CRV Length
CRV Value
Message Type
User-User: Length Of IE=-51
Data
C5218006OGO9i4A00042S10500G012400I3C050I00044300110000A71D55AF1721BBC3000301
Q921 Message IE Decodes
iJrct oco". r., scr :m .r.at oi 0x08
CSV Lengt":. : 2
CRV Value : 0x8003
Message Type : 0x01: ALERTING
Display: Length Cf IE=0
Data
Signal: Length Cf IE=1
Data ."I
User-Usei : Length Cf IE=51
Data
OS23801110S^14AG0142SCOB^l'OOC]24 1'C13COSai0036C3 00110000A/lDbbAF17218B03G00 30i
0A0102CC110G01G01CS10IG0
53 31 Message IE Decodes
Protocol Liscnminator : 0xC3
CRV Length : ;
CRV Value : 0x8003
Message Type : 0x6E: NOTIFY
Notification Irid: length Of IE-1
Data F'.
Data
Connected NL-Dti: Length Of IE-6
Data COOG32303C31
User-Use:: Length Ci IE=33
. . . li ur.catâ . .
debug h225 events command shows all the events that are related to 11.225 state machine.
debug h245 asnl command decodes any of 11.245messages into ASN. I fomiat.
HQ ISdebug h245 asnl
Mar 3C lb:28 :bl.231: H24= MSC INCOMING ENCODE BUFFER::-
12700106QCC881^500CA8013 800!")3CG001 000001 000001 0O0O0CC0O1 0 0 01 000 5 8 0 0 0 0 0 3 S 0A0 0GC
;70008S24 3ClC5CiaCCOO;20C04F80000220404F800003a5014080000485Q11080002B8501bOOG
3G0002G10CG1C00201G0C3 0004 0O0O2B
Mar 30 15:28:51.235: H245 MSC INCOMING PDU ::=
value Mult iêdiaSysteriiControlMessage ::- request : terminalCapabi 1itySet :
...truncated. .
value Mul tiTv.ediaSysteir.Cor.trolMessage ::= request : termi na ICapabi] tySet :
. . . rrur.rat ed . .
Mar 31 15:28:51.33";: H24.-. MSC OUTGOING ENCODE BUFFER::-
027C0i:GCD183175C0G780I33G0014O0010O0001O0O0010O00OCC0010O0100O1800OlA8 301ICsl
010220C11310800101G0C00210001A
Mar 30 15:28:51.339: H245 MSC OUTGOING PDU ::-
value Mult iT.ediaSystc-iviControlKessage ::= request : masterSlaveDeteiminat 10:1
...truncated. ..
Mar 30 15:73:51 .339: :124 = MSC OUTGOING ENCODE BUFFER::= O1003C000O
Mar 31 15:28:51.330: H245 MSC OUTGOING PDU : :--
val ue Mul t iTiiediaSystemCont rolMessage ::= response : terminal Capabi 1itySetAck :
. . . t n,.ica:ed. . .
Mar 30 15 : 28.51 .39S : h'245 MSC INCOMING ENCODE BUFFER::- 218001
value Mul timediaSyst e~Cor.trclMessage ::- response : terminal Capabi 1itySet.Ac< :
...truncated ..
cxecoi
0x02: CALL PROC
Mar 30 15:28:51.395: h245_decode_one_pdu: H245ASNDecodePdu re - 0,
bytesLeftToDecode = 0
Mar 30 15:28:51.399: H245 MSC INCOMING ENCODE BUFFER::= 2080
value MultimediaSystemControlMessag^ ::= response :
masterSlaveDeterminationAck :
. . . crimested. . .
value MultiT.ediaSystemControlMessage :;= request ; openLogicalChannel
{
forwardLogicalChannelNumber 1
forwardLogicalChannel Parameters
{
dataType audioData : g711Ulaw64k : 20
multiplexParameters h2250LogicalChannelParameters :
sessionID 1
rrediaControlChannel unicastAddress : iPAddress :
{
network '0A01FA65'H
tsapldentifier 17699
}
silenceSuppression TRUE
Mar 3C 15:28:51.403: H245 MSC OUTGOING ENCODE BUFFER::=
03000000OC601380OB0500010O0A01FA6545238 0
Mar 30 15:28:51.447: H245 MSC INCOMING ENCODE BUFFER::^
0300000OCC6 013800B050001000A0101010FA100
Mar 30 15:28:51.447: H245 MSC INCOMING PDU ::=
value MultinediaSystemControlMessage ::= request : openLogicalChannel :
{
forwardLogicalChanneiNumber 1
forwardLogicalChannelParameters
I
dataType audioData : g711Ulaw64k ; 20
muItiplexParameters h225OLogicalChannelParameters :
sessionID 1
mediaContrclChannel unicastAddress : iPAddress :
{
network 'OA010101'H
tsapldentifier 4001
1
silenceSuppression FALSE
)
}
...truncated...
debug h245 events command shows all the events that are related toH.245 state machine.
debug cch323 all command shows all debugs that are related to H.323 call control.
debug voip ipipgw command shows the major call legs processing events.
HQ-lttdebug voip ipipgw
Mar 30 15:38:39.106: //15/002F904E0500/H323/setup_ind: Receive bearer cap
infoXRate 16, rateMulc 0
Mar 30 15:38:39.106: //15/002F904E0500/H323/cch323_fastStart__codec_match: ccb-
>remote_ fastStart=Ox47F5082C
Mar 30 I5:3B:39.106: //15/002F904E0500/H323/cch323_fastStart_Codec_match:
symm__niask =l , tempOtherCodec=5, templocalCodec=5,
audioFastStartArray=0x489l2F8C
Mar 30 15:38:39.106: //15/002F904E0500/H323/cch323_fastStart_codec_match:
Setting local audio_cap_mask
3-96
Mar 30 15:33:39.105: //15/002F904E0500/H323/cch323_fastStart_codec_match;
Inbound legs state_rrlc_mode =OxlOF
Mar 30 15:38:39.106": //I E/002F904E0500/H323/cch323_fastStart_codec match:
Executing legacy code
Mar 30 15:38:39 iCG : ,'15/002F904EC500/H323/cch323_build_o1c_for ccapi-
audioFasLSiar-Aridy-:x4e9i 2L-'8C
Mar 30 1.-: 33: 39.110: ',/15./002F3 04E05 00/H3 23/cch323_build_olc tor_ccapi:
c-iar.nel_ir.ro ptr=0x47F6A4 3C, ccb ptr=0x4881 33D0
Mar 30 lr:38:39.110: /15 '002F904EC500/H323/cch323 build olc for Ccapi-
Channel Inf crmat ier.;
Icgica'. Channel Kumtti-i : rwdi : 1
logical Channel \'ui='ber irev): 6^535
Cnannel adaret-s <wd/rev; : 10.] .].1
k.P Channel (fwd.'revj : 24594
P7C1"- Cnanr.el ifwd/rev i: 24595
QoS Capability ifwd/revj : 0
Syrr.rret ric Audio Codec : 5
Sy"~e;.r-.c Audio Codec Bytes: 160
Flow Mooe: o
Silence Suppression: j
. r ru;ica ceo . . .
debug voip ccapi inout command shows call control API processing for all signaling
protocols.
In 11.32} to SIP Cisco Unitied Border Element scenarios, use these troubleshooting commands:
debug h225 asnl
debug h225q931
debug h225 events
debug h245 asnl
debug h245 events
debug cch323 all
debug voip ipipgw
debug voip ccapi inout
debug ccsip messages command isone of the most useful commands to troubleshoot SIP
call control processing.
In SIP to SIP. Cisco Unified Border Element scenarios use these troubleshooting commands:
debug ccsip all command is like debug ccsip messages but shows less output tobe used
when troubleshooting Cisco Unified Border Element in high call volume environments.
debug \oip ccapi inout
Alter acall is up in any ofthe preceding scenarios, use these commands to display it:
shou voip rtp connection command shows the RTP call legs ofan established call.
KC-lfishow voip rtp connection
VoIP RTP active connections :
No. Callld dstCalKd Loca-RTP RmtRTP LocalIP RcmotelP
1 : ' '- 16418 24.'=96 10.!.750.101 10.1.1.1
'' -6 i7 15354 29446 10.12.1.101 10.2.2.18
Found 2 active RTF oonnectxons
show call active voice brief command showsmoredetailed information about RTP call
legs ofanestablished call than the show voiprip connection command.
Troubleshooting Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems Inc
HQ-lOshow call active voice brief r ,rio <=tate>
<TD>. <CaUID, <start>m8.<index> +<connect> pid:<peerîd> <dir> <addr> <atate>
dur hh:m^:ss tx:<packets>/<bvtes> rx:<packets>/<bytes>
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>m8 lost:<lost>/<early,/<late>
delay :< last. =/<~in>/<;max>ms <:codeo
media inactive detected:<y/n> media cntrl rcvd:<y/n> timestamp:<time>
long duration call detected:<y/n> long duration call duration :<sec>
timestamp-. <time>
MODEMPASS <method> buf:<ills>/<drains> loss <overall%>
<multipkt>/tcorrected>
last -cbuf event time>s dur:<Min>/<Max>s
FR <protccol> [int dlci cid] vad:<y/n> dtmE:<y/n> seq:<y/n>
<codec> (payload sizel
ATM .protocol, lint vpi/vci cid) vad:<y/n> dtmf:,y/n> seq:<y/n>
<: codec > ipayload size) . .
Tele <int/l.calHD) [channeled] tx: <tot>/<v>/<fax>rr.s <codec> noise:<l>
acom;tl> i/o;<l>/"=l> 3Bm
MODEMRELAY infa:<rcvd>/<sent>/<resent> xid:<rcvd>/<sent>
total:<rcvd>/=sent>/<drops>
speeds (bps): local <rx>/<tx> remote <rx>/-=tx>
Proxy <ip>:<audio udp>,<video udP>,<tcpO>,<tcpl>,<tcP2>,<tcp3> endpt:
<type>/=manf >
hw- <rea>/<act? codec: <audio>/<video>
tx: "udio pkLs>/<audio byteS>,<video pkts>/<video t,ytes>, <tl20 pkts>/<tl20
brx!"audio pkts>/*audio bytes>,<video pkts>/<video bytes>,<tl20 pkts>/<tl20
bytes*
Telephony call-legs: 0
SIP call-legs: 1
H323 call-legs: 1
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
27BB : 17 19461380ms.1 +30BO pid.1 Answer 2001 active
dur 00:00:12 tx:638/102080 rx:636/101760
IP 10.1.1.1:24596 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms
g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
27BB : 13 19461400ms.1 +3050 pid:2 Originate 5122001 active
dur 00:00:12 tx: 636/101760 rx: 640/102400 n/n/(>mG
IP 10.2.2.19:29446 SRTP: off rtt:0ms pl.O/Oms lost:0/0/0 delay:0/0/Oms
g711uiaw TextRelay: off
"media inactive detected:n media contrl rcvd:n/a timestamP!n/a
long duration call detected:n long duration call duratiomn/a timestamp:n/a
Telephony call legs: 0
SIP call-legs: 1
H323 call-legs: 1
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
Troubleshooting Call SetupIssues 3-97
The figure shows the topology and addressing information for the troubleshooting
demonstration that is outlined on the next pages.
Verify the configuration at the Cisco Unified Border Element
and Cisco Unified Communications Manager
' Cisco Unified
s*V *l#
10.1.1.1
Site Code:
Cisco Unified
Border Element
H.323
t
LoopbackO:
10.1.250.101
Cisco Unified
10.2.1.1
Site Code:
The Cisco Umtled Border Element performs H.323 to SIP interconnection, where the left Csco
Unified Communications Manager is connected using 11.323 and the right Cisco Unified
Iommumcations Manager is connected using SIP.
Because the sites use overlapping directory numbers, the site code was added. The left site uses
tne site code M1and the right side usesthe site code 512.
Verify thauhe corresponding configuration of the Csco Unified Border HIement and both
I isco Unified Communications Managers match.
Troubleshooting Cisco Unified Border ElementH.323 Side
This section focuses on the H.323-side troubleshooting.
Troubleshooting Cisco Unified Border
ElementH.323 Side
Cisco Unified
H.323
Cisco Unified
Border Element
10.1.1.1 LoopbackO: 10.1.250.101
voice service voip
allon-connections h323 to sip
allow-connections sip to b323
I
interface LoopbackO
ip address 10.1.250.101 255.255.255.255
h323-gateway voip interface
h323-gateway voip bind srcaddr 10.1.250.101
I
dial-peer voice 1 voip
destination-pattern 5112...
session target ipv4;10.1.1.1
dtmf-relay h245-alphanumeric
codec gTllulaw
This figure shows the basic configuration of Cisco Unified Border Element for the H.323 side
that connects to the left Cisco Unified Communications Manager.
>2010 Cisco Syslems. Inc
eshc-otiiKi Cisco linn ;t h 'r*l
:...-* [,..,,,n, 1
C- '..,..-k-.' J.! inam He'Mil
l-l"'- TI.-HE'IT ^'-t' PÎBÎ
1 mk.t,.*,*....;,,.,,,!
" .!-.. -'.!!.. :.i => 'jo .
| .,- f.., F, fr-,-.J5 -., ,-.^,..^1
..,,,- M.. .,.,;=. -,.,,-.,
This figure shows the correspondingconfigurationof Cisco UnifiedC'ommunications Manager.
There are two methods of defining an 11.323 trunk to the Cisco Unified Border Element on the
Cisco Unitied Communications Manager:
An 11.225 trunk (gatekeeper controlled) toward the Cisco Unitied Border Element is
configured. The advantage of this method is the gatekeeper-based CAC.
An H.323 gateway is representing the Csco Unified Border Element. Theadvaniage of this
method could be in the case when Cisco Unitied Border Element also acts as a PSTN
gateway. This is a simpler configuration method.
An H.225 trunk (nongatekeeper controlled) toward the Cisco Unified Border Element is
configured. This is the simplest contiguration option.
This Cisco Unified Border Element acts as the 11.323gateway. If the Cisco Unified Border
Element does H.323-to-H.323 calls, an MTP is not mandatory as long as the Cisco Unified
Border Element release is 12.4(6)Tor later and the Cisco Unified Communications Manager is
Version 4,1 or later. Otherwise, you might need a hardware or software MTP as coresident on
the same router as the Cisco Unitied Border Element or elsewhere. Also make sure lhat the
check box "Wait for Far End 11.245Terminal Capability Set" is disabled for II.323-to-SlP
interworking: otherwise, the call setup tails because the SIP side does not send this information.
3-100 Troubleshooting Cisco UnifiedCommunications (TVOICE) v8 0 2010Cisco Systems Inc
Troubleshooting Cisco U
ElementH.323 Sid
r-ctn Rvuhng Irfwrwution - Lnfcaciriricallt
j <..).T*'<.rJ C-t*s' 4 ,
^liM*l . *.!-
,":"",,,'t*S"" """">
!..
i ! =r)".tijnaHL**4j'lf (in'trf - Ihttjund
|| Êmblf 1r.E*uM Fins' *|1 1
"
1 c*.(>^ <i.i.a.n' Orq.n.l.r si.
;*itim Brt.-=mm" r>.r...
C**1h| 0*"' Tl nlrmbur Itm ur*=hQWil CUD C=JIMrlJfl*l
-*4nflprtT IEmclH ( W*rmpft * dtca " all<i*fl'" .
;<"d Nmr*fl"r^g i1'!"* Cuto C^vtfp S
CJ*k>8 hL-v*mnB Plm* cn ',<ll1*ri<r}tr ivi
CIV- IB CN
S'lo'n I! r.i..[-T
l kiiriLMtanL=mcuuauiui^(>vtM'l~ W
^ fl^lrj PKTT Tr^JrxrrJtiOfi I <S , Ker.a . i'
_iw j*. -. D^cJ ciJtd Piî T'rtilopTnidftn rS2
Crfhr.a =:*.!. Iijirfo.rnfcWf. CSS r Won- > fc.
? J*e :-".'.* P-j^ CaHntjfartT TrwiirDrnaiMn CSS
If theCiscoUnified Border Element doesH.323-to-SlP interworking for Cisco Unified
Communications Manager, most SIP proxy servers require the SIP call tobeEarly Offer. This
implies that the H.323 side must be H.323 FastStart. Hence, Cisco Unified Communications
Manager must be configured for inbound and outbound H.323 FastStart, which also requires an
MTP. To configure the MTP, attach aMedia Resource Group List tothe gateway or trunk as
seen previously.
Troubleshooting Cisco Unified Border ElementSIP Side
This section focuses onthe SIP-side troubleshooting.
LoopbackO: 10.1.250.101 $
voice service voip
aip
bind control source-interface LoopbackO
interface LoopbackO
ip address 10.1.350.101 255.255.255.255
session protocol sipv2
session target ipvl:10.2.1.1
incoming called-number 5112...
dtm- relay rtp-nte
codec g711ulaw
lp-i
lio.iliic. ho )itut\ hi,. ij l',>i.Ht'd :joi<L *
fclemsntSIP Side
Cisco Unified Cisco Unified
BorderElement Communications Manager
i 10.2.1.1
This figure showsthe basicconfiguration of Cisco Unified BorderlUement for the SIPside that
connects totheright Cisco Unitied Communications Manager.
3-102 Troubleshooting Cisco Unified Communications(TVOICE) vS.i
ElementSiP Side (Cont.)
*IP lHluUhll*h
i^">sn a*H | L0 ! ^gipr |
"iff"CP"Cf : w-ftg utrrh soae* - urn* > ^
Cisco Unified Communications Manager is required to define a unified communications SIP
trunk to the Cisco Unified Border Element.
The following MTP requirements apply to SIP trunks:
SIP trunk without an MTP: Configure a SIP trunk without MTP if delayed offer or invite
with no SDP is acceptable for trunk outbound calls.
SIP trunk with MTP: Configure a unified communication SIP trunk (with MTP) if early
offer or invite with SDP is a requirement (G.711 calls only) for trunk outbound calls.
SIP trunk inbound calls can be either early offer, or delay offer; both are enabled on SIP trunks
by default.
The figure shows the configuration for a Cisco Unified Border Element that is defined with a
SIP trunk to Cisco Unified Communications Manager.
2010 Cisco Systems, Inc Troubleshooting Cat! Setup Issues 3-103
Cisco Unified Border Calf Flow-H.323 Initiated
This figure shows a call flow for the call setup via a Cisco Unified Border Element that
originates at an 11.323 side.
Cisco Unified B
H323
.VNJ..
$
H 323 SETUP
Fast-Start
Nolify
H 323 CALL
PROC
H323
ALERTING
H 323 CONNECT
Fasl-Start
FacilHy
^1
. ^
-.;-! '
4-p
_.
SIP INVITE
Early Offer (SDP)
....
*
SIP 100
TRYING
~
....
SIP 180
RINGING

-
SIP 200 OK
(SDP)
SIP ACK
The call setup performs H.323 FastStart to SIP early offer translation. The H.323 SETUP,
which includes the fast-start information element, is received at the Cisco Unitied Border
Element. SIP INVITE with early offer SDP parameters is originated. The Cisco Unified Border
Element sends the H.323 CALL PROCEEDING messages immediately.
When SIP 100 Trying and SIP I 80 Ringing are received, the Cisco Unified Border Element
originates an 11.323 ALERTING message to indicate that the called phone has been ringing. At
this stage, the calling IP phone gets the ringbaek tone.
As soon as the SiP 2(10 OK message brings the remote IP phone media parameters in its SDP
body for the SIP side, the Cisco Unified Border Element originates 11.323CONNECT with the
selected codec and RTP port parameters of the H.323 side.
At the end. Cisco Unified Border Element responds with the SIP ACK to finali/e the call setup.
3-104 Troubleshooting Cisco Unified Communications (TVOICE) v8 0 CqS?010 Cisco Systems. Inc
Cisco Unified Border Call Flow-SIP Initiated
This figure shows a call flow for the call setup via a Cisco Unified Border Element that
originates at a SIP side. *
Cisco Unified Border
H.323
$
H.323 SETOP
H.323 CALL
PROC
H.323
ALERTING
H 323 CONNECT
SIP 100
TRYING
SIP 180
RINGING
SIP 200 OK
(SDP)
SIP ACK
(SDP)
SIP re-INVITE
(SDP)
SIP 100
TRYING
SIP 200 OK (SDP)
and SIP ACK
This call setup originates as SIP delay offer (INVITEdoes not contain media information) and
on the H.323 side, it must be mapped to H.323 slow-start.
SIP INVITE without SDP parameters is received at Cisco Unified Border Element. The Cisco
Unified Border Element sends the H.323 SETUP message and SIP 100 Trying message
immediately.
When H.323 CALL PROCEEDING and 11.323 ALERTING messages are received, the Cisco
Unified Border Element originates a SIP 180 Ringing message to indicate that the called phone
has been ringing.
When the call is answered, Cisco Unified Border Element receives the H.323 CONNECT
message that is followed by H.245 media negotiation. Media parameters are also negotiated at
the SIP side. Cisco Unified Border Element sends a SIP 200 OK message that includes an SDP
body. As a response to this message, Cisco Unified Border Element receives the SIP ACK with
the SIP side (IP phone) media information in its SDP message body.
Now Cisco Unified Border Element sends SDP parameters again (selecting the media from SIP
ACK) by using the SIP re-INVITE message. This way, a most preferred codec is selected from
the list of codecs received.
Cisco Unified Border Element receives the SIP 100 Trying message, and then the media
parameters are confirmed by the receipt of the SIP 200 OK with final SDP parameters.
At the end. Cisco Unified Border Element responds with the SIP ACK to finalize the call setup.
2010 Cisco Systems, Inc. Troubleshooting Call Setup Issues 3-105
This section describes a call-setupdebug output on a Cisco Unified Border Element,
CUBE* ;
CUBE#debug h225 agn
alp:5112001*10.1.350.101;5060 SIP/2.0
Via; SIP/2, 0/TCP 10.2.1.1:50G0;branch=z9hs4bK14eB3bae4
':,,', I
Date; Tue, 30 Mar 2010 12;40;00 GMT
Call-ID; 59aa9S80-bbi1(120-1 - 101020aS10.2 .1.1
... truncated ...
/SlP/Into/sipSPlChecklpip; ' i ,'..- ,< ->= . . '
.., truncated ...
/SiP/info/sipSPIGe tCallcronf ig: ' ' . ; >, '-.'.:
... truncated ...
/ SIP/Inf o/sipS PICetCon tent SDP : " ; .;;..,... i. ..>.
/SIP/Info/BipSPIContinueNewHaglnvite : ccsip api call setup ind retu rned;
. .. truncated . . .
This figure and the figures that followshow the output of the debug ccsip all and debug h225
asn commands at the Cisco Unified Border Element, This output is for a call that originated at
the SIP-side Cisco UnifiedCommunications Manager calling the ll.323-side Cisco Unitied
Communications Manager using SIP delay offer mode.
Note The debug cch323 all command would show similar output for the H.323 side.
Caution In production environments, it is recommended that you use the command debug ccsip
messages instead of the command that is shown in the figure, because high CPU usage
could result
The output startswithreceiving SIPINVITE, followed by the call matching to the outbound
dial peer. The media flowuses the default mode flow-through. Because the SIP INVITEdoes
not contain any SDP information, this call uses the delay offer mode.
3-106 Troubleshooting Cisco Unified Communications (TVOICE) v8.0 2010Cisco Systems. Inc
Troubleshooting Cisco Unifi
Element (Cont.)
Sent) value H323 Userloformation
h323-ftesaagft-body setup
BOurceCallSiginlAddxeBS ipAddr
ip -OkOmeS'H
port 32760 I . . . truncated
(Received) value B323 UserlnforBiatioo
h323-message-body callProceeding :
. . . truncated . . .
(Received) value H32J Userinfornation ::
h323-uu-pdu
h323-message-body alerting
The Cisco Unified Border Element sends the 11.323 SETUP message without media
description. The H.323 CALL PROCEEDING is receivedat the CiscoUnifiedBorderElement
that is followed by the receipt of H.323 ALERTING.
) 2010 Cisco Systems, Inc. Troubleshooting Call Setup Issues 3-107
Jnihc u I luMi-i
ioment (Com
via; SIP/2.Q/TCP 10.2.1.1;5060;branch=z9HG4bKH863bae4
From: tsip:2001#10.;.l.l>; tag= 72 b0 6df - 6cd3 -4bB 9 92e 1 -defl 4750 Ob57d -
2SB2S756
To: <sip!S112001il0.1.250.101>
/SIP/Event/sipSPIEventlnfo: Queued event from SIP SPI :
SIPSPI EV CC CALL_PP,OCEEDINS
/SI P/Inf o/cc sip gw set sipspi mode: --- .. -,i >;.,;;;. - ., ;..-,;
... truncated ...
/SIP/Event/BipSPIBventinfo: Queued event from SIP SPI =
SIPSPI EV CC ..
/SIP/Event/aipSPIEventliif 0: Queued event from SIP SPI :
SIPSPI EV CC - .
... truncated ...
Vis: SIP/2.Q/TCP 10,2 .1.1:E060;branch.i9hG4bKb3dBBebOCI
From; tsip:2001810.2.1.1>;tag=72fbO6df-6cd3-4b89 -92e1-deB47500b57d-
21S65S24
To; <rsip;511200ie 10.1.250.101> I tag=.A3 B2 6C- 23AE
TheCisco Unified Border Element sendsthe SIPTryingand marksthe call as SIP-to-H.323.
As the call setuphas entered the alertingphase(11.323 ALERTING was received earlier), the
CiscoUnified Border Element sendsthe SIP Ringing to the SIP-connected Cisco Unified
Element (Cont.)
(Peceived) value H323 nsr Information -.:-
{
h323-mesaage-body connect ;
{
hUSAddreus ipAddress :
{
ip 'OAOIOIOI'H
port 36773
J ... trunca ted ...
Via- SIP/2.0/TCP 10.2.1.1:5060;branch-9hG4bKb3dB8eb00
FroiD: <sipi20 0110.2.1.1>. tag.72fb06df-6cd3-4b8 9-92el-d8B47500b57d-
21965524
To; saip;S1120Dl10.1.250.101>itg-A3826C-23AE
.. . truncated ...
c*in iP4 -' :; ..:;;
t=Q 0
n-audio :" :: rtp/avp o is ioi
c.ih IP4 _: ij "- ;' '.
a=.inactive
a-rtpmap:0 ?."::". Z-'"'.
a*rtpmap,19 CH/B0O0
When the call is answered, the Cisco Unified Border Element receives the H.323 CONNECT
message from the H.323 side. This message isfollowed by the H.245 phase toexchange the
media information at the H.323 side.
The Cisco Unified Border Element sends the SIP 200 OK with the media information17320
RTP port numberto the SIP-connected Cisco Unified Communications Manager. The SIP
200OKmessage alsocontains the media IPaddress for the SIPcall leg 10.12.1.101, which is
the outgoinginterface to reachthe SIP-connected CiscoUnifiedCommunications Manager.
Note If youreview the CiscoUnified Border Element configuration, the loopback 0 interface is
used only for the SIP signaling traffic, not the media.
Troublestiooting Call Setup Issues 3-109
Troubleshooting Cisco UnitWt, U id;
sip:511200 110.1.2 50.101:5060;transportstop SIP/2.0
Via: SIP/2.0/TCP 10.3.1.1.5060ibranch=i9hG4bKc738408f8
From: <sip:2 00 1eiO,2.1.1>itag=72fb06d-6clJ3-4ba9-92el-de84750DbS7d-
21965524
To: <sip:5112001*10.1.250.101>;tag=A3S26C-23AE
v = 0
0 =CiB0oSystem9CCM-s;P 2000 1 IN I PI 10.2.1.1
s = SIP Call
c = IN 1P4
t*0 0
nwaudio . " /AVP 0 101
a=rtp=nap:0
a=ptime:20
a=rtpmap:101 telephone-event/80 00
a=fmtc:101 0-15
... truncated ...
/SIP/Info/sipSPIDoAudioHegotiation: ; - -..-. ;.;:,-.. ., , ,1,,;., ,,,L
' < line 1 payload type=0, codec bytea=160, ;!., ';~]1;,.,.-,
dtmf relay=rtp-nte stream type=voicE+dtmf (II.
fhis figure shows the SIP ACK that is received from the SIP-connected Cisco Unified
Communications Manager. It contains thefollowing:
The IP address of the IPphone that is registered to the SIP-connectedCisco Unified
Communications Manager is 10.2.2.12.
RI P port number at the IP phone is 4000.
G.71 1 is the mu-lawcodec for the IP phone.
Thebottom of theoutput summaries themedia information that is negotiated so far at theSIP
side.
Element (Cont.)
-.--. sip: ZOOM10. 2. 1.1: 5060 j transport-tcp SIP/2 .0
La: Sip/2.0/TCP 10.1.250.101:5060;branch.i9hG4bK4761
smote-Party-ID:
9 ip:2001*10.1.250.101 >iparty.ca11ingiscreen.yaa[privacy.off
ron: <Bip=S112001*10,1.2 50.101>jtag.7F230C-AF7
a; ssip:20 01*10,2.1.1>ftag-72fbO6df-6cd3-4b89-92el-de847500b57d-29828758
.. truncated ...
-0
.CiflcoSyatemsSIP-Qw-nserAgent 1755 4475 IK IP4 10.1.250.101
.SIP Call
.IN IP4 . : .. . . . 1
-0 0
audio >"" r;ir/*VP 0 101
. IN IP4 ". ;,.".:.
= ctpmap;0 PCHU/8000
= rtpmap:101 telephooe-event/BOOO
=mtp:101 0-16
ptime:20
. . truncatad .. .
Now the Cisco Unified Border Element originates the re-INVITEmessage with the SDP body
as required by thedelayoffer mechanisms. Notethat the mediaparameters have not changed:
The IP address of the Cisco Unified Border Element outgoing interface is 10.12.1.101.
The Cisco Unified Border Element RTP port number is 17320.
G.711 mu-law codec is chosen.
2010 Cisco Systems. Inc. Troubleshooting Call Selup Issues
/SIP/Call/sipSPICalllnfo:
The Call Setup Informatlo iS;
Call Control Block (CCB) 0X49475D68
State of The Call STATE ACTIVE
TCP Socketa Used YES
Source IP Address (Sig 1 10.1.250.101
Destn SIP Beg Addr:Port 10,2 ,1.1 ;S060
Destn SIP Reap Addr:Port 10.2 . 1,1 :36B60
Destination Nape 10.2.1.1
Number ot Media Streams:
Media Stream
Negotiated Codec Bytes 160
Nego, Codec payload 0 Itx) . 0 (rx)
Negotiated Dtmf-relay 6
Dtmf-relay Payload 101 (tx), 101 [rx)
Hiedebug output concludes the call setup with the call informationat the SIP call leg.
The pon 4000 is an intermediateport number at the IP phone that is later replaced by the real
port.
3-112 Troubleshooting Cisco Unified Communications (TVOICEl v8 0
i (Cont.)
CUBEishow voip rtp connectiooa
VoIP BTP active connections :
10.2.2.12 |S(P
H.323
1 21 22 17320 24596 10.12.1.101
]2 22 21 36773 24614 10.1.250.101 10.1.1.1
Found 2 active rtp compactions
When the call has been set up, the command show voip rtp connections displays both call
legs:
The first call leg is the SIP call leg.
The second call leg is the H.323 call leg. This leg shows the IP address of the H.323-
connected Cisco Unified Communications Manager as the remote IP address.
As an alternative, the show call active voice brief command provides similar information about
the call legs.
Immediate Remote-Call Drops
"fhis topic explains how to recognize codec-related issues and how to avoid codec mismatches
that can cause remote calls to drop immediately when answered.
3^1 JjaIîi.sjiW.kt3& -< r &ig*.dt*&&
Immeniatft Remote-
Sel 10 Use
G729
IP phone A
SCCP -*$y/
IP plone B
-x-
WAN
The SIP phones in this example
are third-parly phones thai
only support G 711 and G 723
Set to use
G723
SIP IP Phone C
SIP IP Phone D
SIP H323 or Intercluster Trunk V}"W ^
If a call to a phone at the remotesite immediately dropsafter the remoteuser picksup the
handset, it is not a dial plan problem.
You can easilv tell if the problem is in the dial plan or in another part of the Cisco Unified
Communications Manager configuration that is based on whether the phone rings. If the phone
at the tar end rings, the dial plan and call setup arc working properly.
The most common problemwith a call that sets up and rings but then drops right away is a
coder-decoder (codec) mismatch. This mismatch occurs when the Cisco Unitied
Communications Vlanagerserver instructs one of the phones to send or receive a codec that it
cannot manage. If no transcodc resources are available, the call will set up but then drop during
the media setup phase.
You must provide hardware transcoding resources at the tail end or change the region
informationto a codec that the remote phone can manage.
The example that is presented here shows (j.729 going to G.723, You can resolve this issue by
changing both phones to a common codec.
Note Transcoding resources cannot convert between G.729 and G.723. Transcoding resources
can convert only between G 711 and G.729 or G.723 and vice versa.
3-114 Troubleshooting Cisco Unified Communications (TVOICE) vB0 2010 Cisco Systems, Inc
*
Summary
Summary
Issues with call setup to another cluster can result in calls
failing, calls using an alternate path like the PSTN, and
remote-end hopoff issues. Some unique issues that can
occur to calls between Cisco Unified Communications
Manager clusters include overlapping dial plans, calls routing
to the PSTN instead of going through the WAN link first, and
calls to remote sites dropping immediately after call setup.
Overlapping dial plans can cause a postdial delay that can
make a user assume that the call has failed. This delay is
based on a system setting for the interdigit timeout.
Summary (Cont.)
Ifcalls between clusters are not setting up correctly, many of
the possible causes relate to permissions, digit manipulation,
orCAC.
New issues arise when a gatekeeper is used for intercluster
calling: registration, incorrect call routing, and call admission
issues.
Cisco Unified Border Bernent can be used for intercluster
calling, especially when CCD is used. Troubleshooting the
issues at the Cisco Unified Border Elements are similar to
H.323 and SIP gateways on the \folP side.
Call s d rop after the call is answered usual ly because of a
codec mismatch.
In this lesson, you have learned to explain the common calling issues that can occur in a
multisite Cisco Unified Communications Manager deployment and identify the most likely
causes of these issues.
References
Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide
Release H.Otll. February 2010,
Imp: www .Cisco.com en I'S doo \oicc ip csinun cucni'admin.'X I) I
ccmct'ii hccm-S01-cm.html
Cisco Systems. Inc. Cisco Unified Communications Manager Features and Services Guide.
Release 8.0(1 l, April 2009 and updated April 2010.
Imp: wuw.cNco.com en I S docvvoice_ip_coii!m cucin.;'adiT)in'8_0_l cemfcat
tûd-Sill-cni.htinl
Cisco Systems. Inc. Cisco Unified Border Element Configuration Guide, Release 15.1.
March 2010.
Imp: www.cisco.com en US docs ms \oicc;c!ibe.contiguralion.guide'l5 I'
\b I? i Book.html
Cisco Systems, Inc. Troubleshooting Gatekeeper Endpoint Call Admission Issues, February
2006,
http- uww.ciscu.com en US tech tkK)"?" technologic* lech nolcO1)! S6;.0OK0Ou654c shtml
3-116 Troubleshooting Cisco Unified Communications (TVOICE) v8 0 <> 2010 Cisco Sybils. Inc
Lesson 4
Troubleshooting Off-Net
Calling Issues
mm Overview
A call that is placed off site must go through a gateway. This gateway could be a Cisco Media
Gateway Control Protocol (MGCP), H.323, or Session Initiation Protocol (SIP) gateway. This
HV lesson discusses how to troubleshoot these different types of gateways and provides an
understanding of the most common issues that are associated with placing a call off site.
During call processing, Cisco IOSvoice gateways analyze digits and can performdigit
^m manipulation. Understanding these functions andhow a CiscoIOSvoice gateway performs
these tasks differently fromCisco Unified Communications Manager can help troubleshoot
common Cisco IOS voice gateway issues.
Cisco Unified Communications Manager provides the dial plan for most devices in a Cisco
Unified Communications system. This dial plan is often the most complex troubleshooting area
of placing offsite calls in your Cisco Unified Communications system. This lesson reviews dial
plans, including globalized call routing, and addresses the most common issues that can occur
with these dial plans.
Other common voice issues that this lesson covers include troubleshooting missing or incorrect
caller ID, ringbaek issues, one-way audio, dead air, dropped calls, and public switched
telephone network (PSTN) setup failure.
Objectives
Upon completing this lesson, you will be able to explain the common catling issues that can
occur with off-net calls and identify the most likely causes of these issues. This ability includes
being able to meet these objectives:
Describe some of the most common sources of issues when placing calls off-net
Describe the troubleshooting procedure that you can use to troubleshoot MGCP, H.323,
and SIP gateways
Explain how a Cisco IOS voice gateway performs digit analysis and how the gateway
performs inbound and outbound dial-peer matching
Explain how incorrect DDIs can cause calls to the PSTN to fail because of incorrect or
incomplete digits
1*
Explain theprocedure to ensure thatCiscoUnitied Communications Manager andthe
gateway arecorrectly set up with respect toroute patterns, route groups, route lists, and
CoS
Describe various scenarios and common voice issues that can occur in a Cisco Unified
Describe the issuesof globalized call routingand plus (+) dialingand their related digit
manipulation
3-118 Troubleshooting Cisco UnifiedCommunications (TVOICE] v8.0 2010 Cisco Systems. Inc
mm
Common Off-Net Calling Issues
This topic describes some of the most common sources of issues when placing calls off-net,
including PSTN access.
Overview of Common Off-Site Caltim
Manager Issues:
Gateway configuration errors
Dial plan configuration errors.
CoS and digitmanipulation problems
Routeplan configuration errors
Codec issues withthe region settings
Problems with the location settings
Gateway Issues:
CiscoIOSgatewayconfiguration errors
Digit manipulation and COR errors
Problems with QoS settings on
WAN links
All trunks to PSTN used, PSTN
issues
Invalid number dialed
Cisco Unified Communications Manager Issues
These are common reasons for offsite calling issues that involve Cisco Unified
Communications Manager:
MGCP gateway configuration errors
11.323 gateway configuration errors
SIP trunk errors
Dial planconfiguration errors:
Problems with class of service (CoS)
Route plan configurationerrors
Digit manipulation
Coder-decoder (codec) issues with theregion settings
Problems with location settings for Call Admission Control (CAC) issues
Troubleshooting Call SetupIssues 3-119
Gateway Issues
These arc common reasons for offsite calling issues that involve gateways:
Cisco IOS MGCP gateway configuration errors
Cisco IOS 11.323 and SIP dial-peer configuration errors
Digit matching
Digit manipulation and class ofrestriction (COR)
Session target issues
Registration. Admission, and Status (RAS) issues (11,323 only)
Problems with quality ofservice (QoS) settings on WAN links
All trunks to PSTN are used
Invalid number
Carrier problems
2010 Cisco Systems,
Gateway Troubleshooting
Thistopic contains checklists thatyoucanusetotroubleshoot MGCP, 11.323, andSIP
gateways.
MGCP Gateway Troubleshooting Procedure
Verify the following configuration on the Cisco IOS
MGCP gateway:
Verifythe mgcp commands.
Verifythe ccm-manager commands.
Ifusing a PRI circuit, ensure that Q.931 backhauling is
configured.
Verify this gateway configuration on Cisco Unified
Ensure that the hostname of the gateway is entered for a
Cisco !OS MGCP gateway.
Check the endpoint configuration.
To troubleshoot call issues with a Cisco IOS MGCP gateway, follow these steps:
Step 1 Verify these settings on the Cisco IOS MGCP gateway:
The mgcp commands are presentand have beenenteredcorrectly.
The ccm-manager commands are presentand have beenenteredcorrectly.
If usinga PRI circuit,ensurethat Q.931 backhauling is configured. This allows
you to use the Q.931 Translatorto viewthe ISDNLayer 3 activity on thedata
channel (D channel) from Cisco Unified Communications Manager.
Step 2 Verify thesegatewaysettings on the Cisco Unified Communications Manager
server:
The hostname of the gateway is entered for a Cisco IOS MGCP gateway. The
hostname is case-sensitive and must match exactly what is configured on the
gateway.
Check the endpoint configurationon the Cisco Unified Communications
Manager. In the Cisco Unified Communications Manager administrative web
interface, choose Device > Gateway > See Endpoints.
>2010 Cisco Systems. Inc Troubleshooting Call Setup Issues 3-121
alHU!* ! fcslMMi*s,>..,?>
Verify registration of the gateway with Cisco Unified
Communications Manager by using these
commands:
* show ccm-manager
' show mgcp endpoint
show mgcp statistic
Verify if call setup functions correctly by using these
commands:
" show mgcp connection
' debug mgcp error
debug mgcp packet
* debug mgcp state
Step 3 Use these commands to verify that the gateway has registeredwith Cisco Unified
show ccm-manager: Verities that the gateway is registered and pointing to the
correct Cisco Unified Communications Manager server
shot* mgcp endpoint: Verifies thestatus of theendpoints thatareconfigured
show mgcp statistic: Shows statistical information for the MGCP activity
Step4 Use thesecommands to verifythat call setupfunctions correctly:
show mgcp connection: Displays any active MGCP-eontrolled connections
debug mgcp error: Shows MGCP error messages
debug mgcp packet: Displays MGCP packets
debug mgcp state: Displays MGCP state changes
The output that follows is anexample from theshowccm-manager command. Notice in the
output that the registration iscurrently tothe primary Cisco Unified Communications Manager
server at 10.1.1.2. Also, a backup Cisco UnifiedCommunications Manager server with an IP
address of 10.1. I.I is ready. The last failover time determines when the last reboot or failover
occurred. Notice that the backhauling of Q.931 signaling is currently open to the 10.1.1.2Cisco
Unified Communications Manager server. Whenyou arc troubleshooting the automatic
download of configuration from the CiscoUnifiedCommunications Manager, the sectionof
output that manages configuration download attempts can helpyoudetermine if download
errors are occurring.
nQ-litshow ccm-manager
MGCP Domain Name: :1Q-I
Priority S'atjs Host
Primary Registered 10.1.1.2
First Backup Backup Ready 10.1.1.1
3-122 Troubleshooting Cisco Unified Communications (TVOICE) v8 G
Second Backup None
Current active Call Manager: 10.1.1.2
2428
3 0 seconds
15 seconds
20:33:03 PST Dec 12 2006
20:33:03 PST Dec 12 2006
(elapsed time:
(elapsed time:
Backhaul/Redundant link port:
Failover Interval:
Keepalive Interval:
00 :00:Oil
00:00:01)
Last failover time:
Switchback mode:
MGCP Fallback mode:
Last MGCP Fallback start time:
20:09:03 PST. Dec^^.aG'QS';from (10.1.1.2)
20:09:48 PST Dec 12 2006 from (10.1.1.1)
Graceful
Not Selected
None
None
Disabled
Backhaul Link info:
Link Protocol:
Remote Port Number:
Remote IP Address;
Current Link State:
Statistics:
Packets recvd:
Recv failures:
Packets xmitted:
Xmit failures:
PRI Ports being backhauled:
Slot 0, port 0
Configuration Auto-Download Information
Current vers ion-id: 1173329637-747a4d52-62ee-45aa-ab7d-a3 0c7493dc0d
Last config-downloaded:00:00:00
Current state: Waiting for commands
Configuration Download statistics:
Download Attempted : 3
Download Successful : 3
Download Failed. : 0
Configuration Attempted : 1
Configuration Succeasul : 1
Configuration FailedfParaing): 0
Configuration Failed(config) ; 0
Last config download command: New Registration
FAX mode: ciscc
Configuration Error History:
TCP
2428
10-1.1.2
OPEN
30
0
30
0
J2010 Cisco Systems, Inc Troubleshooting Call Setup Issues 3-123
H.323 Gateway Troubleshooting Procedure
This section describes the procedure to troubleshoot an H.323 gateway.
Verify the following configuration on the Cisco IOS H.323
gateway:
' Verify that the dial-peer commands are correct.
Venfy thai the voice class H.225 timeout forTCP is set to 3 seconds.
Ensure the use of the preference command to determine Cisco Unified
Communications Manager server order.
' Check that the H.323 binding is configured.
Verify this gateway configuration on Cisco Unified
Venfy the settings of the gateway, including the IP address for a Cisco
IOS H.323 gateway.
Check the route pattern, route lists, and route groups in Cisco Unified
Communicalions Manager.
To troubleshoot call issues with a Cisco IOS11.323 gateway, followthese steps:
Step 1 Verify these settings on the Cisco IOS 11.323 gateway:
Verity that the commands under the dial peer arc correct for Cisco Unified
Communications Managerredundancy. There shouldbe one dial peer for each
Cisco Unified Communications Manager server towhich thegateway could
register.
Verify that the voice class 11.225 timeout for TCP is set to 3 seconds or less. The
default setting is longer than the Q.931 call proceeding timer of 10seconds. As a
result, if this setting is not changed and the first-choice Cisco Unified
Communications Manager is not available, the call will timeout before the
second-choice Cisco Unified Communications Manager is attempted.
Verify that the preference command has been used to determine the Cisco
Unified Communications Manager server order. The lower the number, the more
preferred the server is.
Check that the 11.323 binding is configured. The 11.323 binding ensures that the
source IP address that an 11.323 packet arrives with on the Cisco Unified
Communications Manager is predictable and consistent. This configuration
should be considered mandatory.
Step2 Verify these gateway settings on the CiscoUnified Communications Manager
server:
Verify that the settings of the gateway are correct. Pay particular attention to the
IP address to ensure that it matches the gateway address that is used for 11.323.
To define the address on the gateway, use the h323-galtway voip bind srcaddr
ip addr command.
3-124 Troubleshooting Cisco UnifiedCommunications (TVOICE) v8 0 2010 Cisco Systems. Inc
mm
Check that the route pattern has the correct partition that is applied and that the
pattern that is defined is correct.
Ensure that the route lists are correct and that the digit manipulation, if any, is
proper.
Check the route groups and verify that the gateways are listed in the route group
in the correct order.
Verifydigits that are sent and received on a POTS call leg
by using these commands:
* show dialplan number number
* debug vtsp session
* debug vtsp dsp
* debug voice dial
' show isdn status
debug isdn q931
Verifythat call setup on a VoIP call leg functions by using
these commands:
* debug voice ccapi inout
debug voip dialpeer inout
* debug cch323 h22S
" debug h225 q931
Step3 Use these comtnands to verifythe digitsthat arc sent and received on a plainold
telephone sen ice (POTS) call leg:
show dialplan number number
debug vtsp session
debug \pm signal
debug wiice dial
show isdn status
debug isdn q931
Step4 Usethesecommands to verifythat call setupon a VoIPcall legfunctions correctly
debug voice ccapi inout
debug \oip dialpeer inout
debug cch323 h225
debug h225q93l
3-126 Troubleshooting Cisco Unified Communications (TVOICE) v8 i 12010 Cisco Systems. Inc
SIPGateway Troubleshooting Procedure
This section describes the procedure to troubleshoot a SIP gateway.
S!P Gateway Troubleshooting Pi
Verify the following configuration on the SIP gateway:
VoIP dial peers are configured and have SIPv2 enabled as well as a valid session
target.
SIP UAretry settings are verified.
Proper DTMF relay is seleded.
Verify the following gateway configuration in Cisco Unified
SIP trunk is configured and points to gateway IP address.
Verify call setup withthe SIP gateway by using these commands:
> show sip-ua status
show sip status
show sip statistics
debug ccsip calls
debug ccsip error
' debug ccsip event
debug ccsip error
debug ccsip message
To troubleshoot call issues with a Cisco IOS SIP gateway, follow these steps:
Step 1 Verify these settings on the Cisco IOS SIP gateway:
Verify thattheVoIP dial peeris configured (or dial peers areconfigured) and
have SIP version 2 (SIPv2) enabled and a valid session target.
Verifythe SIPuser agent (UA) retrysettings configuration.
Ensurethat the properdual tone multi frequency (DTMF) relay is selected.
Step2 Verify that a SIP trunk is configured inCisco Unified Communications Manager
and that an IP address is used as the destination address.
Step 3 Use these commands to verify call setup with the SIP gateway:
show sip-ua status
show sip status
show sip statistics
debug ccsip message
Gateway Digit Collection and Analysis
This topic describes how a Cisco IOS voice gateway performs digit analysis and how the
gateway performs inbound andoutbound dial-peer matching.
's*!wi*&fc~*ni^^ fcW
.,':;iOWay-
Dialing
555-0040
^
Analog
DTMF, Digit by Digit
ISDN En-Bloc
lllflfl
555-0040
ISDN Overlap Receive
TinmnnnFTi,
ISDNSETUP Message
ISDN INFORMATIONMessage
Ifagateway isconnected to PSTN orlegacy telephony equipment by using an analog line, like
Foreign Exchange Office (FXO), for instance, digit collection iscarried out using DTMF
signaling as soon as supervisory signaling has established a voice path. In some parts of the
world, pulse signaling may still be used, but this older mechanism isgradually disappearing.
Digit collection onanalog lines isdigit by digit; a gateway interprets every collected digit and
attempts to analyze the collected dialing pattern or substring.
With digital PSTN connectivity, message-oriented digitcollection takes place. Thecalled
number canbeeithercollected en-bloc, using the ISDN SETUP message, or when theentire
number string is sent.
Incountries whose national numbering plan is noteasily defined with static route patterns, a
Cisco IOS gateway can beconfigured for overlap sending and overlap receiving.
The first SETUP message includes only asubstring ofacalled number; for example asingle
leading digit. 1he SETUP message also includes a user information element (canoverlapsend
tlag set totrue) toindicate tothe other endthat overlap sending is used. The other side responds
with a SETUP ACKonly and waits for more digits to be sent.
The gateway is then sending each additionally received digit ina separate INFORMATION
message. When the other side receives enoughdigits and is able to routethe call, it finishes the
connection handshake as usual (for example, witha sequence of PROCEED, PROGRESS,
ALERTING. NOTIFY, and CONNECT messages). Notethat the exact handshake candiffer
and depends on the ISDNswitch type (if used with MGCP-controlled ISDN interfaces).
Digit Coliection and Dial-Peer Matching Review
This section reviews how Cisco IOS gatevteys collect digits and match them with adial peer.
Digit Col
Review
Gateway collects digits until itcanmatch anoutbound dial peer.
After a match is made, the router immediately placesthe call.
Nofurther digits are collected.
Digits 5550040 Dialed
destination-pattern 555
port 0/1
session target ipv4:10.1B.O 2
r ; ;: matches Only the
cdlected digits of 555 will be
forwarded.
Digits555D040 Dialed
dial-peer voice 1 pots
destination-pattern 555....
port 0/1
djal-paet voice ~ voip
session target lpv4;10.1B.0.2
D'al-pesr Zmatches. The
collected digits of 5550040 will
be forwarded.
The call-routing logic on Cisco IOS routers that use the H.323 and SIP protocols relies on the
dial-peer construct. Dial peers are like static routes; they define where calls originate and
terminate and what path the calls take through the network. Dial peers are used to identify call
source and destination endpoints and to define the characteristics that are applied to each call
leg in the call connection. Attributes within the dial peer determine which dialed digits the
router collects and forwards to telephony devices.
One of the keys to understanding call routing with dial peers is the concept ofincoming versus
outgoing call legs and, consequently, of incoming versus outgoing dial peers. Each call that
passes through the Cisco IOS router is considered to have two call legs, one entering the router
and one exiting the router. The call leg that is entering the router is the incoming call leg, while
the call leg that is exiting the router isthe outgoing call leg.
There are two main types ofdial peers, depending on the type ofcall leg with which they are
associated:
POTS dial peers that are associated with traditional time-division multiplexing (TDM)
telephony call legs
VoIP dial peers that arcassociated with IPcall legs
When matching the destination pattern, the Cisco IOS gateway performs aleft-aligned match -
that is. the pattern ismatched with the beginning ofthe received string.
Gateway Dial-Peer Matching Review
This section describes how aCisco IOS gatewav matches adial peer
3-130
Cisco Unififrt _ ,
Comrrunicalrons Manager *=!
Express-SRST IP Phone
Cisco United
CommunicationsManager
For agateway in cither SRST or Cisco Unified Communications Manager Express mode or
only agaieway extension from the Cisco Unified Communications Manager, you can select
many possible paths,
Via dial-peer VoIP, you can connect Cisco Unified Communications Manager other Cisco
Unified Communications Manager Express routers, gateways, gatekeeper, or Cisco Unified
Border Element.
For dial-peer POTS, the paths can lead to analog devices, IP phones, PBXs, or the PSTN.
Depending on the design, the path will vary; for example, aCisco Unified Communications
Manager Express primarily has IP phones, (axes, and the PSTN that are connected.
The design and configuration of path selection is now acombination of the digit analysis digit
matching, selecting the relevant path, and digit manipulation. You can enhance that
combination by assigning calling privileges, which are covered in the next lesson.
Routers must match the correct inbound and outbound dial peers to complete acall
successfully. For all calls going through the router, Cisco IOS Software associates one dial peer
to each call leg. The figure illustrates possible types ofcalls that can be identified:
Call 1is from another 11.323 gateway that is across the IP network to atraditional PBX that
is connected to the router (for example, via aPRI interface). For this call, an incoming
VoIP dial peer and an outgoing POTS dial peer are selected.
Call 2is from an analog phone that is connected to aForeign Exchange Station (FXS) port
on the router to aCisco Unitied Communications Manager cluster that is across the IP
network. For this call, an incoming POTS dial peer and an outgoing VoIP dial peer are
selected by the router.
Call 3is from an IP phone that is controlled by Cisco Unified Communications Manager
Express or SRST to aPSTN interface on the router (for example, aPRI interface). For this
call, an automatically generated POTS dial peer (corresponding to the ephone that is
configured on the router) and an outgoing POTS dial peer are selected.
b$w&8&ÂM&&4t;
Inbound dial-peer matching:
Called number wth incoming called-number
Calling numberwith answer-address
Calling number with destination-pattern
For POTS: voice-port matches with dial-peer port
Still no match: default dial-peer0 is used
Outbound dial-peer matching:
Gateway tries tomatch the called number with destination-
pattern
If multiple matches arefound, the best (numerically lowest)
preference wins '
Ifequal preferences are found, a ra
"m =:,.-!: n
is chosen.
When aCisco IOS gateway routes acall, the inbound and outbound dial peers must be
matched, Ihe gateway will search through all dial peers and apply matching criteria When a
dial peer has been matched, the gateway selects it as the inbound or outbound dial peer.
To match incoming call legs to incoming dial peers, the router selects adial peer by matchirm
the information elements mthe setup message with four configurable dial-peer attributes The
information elements are the called number and the calling number.
Inbound Dial-Peer Matching
Calls that are coming into agateway match dial peers that are based on the source of the call If
acall enters by using avoice port, the inbound dial peer is aPOTS dial peer. If acall enters by
usmg IP, the inbound dial peer is aVoIP dial peer. Inbound dial-peer matching for digital
POIS and VoIP dial peers is prioritized in this way:
1. If the called number matches with the incoming called-number configuration on adial
peer, this dial peer is selected as the inbound dial peer. No further matching is performed,
2. Ifno dial peer has been found, the callmg number is checked. Ifthe answer-address
configuration of adial peer is matched, this dial peer is selected and no further matching is
performed,
3. If the calling number matches with the destination-pattern configuration of adial peer
this dial peer is selected and no further matching isperformed.
4. If none of the above was successful and the call is inbound on aPOTS port, adial peer with
a matching voice-port contiguration is searched.
5. If still no match isfound, the default dial- peer 0 is used.
3-132
Tip
Default dial-peer matching is undesirable because default call characteristics may not be
what you want.
Troubleshooting Cisco Unified Communications (TVOICE) vB 0
The router needs to match only one ofthese conditions. It is not necessary for all ofthe
attributes to be configured in the dial peer or that every attribute match the call setup
information The router stops searching as soon as one dial peer is matched, and the call is
routed according to the configured dial-peer attributes. Even ifthere are other dial peers that
wouldmatch, onlythe first match is used.
Note Atypical misconception about inbound dial-peer matching is that the session-target of a
dial peer is used. This is not true. Instead, use the incoming called-number or answer-
address command to ensure that the correct inbound diai peer is selected.
Outbound Dial-Peer Matching
Outbound dial-peer matching isprioritized in this way by default:
1. The gateway searches through all dial peers and tries to match the called number with the
destination-pattern configuration. The dial-peer with the closest match is selected.
2. If multiple equal matches are found, the dial peer with the lowest preference configuration
wins.
3. Ifequal preferences are found, arandom dial peer isselected.
Note You can change default outbound dial-peer matching by using the dial-peer hunt
command.
,2010 Cisco Systems. Inc Troubleshooting Call Setup Issues 3-133
Issues with Discard Digits Instruction
This topic explains how incorrect discard digits instructions (DDIs) can cause calls to the PSTN
to fail because of incorrect orincomplete digits.
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Called-partylr=in:>tomidl:[
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The figure provides areview ofall digit manipulation options that are available with Cisco
Starting with acall that comes from agateway or trunk (inbound) or from aphone, the first
digit manipulation options are found at the ingress gateway and trunks. These options can be
configured with incoming calling-party prefixes, or by stripping digits, or by performing a
transformation that is based on atransformation calling search space (CSS) to modify the
calling party. Ihe called party can by modified with significant digits, with aprefix directory
number, with the incoming called-party transformation, with adding aprefix, by stripping
digits, or by performing atransformation that is based on the transformation CSS. No digit
manipulation can be performed for calls that are placed from aphone.
Note
The external phone number mask is configurable atadirectory number of the phone,
however, the decision to apply the external phone number mask or not to apply it is made
only laterat the matched route pattern or translation pattern or at theroute list.
When the called number matches adirectory number of the phone, no further called-party
transformation is possible. This would not make any sense, because the endpoint, which should
receive the call, is already identified. The calling-party number, however, can be modified by a
calling-parry transformation CSS.
When the called number matches atranslation pattern, hunt pilot, or route pattern, you can
configure aset ofdigit manipulation methods for both the called and the calling-party numbers.
Process them in this order:
Called party transformations: Discard digit instruction, transformation mask, and prefix
digits. In addition, you can set anumbering plan and number type.
3-134 Troubleshooting Cisco Unified Communications (TVOICEl v80
Calling-party transformations: Use an external phone number mask, transformation
mask, orprefix digits. In addition, you can set anumbering plan and number type.
When the call request matches atranslation pattern, anew call request is sent with the
transformed called and calling numbers.
When the call request matches aroute pattern referring to aroute list, digit manipulation
parameters can be configured per route group at the appropriate route list. The digit
manipulation settings that can be configured are the same as those that are found at route
patterns, translation patterns, and hunt pilots.
At egress devices (gateways and trunks), global transformations can be applied by configuring
called- and calling-party transformation CSS. For the calling-party transformation, the Caller
ID and the directory number can be set inaddition (processed after the calling-party
transformation CSS). For both the called- and the calling-party number, you can select a
numbering plan and number type.
Incoming Number Transformation Considerations
This section describes some special considerations when using incoming-party digit
manipulationtechniques.
3-136
...u.-*&*q., y.^
tnni.y Uumbw lEaiibiOthtaikuii
Incoming calling-party settings allow stripping ofa certain
number ofthemost significant digits andthen add a prefix to
a catling number or attach more complextransformation
through CSS, or combine them all.
incoming called-partysettings allow the same but fora called
number(H.323 gateways and trunksonly).
Both can beset on trunks, gateways, device pool (overwrites
devicesettings)and service parameters.
Incoming calling-partysettings
Prefix: 1408, Slnp digits' 4, CSS. addOll.css
12125550001 5550001 14085550001
Transformation pattern
-ddon Pt
DDIPreDot, Prefix-0111303
01113035550001
The incoming calling-party settings allow configuring manipulation of calling number digits for
call ingress, The incoming calling-party settings can strip leading digits, or add aprefix, or
configure more complex digit manipulation via atransformation CSS. These three options can
be combined. 1he figure illustrates the sequence that forms the resulting calling number:
1. Digit stripping is first inthe sequence.
2. Adding a prefix issecond in the sequence.
3. More complex transformation via CSS is third in the sequence.
The same mechanism exists for both incoming calling-party and incoming called-party settings
that can beconfigured for 11.323 gateways and trunks.
The incoming calling and called-party settings can be configured either attrunks otat
gateways, on device pool and service parameters. Ifdevices and device pool settings are
present and the Use Device Pool CSS check box is marked at adevice configuration page, the
device pool settingstake precedence.
Troubleshooting Cisco Unified Communications (TVOICEl v80
2010 Cisco Systems
Outgoing Number Transformation Considerations
This section describes some special considerations when using certain digit manipulation
techniques for outbound number transformation.
Outgoing Number Transformation
Considerations
Impact ofcalled-party digit manipulation performed at the
route pattern:
The transformed number is displayed at the calling phone.
- The transformed number is written into CDRs.
- Thesame applies tocalled-party transformations applied
to the egress gateway or trunk.
All ofthe above does not applywhen digit manipulation is
performed at the route list.
When implementing digit manipulation for the called number at the route pattern or via global
transformations at the egress gateway, the appropriate transformation is also represented at the
phone display. The calling phone changes the called (and later the connected) number
according to the performed transformation. In addition, transformations that are performed at
the route pattern are also considered when creating Call Detail Records (CDRs). In contrast,
digit manipulation that is performed at the route list is not reflected at the phone display or in
CDRs.
Toapplya , -. .--: '.: .,... ioa trunk orgateway directly
(atthe device) orindirectly (via thedevice pool), you must consider these
issues
All called-party transformation settings configured at the route pattern
andat the route list are ignored when checking thecalled number
against the catled-partytransformation patterns.
ThepreIran sformed (originally called)numberis used instead.
If the pretransformed numbens not found in anytransformation
pattern, thentheroute pattern androute list digit manipulation is
appliedas usual.
If transformation isrequired before checking thepretransformed
number against the transformation patterns. You must use translation
patterns and perform digit manipulation there.
The . ...,! the. ..:..; : if. .<. m,*-. i (":.
When using called or calling-party transformation CSSs at trunks and galewavs, the applied
logic is not very intuitive. Therefore, you must be aware ofhow global transformations work.
The transformations that are configured at the route pattern and at the route list are ignored
when searching for amatch of the called or calling number in the corresponding transformation
patterns. The so-called pretransformed number, this is the original number that is used in the
call-routing request, is used instead.
Note
If a call matched a translation pattern previously, the new call-routing request (after
transformations configured at the translation pattern were applied) isconsidered. In this
case, the pretransformed called- and the calling-party numbers are not from the original cal
they arefrom thecall that was generated by thetranslation pattern.
If the pretransformed number is not found in the available transformation patterns, then the
standard digit manipulation logic ofroute patterns and route lists is applied.
Based on the rules that are described, ifyou need to change the called or calling-party numbers
that are used at the original call request before they are matched against the correspondiim
transformation patterns, you have to change them with atranslation pattern; you cannot change
themat the route pattern or route list.
3-138 Troubleshooting CiscoUnified Communications (TVOICE) v8 0
9km-
Cisco Unified Communications Manager Digit Manipulation
Operation
Consider several dependencies when you configure amix ofsome ofthe digit manipulation
methods that were listed earlier. The figure illustrates how Cisco Unified Communications
Manager performs digit manipulation at calls where such dependencies exist.
Manager Digit Manipulation Operation
d
From gateway
trunk or phone
Translation pattern
Result of digit manipulation
performed here is basis for later
manipulation (pretranslonned
patterns r-T ;, PT-D]
Route pattern
->: Sand ) S, PT-D and T-O
! f = PT-S * S-RP DM
T-D =PT-D +D-RP DM
1
Route list, route group:
-. i. and :-., PT-D and T-D
IfS-RLDM; TS = PT-S * S-RL DM
D-RL DM' T-D = PT-D * D-RL DM
Gateway
or trunk
Calling
H
Caller ID DN
configured?
The digit manipulation options are not simply performed sequentially. Their application
depends on the existence ofother digit manipulation settings that affect these elements:
Called-party transformation settings that are configured at the route pattern and route list
and called-party transformation CSSs applied to gateways and trunks. Ifany combination
out ofthese three options is configured, special interactions need tobe considered.
Calling-party transformation settings that are configured atthe route pattern and route list
and calling-party transformation CSSs applied to gateways and trunks. Ifany combination
out ofthese three options isconfigured, special interactions need tobe considered.
The figure shows adecision tree for both the calling- and the called-party number. These
abbreviations are used in the figure:
B px-spretransformed source: This isthe calling-party number asseen in the call
request.
m pT-Dpretransformed destination: This is the called-party number as seen in the call
request.
T-Stransformedsource: Thisis thecalling-party number as transformed bythe
appropriate digit manipulation feature.
T-Dtransformed destination: This is the called-party number as transformed bythe
appropriate digit manipulation feature.
S-RP DMsource route patterndigit manipulation: This is the setofcalling-party digit
manipulation methods as configured at theroute pattern.
D-RP DMdestination route pattern digit manipulation: This is the set of called-party
digit manipulation methods as configured at the route pattern.
S-RI, DMsource route list digit manipulation: This is the set of calling-party digit
manipulation methods as configured at the selected route group ofthe route list that fs
referenced by the routepattern.
D-RL DMdestination route list digit manipulation: This is the set of called-party digit
manipulation methods as configured at the selected route group ofthe route list that is
referenced by the route pattern.
When the call request that is coming from an entity such as aphone, gateway, trunk, or
translation pattern matches aroute pattern, the called and calling numbers as received in the
call request are remembered as pretransformed calling-party (PT-S) and pretransformed called-
party (PI-D) numbers. In addition, the transformed calling (T-S) and called (T-D) party
numbers are built by applying the corresponding set ofdigit manipulation methods as
configured at the route pattern (S-RP DM for the calling-party number and D-RP DM for the
called-party number},
Note If no digit manipulation is configured at the route pattern for the called or the calling-party
number, then the affected transformed number equals the pretransformed number
If the route pattern does not directly refer to agateway or trunk but to aroute list, the following
happens:
The route lists selects the egress device that is based on its route groups and the available
route group members.
Note The same digit manipulation methods that are available atroute patterns, translation
patlerns, and hunt pilots areconfigurable per route group at the configuration pages of a
route list.
It any of the available called-party transformation methods is configured (by default they
are ail unconfigured), then the complete set ofcalled-party transformation methods that arc
configured at the route pattern is ignored and the complete called-party transformation
configuration of the selected route group isapplied.
Note It is always the complete setof digit manipulation methods of either the route pattern (if all
methods atthe route group of the route list areunconfigured) or the route list that is applied
tothe called number provided inthe call-routing request.
In other words, the transformed called number that is received from the route pattern (T-D)
is cither kept (ifno called number digit manipulation is configured at the route list) or
replaced by applying the appropriate set ofrules (D-RL DM) toIhe pretransformed number
(PT-D).
The same logic independently applies tothe calling-party transformation. Therefore, it is
possible to perform digit manipulation to one number (for example the called number) from
the route pattern while the other number (for example, the calling number) is using the digit
manipulation settings of the route list.
At the egress gateway or trunkeither directly selected from the route pattern or via aroute
list digit manipulation operation continues asfollows:
Calling-party number
Ifa caller ID directory number is set, the configured number isused for the calling-
party number.
Ifno caller ID directory number isset, Cisco Unified Communications Manager
checks if a calling-party transformation CSS isconfigured.
Note You canset a calling-party transformation CSS at thegateway ortrunk oryou canactivate a
check box thatinstructs Cisco Unified Communications Manager to usethe calling-party
transformation CSS that is configured at the gatewayor device'sdevicepool. Ifboth are
configured, the device pool transformation CSS ispreferred over the locally configured one.
Ifno calling-party transformation CSS isconfigured, the transformed calling-party
number (T-S) isused as caller ID. As explained earlier, the transformed calling-
party number was built by applying either the rules ofthe route partem (S-RP.DM)
or the route list (S-RL DM) tothe pretransformed calling-party number (PT-S).
Ifacalling-party transformation CSS isconfigured, the pretransformed calling-party
number (PT-S) is checked against the available calling-party transformation
patterns.
If found, thetransformation rulesthatareconfigured at thecalling-party
transformation pattern are applied to the pretransformed calling-party number and
the resulting number is used as the caller ID.
If no match is found, thetransformed calling-party number is used for thecallerID.
Called-party number
The same logic that was described for the calling-party number applies to the called-
party number. The only difference is that the first check (caller ID directory number
configured) is not applicable to thecalledparty.
Misconfigured Discard Digits Instructions Example
3-142
This section describes the PreDot DDI example.
lisconnjuici* l* m
instructions t&amose
" PiB:>-alll!3-:i>si.'aT'a>h.(i-
,. P'i&silVIOC-^C
"" F&:. 1 Co'CJ
P=w"io1"il5ti3triiTrsil.if=l
Cisco Unified
Communicalions
Manager
.J1MV.S&? ^* f ..^b, ?*.... ^
Match. 91212.XXXXXXX
Discard: PreDot
User Dials: 91212555Q120
PrelransformDigitStnng=912125550120
|PretransformTagsListÂCCESS-CODE.SUBSCRIBER
|UnconsumedDigits=
|TagsList=SUBSCRIBER
Reorder Tone
(Number
not assigned!
4>/
12125550120 Mew York
ADDI removes parts ofthe dialed digit siring before passing the number on to the adjacent
system. ADDI removes certain portions of the dialed string, for example, when an access code
is needed to route the call to the PSTN, but the PSTN switch does not expect that access code.
Digit stripping is configured under the called-party transformations by selecting aDiscard
Digits Instruction. It can be configured at route patterns and at route groups ofaroute list or
applied through CSS (as called-party transformation) to agateway ortrunk.
For North American Numbering Plan (NANP) patterns i(a>), the entire range ofDDIs is
supported, with non-fa patterns only DDIs <Nonc> NoDigits, and PreDot can be used.
Note
<None> means that noDDI isconfigured at theroute pattern, but ispermitted in theroute
group configuration of a route list. NoDigits means that any DDI configured at a route group
of a route list has to be ignored. In all other cases, the DDI atthe route group of a route list
has higher priority than theDDI that isconfigured at theroute pattern.
For the PreDot DDI to work, the route pattern has to include a"."(dot) sign which is not
dialed but uêd by Cisco Unified Communications Manager to determine how many digits to
strip (all before the dot).
In Cisco Unified Communications Manager Administration, the Discard Digits menu that is
shown in the tigure is available thru Call Routing >Translation Pattern or Call Routine >Route
Pattern.
PreDot. PreDot-Tratlmg-? and NoDigits DDIs are the only DDIs that can be used ifthe pattern
does not contain the (a sign.
Troubleshooting CiscoUnified Communications (TVOICE) v8 0
In the shown example. Cisco Unified Communications Manager applies the PreDot DDI to the
91212.XXXXXXX route pattern, strips the 91212 from the dialed digits, and sends only the
5550120 tothe PSTN. Because the original called number isthe national number type, where
area code is important, this incorrect digit stripping produces awrong number. Either the call
reaches a wrong party, or thePSTN returns a reorder tone.
On the left side, Cisco Unified Communications Manager tracing output shows how the called
number ismanipulated. The PrctransformPositionalMatch field contains the number to be
manipulated at the colon position, while the PositionalMatchList field contains the called
number after the DDI operation.
The DDI on the route pattern in Cisco Unified Communications Manager may not work for two
reasons:
Ifyou use aroute pattern that does not contain the @wildcard, the only discard instructions
that will work arePreDot, None, andPreDot/Trailing #. While youcanusetheother
discard instructions, they arcvalid only if you use the @wildcard intheroute pattern.
Ifyou define adiscard instruction in the route pattern and configure the discard instructions
ofNoDigits instead ofNone in the route group configuration, the route group programming
overrides the route pattern. Overriding causes the route group to restore the number tothe
one that the user originally dialed.
Use the Cisco Unified Communications Manager Dialed Number Analyzer tool toassist in
seeing the call flow. The Cisco Unified Communications Manager Dialed Number Analyzer
application will list how the discard digits are seen. In other words, ifyou are discarding digits
under the Route Pattern field and have discard instructions under theroute list, Cisco Unified
Communications Manager Dialed Number Analyzer will show the route list discard
instructions. Cisco Unified Communications Manager Dialed Number Analyzer isa very handy
tool to see discard instructions.
Overview of Cisco IOS Gateway Digit Manipulation
When using number modification on the gateway, the selection depends on the cornplexitv of
requirements.
*"'l*ltn Kit U.^vi,i/
d an ip illation
1 .,. i
In the case of aCisco Unified Communications Manager simply sending calls out and receiving
them, basic digit manipulation is enough.
Ifyou have more complex requirements, like searching within the number for astring or
replacing a part ofthe number in the middle or on different places, you have to use voice
translation rules.
<& 2010 Cisco Systems. Inc
Voice Translation Profile Search-and-Replace Example
In this example, you will shrink incoming called E.164 numbers to four digits. You must prefix
the calling number with 9, 91, or 9011 to allow, for example, acallback from the missed call
list.
Voice Translation Profile Search-
Example
." 7 HS,' type subscriber subscriber
'" ",' ;".&' type national naliana!
'*' .'9011&' lypfi ifitKrnsljonal internal!'"
[ce trenslation-rule 1
jle 1 /"40S5552/ 111
Ice translation-rule 2
jle 1 /'./ /9t/ typo subscriber subscriber
iile 2 /*.*/ /91fc/ type national national
uls 3 /".*/ /9011S/ type international international
ice translation-profile pstn-in
ranslate called 1
ranelete celling 2
Strip thefirst sixdigits offtheincoming called number tousetheextension number.
Prefix incoming calling-party ID with 9or 91 or 9011, based onthe incoming TON.
For the called numbers, use these match rules:
/A,*/ I9&I type subscriber subscriber: Match any number with type subscriber and add a
9 in front of it.
/\*//91&/ type national national: Match any number with type national and add a91 in
front of it.
/A.*/ /9011&/ type international international: Match any number with type international
and add a 9011 in front of it.
The service provider is usually sending flags for the type ofnumber (TON). So, for example,
the incoming call from Germany would be 49 40 4132670 with the TON set to international.
Now you have to add 9011 to the number so that the number can be called back.
In Germany for example, you have toadd the 000 tothe call from United States 140X5551001
to display it correctly on the phone and to call it back. For gateways, you can also configure
this in the service parameters. This is ashared parameter. Ifyou have different access codes in
different countries, you have toconfigure them onadevice pool orgateway level.
Incase of anunknown TON, the send number would look like 011 49404132670. You would
add anothersubrule andadd, in this case, the 9 in front of the number.
You never send the PSTN prefix to the service provider and the service provider isnever
sending it to you.
Dial Plan Issues
3-146
This topic explains the procedure to ensure that Cisco Unified Communications Manager and
the gateway are correctly set up with respect to route patterns, route groups, route lists, and
CoS.
Matches dialed number for external calls
Pointsto a route listforrouting
First level of path selection
Points Io prioritized route groups
First
Choice
" Second
Choice
Route group:
Second level of path selection
1 Points to the actual devices
zx
x|BJFirsl
Choice
Devices.
Gateways (H 323. MGCP)
Trunks (SIP, H 323)
IP WAN
??M
Second
Choice
Path selection is performed after digit analysis. Only after the call-routing logic finds the best
match in the call-routing table, will path selection come into play by deciding where to send the
call.
The decision is based on the configuration of the matched entry of the call-routing table. For
external destinations (route patterns), path selection has to choose the egress device such as a
gateway or trunk. Ifthe primary choice isunavailable ordocs not work, you can select a
backup.
Ifno backup path is configured orworking for the matched route pattern, there is no fallback to
another entry (second best match) ofthe call-routing table. The call will tail in such ascenario.
In summary, the call-routing decision -thatis selecting the best matching entry ofthe call-
routing table is done only once, regardless whether the call is successful. You can configure
multiple paths (egress devices) for agiven call-routing entry (route pattern) and thus provide
multiple choices (primary andbackup paths).
Route patterns are strings of digits and wildcards, such as 9.40852MXXX, configured in Cisco
Unified Communications Manager and are part ofthe call-routing tabic. Ifmatched by the call-
routing logic, the route pattern can point directly to adevice such as atrunk or agateway or
point to aroute list. Route lists provide the first level ofpath selection ifmultiple paths exist to
reach the called number that matched the route pattern. They include aprioritized list ofroute
groups and allow digit manipulation to be configured per route group. Arouie group is the
second level of path selection. It points todevices thatarc selected based ona distribution
algorithm(circular or top down).
It is recommended touse the complete route pattern, route list, and route group construct
because it provides the greatest flexibility for call-routing, digit manipulation, route
redundancy, and nature dial plan growth. Ifroute patterns point directly to devices, the
configuration might need to be changed later when additional devices arc added. Also note that
asingle device cannot be used in both ways: being amember in aroute group and being
referenced directly from a route pattern.
Review of Local Route Group Feature
Local route groups have been introduced with Cisco Unified Communications Manager
Version 7.0.
Review of Local Route Group Feature
The main characteristics of local route groups:
Decouple the selection of theegressdevice from the matched
route pattern
Select egress device based on sourceofthe call
Special entry that can beadded tolist ofroute groups at route list
- "Local Route Group" is shown inaddition to configured route
groups
- Can be added one timeper route list{notmandatory)
New setting in device pools
- "Local RouteGroup," which allows for the selectionof any
configured route group
Can be unset (left as <None>)
Local route groups decouple the selection ofthe egress device from the route patterns that are
used to access the gateway. With local route groups, the egress device (gateway or trunk) is
selected based on the matched route pattern. This can greatly reduce the complexity and size of
dial plans inCisco Unified Communications Manager.
Local route groups are implemented by two new settings:
a Standard Local Route Group is anew entry in the list ofroute groups that can be added toa
route list. Aroute listcan include this entry uptoonetime but it is notmandatory toadd
thisentry tothe list of route groups of a route list.
At the device pool, there is anew drop-down list box, Local Route Group. The list
box includes all configured route groups. You can unset the Local Route Group
parameter (set to <None>, the default value), oryou can select one route group.
This way. device pools can be associated with alocal route group. Route patterns that use the
local route group offer aunique characteristic: they allow for dynamic selection ofthe egress
gateway that is based on the device that originates the call. By contrast, calls that are routed by
route patterns that use static route groups will route the call tothe same gateway, no matter
what device originated the call.
D2010 Cisco Systems, Inc.
Cisco Unified Communications Manager Dial Plan
1his section illustrates atypical Cisco 1Inified Communications Manager dial plan.
iX2flUL*lBtii
i*m~~****- Jb&=*& i\
Iviansnar Itiaf uiia*-i
"^"J^ Caller dials
2001 3014001
Ext Mask' 408S55XXXX
Digit Manipulation
Called: DDI PreDot
Digit Manipulation
Calling
Called Prefix 1212^5':
The implemented dial plan uses digit manipulation to formulate calling and called numbers to
the format that isexpected by the target network.
The figure shows the route pattern 301 .XXXX that performs the DDI PreDot. Ifthe caller dials
3014001 to reach the branch office IP phone, the two path options with different number
manipulation requirements exist. The route pattern strips the site code 301 and extends the call
to the route list Branch NY. The route list contains three route groups that are ordered in the
priority sequence fromleft to right.
The preferred path is through the company WAN network. The route group WAN adds the site
code 300 to the calling number to form 3002001 and adds the prefix 301 to the called number
to form 3014001.
The second route group, in order of priorities, is the local route group. This is the device pool
route group that usually points iothe onsitc PSTN gateway of the caller, as shown inthis
tigure. The San Jose gaieway is the closest gateway to the caller that progresses the call to the
PSTN'.
The least-preferred path is through another site gateway (Atlanta, in this figure) that would be
taken ifthe most-preferred paths were unavailable. Both PSTN path options require called-
number modification to reach the format that is mutable in the PSTN. The called number is
prefixed with the 1212555 to form the PSTN number 12125554001. The callmg number is
modified to 4085552001 using the external number mask that was originally configured at the
caller'sphone directory number configuralion.
Depending on uhich path is taken, the called Cisco Unified Communications Manager sees
different calling numbers, and it can optionally implement the digit manipulations to strip all
digits but the 2001.
3-1
48 Troubleshooting Cisco Unified Communicalions (TVOICE) v80
lanagerDial Plan (Cont.)
Manageress
Emerg_Pt, Local_Pt, Nil_Pt, PremiumJ>t, tntl_PP
Denied
Lobby ess
EmergPt, Local^Pt
49B9555020 406-555 0100 311
(Germanyl
Cisco Unified Communications Manager dial plan can get more complex with the
implementation ofCoS, asisshown inthis figure.
You can configure the route patterns for emergency, local, national (not shown on the figure),
premium and international calls to define separate calling permissions. Also, translation
patterns are often used to implement CoS. For instance, translation patterns can effectively
block agroup ofdestinations, like expensive premium numbers (900) in this example.
Callers that are organized ingroups ofdifferent permissions then gain orlose an access to
selected destinations. The IP phone ofthe manager can reach all numbers except the premium,
while the lobby cancall emergency and local numbers only.
The calling search space and partitions are integral parts ofthe Cisco Unified Communications
Manager dial plan.
Verify Cisco Unified Communications Manager Dial Plan
This section explains how to verify the parts of Cisco Unified Communications Manager
configuration that affect thedial plan.
Call Routing > Route Plan Report
3 ri ,
" """
5,1> - 1
S3-,,,,,,
Str, . 1
V ,.. ,
,,.,,
^, I ... , ,...
^l.,,
IIH--- . .., . ,-
53
rf [ j L ,-
i'l '
Jil
1.-,,
a,.,,..,
S^l N son:,
The figure shows aroute dial plan report. It represents avery efficient way to get aquick
overview of thedial plan, including route partition distribution.
Verify Cisco Unifi'
Manager Dial Plan
CallRouting>Class of Control >CSS
^^H'"""^""
",
"o-i^r^'-
I'Z-** ":"1""
"
M^_e-*.H-**H!*.["
Call Routing >Route Plan Report
Most dial plan problems are caused by the incorrect configuration ofcalling search spaces.
Display the CSS of each caller component that is reported as having issues. Based on the
member partitions, verify that the proper components ofthe Cisco Unified Communications
Managerdial plancanbe reached.
}2010 Cisco Systems, Inc
irify Cisco Unified Communication.;
tr Dial Plan l
Call Routing >Route/Hunt >Route Pattern
[>> Pinr trim.ton
. CallâHaflt.
Route patterns are the configuration elements that trigger the path selection process and
perform the optional digit manipulation. Ifyou use this implementation option, make sure that
the correct called number has been produced and that the route pattern points directly to the
proper route list, trunk, or gaieway. It is recommended, however, to use the complete route
pattern, route list, and route group construct because il provides the greatest flexibility for call
routing, digit manipulation, route redundancy, and future dial plan growth. Ifroute patterns
point directly to devices, these devices cannot be reused in any other place, and the
configuration might need to be changed later when addilional devices arc added.
3-152 Troubleshooting Cisco Unifed Communications (TVOICE) v8.i
>2010Qsco Systems, Inc
Verify Cisco Unified Communicati<
mager Dial Plan (Cont.)
Call Routing > Route/Hunt > Route List
A Cf-n i l.-md
ISTKI 1
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,,
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LAJ i*ri, Ffcn*tn--ij"*-*. , C Cl"*1**4flV &
Route list is a collection of route groups that are sequenced in the order of their priority. At the
route list level, per individual route group, you can use selected digit manipulation
mechanisms. Route list digit manipulation comes after the route pattern digit manipulation and
it has priority. It is possible to disable DDI at the route list by setting the route pattern DDI to
NoDigits. The route pattern DDI set to the default <None> allows the route list to perform the
DDI if configured.
DDIs at the route list are NANP-tagged, but some of them can still be used for non-NANP
numbering plans, namely PreDot, PreDot-#, and NoDigits.
)2010Cisco Systems, Inc Troubleshooting Call Setup Issues 3-153
ify Cisco Unified Coninmrnrnhnri.
Manager
Call Routing > Route/Hunt > Route Group
J
Call Routing > Route/Hunt >Gateway
F-U.JJ- .,-.>.:.,.,
tl. -.='~ -4
Ur..,.-.,
IMIjeU..-
.. 1
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-. '-... !.... . . h,. *.<J
*.,=,. .
" '"'-' L.. ,11,=.. SS=,.-H
Llll Horilfnt lnremi*lj
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sii-ii>: 1-4-i-
-.*r; '..,, =,,.,-. ,.-,., ...
UP {*;!,(:., ;cjct
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rv>. :n
A route group is the second level of path selection. It points to devices that arc selected based
on a distribution algorithm (circular or top down). Verify that the route group lists all devices in
their correct order and thai the correct distribution algorithm is applied.
3-154 Troubleshooting Cisco Unified Communications (TVOICE) v8 i 2010 Cisco Systems, Inc
Cisco IOS Gateway Dial Plan
This sectionreviews the CiscoIOSgateway dial plan.
Cisco IOS Gateway Dial Plan
Site Code: 300
4085552XXX
<-f^Mmuimwmtmm*i>mm
Digit Manipulation
Calling: Prefix 3i'0
Called Prefix 301
Site code: 301
2125554XXX
WAN
172.17.10.1
4001
PSTN
Calling: Prefix lOWi
Called: Prefix 1212555
ling
The figure shows atypical H.323 or SIP gateway dial plan. The major component ofagateway
dial plan is adial peer. Based on the two dial peers shown, two path selection options exist in
the figure: ihe first uses the IP WAN network (primary) and the second uses the PSTN
(secondary or backup).
To manipulate the calling and called number to the expected formats, the VoIP dial peer adds
the site code 301 to the called number and the site code 300 tothe calling number. The POTS
dial peer has to manipulate the numbers to the format that is used in the PSTN. The prefix
1212555 isadded tothe called number toform the national number type. The prefix 408555 is
added tothe calling number topresent thecorrect caller ID.
ue1
voice translation-rule 11
rule 1 /"./ /3006/
rule 1 /'4. . .5/ /301t/
raiee translation-prof lie BK-WftN
translate calling li
translate called 12
rule 1 /' .*/ ,'40B555[/
'ice translation-rule 22
rule 1 /'A. . .j/ /1212555s/
'oice transla tion-prof ile BR-PSTN
translate calling 31
translate called 22
dial-peer voice I voip
destination-pattern 4....
translation-profile outgoing EB-WAN
preference 0
session target ipv4:172.17.10.1
translation-profile outgoing BR-PSTN
preference 1
port 1/0
The figure shows the typical Cisco IOS 11.323 gateway configuration for digit manipulation
using flexible voice translation Riles and primary and secondary dial peers.
COR can be used toimplement the calling privileges:
dial-peer cor custom
name 911
name local
name Id
name intl
dial-peer cor list lobby
member 911
member local
dial-peer cor list manager
member 911
member local
member Id
member intl
dial-peer cor list 911call
member 911
dial-peer cor list localcall
member local
dial-peer cor list Ideal1
member Id
dial-peer cor list intlcall
member intl
corlist outgoing 911call
port 1/0
destination-pattern [2-9]
corlist outgoing localcall
port 1/0
destination-pattern [2-9].. [2-9]
corlist outgoing ldcall
port 1/0
i
destination-pattern OUT
corlist outgoing intlcall
port 1/0
Like the previously shown Cisco Unified Communications Manager dial plan, this gateway
offers the two classes ofservice- for a manager and for alobby. Use the troubleshooting
commands that are explained in the next section to manage the issues ofagateway dial plan.
)2010 Cisco Systems, Inc Troubleshooting Call Selup Issues 3-157
Verify Cisco IOS Gateway Dial Plan
In case ofcall-routing issues, verify the gateway dial plan configuration.
Verify Ci&co IOS iaateway Dial Han
Major gateway dial plan troubleshooting commands:
! show dial-peer voice
a show dialplan number
debug isdn q931
* debug voice ccapi inout
* debug voip dialpeer inout
* debug cch323
* debug ccsip
show voice translation-rule
test voice translation-rule
* debug voice translation
* show dial-peer cor
The configuration and correct operation of Cisco IOS gateway is typically verified using these
show and debug commands:
1he show dial-peer voice command displays dial-peer configuration and cause-code ofthe
most recent call.
BS-LKshow dial-peer voice 2929
VoiceEr.capPeer2 92 9
peer type = vo.ee, system default peer - FALSE, information type -
vo:c e,
descr: pi _cn = ' ' ,
tag 2929, dest mat ion-pat. tern = '1 |2-9] ..|2-9] S',
voice reg type = C, corresponding tag = 0,
a.low watch - FALSE
answer-address - "', preference-O,
Cl.II": Restriction = None
CILTD iVetworK Ku~bei = " '
C7.TD Second Huiber sent
OLID Override RDKIS - disabled,
rtp-ssic mux - system
source carrier-id = '', target carrier-id ----',
source trunk-group-]abel = '', target trunk-group-label = ~',
numbering Type - "unknown'
group - 2329, Admin state is up, Opeiation state ia up,
Outbound state is up,
lnco-r.mg called-number = ~', connections/maximum - O/unlimted
DTMF Relay ~ disaoleu,
"RI classes:
Destination -
huntstop = disabied,
:n bound application associated: 'DEFAULT'
out )cund application associated: ''
di". is -m ap -
perm jaaion :bolt:
incoming COS .is::maximum capability
=1^^^
outgoing COR list minimum requirement
Translation profile [Incoming):
Translation profile (Outgoing):
incoming call blocking:
translation-profile = "'
disconnect-cause = "no-service'
advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamiiy 4
mailbox selection policy: none
type = pots, prefix =
forward-digits all
session-target = "', voice-port = "0/0/0:15',
direct-inward-dial - disabled,
digit_strip - enabled,
register E.164 number with H323 GK and/or SIP Registrar = TRUE
fax rate = system, payload size = 20 bytes
supported-language = ''
preemption level - "routine'
bandwidth:
maximum = 64 KBits/sec, minimum = G4 KBits/sec
voice class called-number:
inbound = "', outbound = "'
dial tone generation after remote onhook = enabled
mobility=0, snr=, snr_noan=, snr_delay=0, snr_timeout=0
Time elapsed since last clearing of voice call statistics never
Connect Time = 0, Charged Units = 0,
Successful Calls = 1, Failed Calls = 0, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "10 ",
Last Disconnect Text is "normal call clearing (16)",
Last Setup Time = 365419446.
Last Disconnect Time = 0.
For abrief dial-peer overview, use the show dial-peer voice summary command.
BR-lttshow dial-peer voice nummary
dial-peer hunt 0
AD
PRE PASS
TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET
STAT PORT KEEPALIVE
911 pots up up 911 0 up
0/0/0:15
29 pots up up [2-9] $ 0 up
0/0/0:15
2929 pots up up
0/0/0:15
11 po-S up up
0/0/0 :15
555 vcip UD up 555 S 0 syst ipv4:10.1.1.1
521 voip up up 521555 $ 0 syst ipv4 :10 .1.1 .1
1 pots up up 0
down 0/0/0:15
The show dialplan number command shows the matching dial peers.
BR-l#show dialplan number 5215553001
Macro F.xp. : 5215553001
VoiceOverIpPeer521
peer type = voice, system default peer = FALSE, information type =
voice,
descript ion = ~',
tag = 521, destination-pattern = "521555....$',
voice reg type = 0, corresponding tag = 0,
allow watch = FALSE
answer-address = "', preference^,
1 [2-9] . . [2-9] ... - 0 up
OUT 0 up
. . .truncaled . . .
The debug voice ccapi inout command shows the debug output for call control API for all
signaling protocols.
BK-lit debug voice ccapi inout
voip ccapi ir.out debugging i s on
Mar 3G 19:18:30.959: ,',- -1/DC63 04 FBB15 C/CCAPI /cc_ api_display ie_subfields:
cc_ap; cai. setup_ir.d_commcn:
cisec-username-
ccCaiilnto IF subficids -----
Cisco am =6 0 65 5 54444
Cisco anitype=2
c .scc-:=r..plan-'.
cisco-ar.ipi - Z
cisco-anisi=0
dest-5215;530C 1
Cisco desttype=2
cisco-destpian=l
CISCO-idie-FFFVFtFF
ciscc-rdn=
Cisco iastrdr.=
Cisco-rdntype- 1
ciscc-rduplan- -1
ciscc-rcnp;- 1
c . sco -rd:is i = -1
cisco-redirectreascn=-: fwd final type -0
fma". redirectNumter =
hunt_grcup_timeout =G
Mar 30 19:13:3=.959: ,'/ -1/DC6304FBai5C/CCAPI/cc_api call setup ind coram-
Ir.terface=Cx48BIE538, Call Infot
Ca-lm.g Number--6065554444 ,(Calling Name^) <TON=-lJationa 1, NPIÎSDN,
Screer_mg=Kot Screened, Presentational] owed) ,
Called Numbei=5215553001fTON=Hational, NPI-ISDN),
Lal_i.-,q Trar.slated=FAL5F, Subscriber Type Str-RegularLine,
"ma Ijesr. inat lonl- laq=TRUF.,
Incoming Dial-p.eer=?, Progress Indication=ORIC!NATING SIDR IS NON ISDN(3)
Calling IE Present =TR'JE,
Source Trkgrp Route Label-, Target Trkgrp Route Label-, CLID
Trar.sparent =FALSE! , Call Td^-1
Mar 32 ;9:IS :30.9=9: //-L/DC6304FB315C/CCAPI/ccCheckCl ipClir .-
In: Calling Kurrcer=6065554444(TON=National, NPIÎSDN, Screening=Not
Screened, Present at ion-Ailowed)
y.ar 30 '-3:18:30.959: // L/DCfi304FB8]5C/CCAPI/ccCheckClipClir:
'''c'~ : Ca^ lr:9 ^-mDer=6065554'344 [TON-Nat ional , NPr-TSDN, Screen) ng =Not.
Sereeuea, ,-i esent a: -on-Al lowed;
. . . Lru.ica led . . .
Call InfoiCalling Kurier-6065554444 (TON=National, NPI-TSDW, Soreemna=N.-t
Screened, Presenta:. icn=Allowed} ,
Called Kumtaer-52]5553G01(TOlUNat iona 1, KPI=ISI1N.)
Mar 30 19:18:3C.963: '/9-'/DC6304FBBi5C/CCAPI/cc process_call setup ind;
Event = 0x4 3CBBC4 0
Mar 20 19 :IS :30.962 : .'/-'.; xxxxxxxxxxxx/CCAPI /cc_setupind_maLch_search:
Try wit!: the demoted called number 5215553001
Mar 30 19:18 :3C.967 : //697/DC6304FB815C/CCAPI/ccCallSetCoutext:
Context=0x4SFb4A2C
Mar 30 -.9:13:30.967: .'/657/DC6304FB815C/CCAPI/cc_piocess_Call_seLup_ind:
=>^CCAFI handea cid .397 with tag 1 to app "_ManaqedAppProcess Default"
Mar 3C 19: 13 :3: .y67: /,'697/DC63 04FB8] 5C/CCAPI/ccCa] 1Proceeding :
Progress Inaicat ion=NLT,L :0)
Mar 30 19:18:30.97-.: //6 9V /DC63 04 FB81 5C/CCAPI/ccCal lSetupRequcst:
Destination^, Calling IE Present=TRUE, Mode=0,
Outgoing Dial peer^21, Parama=0x48F55D9C, Progress Indication=ORIGmATiNG
SIDE IS NCI ISD:;i3 i
Mar 30 13:16:30.971; //697/DC6304FB815C/CCAPI/ccChcckClipClli:
In: Calling Nu;nber=6065554444(TON=National, NPI-TSDN, Screening-Not
Screened, Present at ^. on-AI lowed,
3-160 Troubleshooting Cisco Unified Communications (TVOICE] v8 0 2010 Csco Systems Inc
mm
Mar 30 19:18:30.971: //697/DC6304FB815C/CCAPI/ccCheckClipClir:
Out: Calling Number=60G5554444(TON=National, NPI=ISDN, Screening=Not
Screened, Presentation-Allowed)
Mar 30 19:18:30.971: //697/DC6304FB815C/CCAPI/ccCallSetupRequest:
Destination Pattern=521555 S, Called Number=52l5553001, Digit
Strip=FALSE
Mar 30 19:18:30.971: //697/DC6304FB815C/CCAPT/ccCallSetupRequest:
Calling Number=6065554444(TON-National, NPI=ISD, Screening=tJot Screened,
Presentation=Allowed),
Called Number=52155530 01(TON=National, NPI=ISDNI,
...truncated ...
The debug voip dialpeer inout command shows how dial-peer matching is performed at
the gateway.
BR-lltdebug voip dialpeer inout
voip dialpeer inout debugging is on
Mar 30 19:25:08.660: //-1/xxxxxxxxxxxx/DPM/dpAssociatelncomingPeerCore:
Calling Number=5554444, Called Number=555300l, Voice-Interface=0x48BlB538,
Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search
Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Mar 30 19:25:08.660: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore;
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=l
...truncated .,.
Mar 30 19:25:08.660: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result-NO_MATCH!-l) After All Match Rules Attempt
Mar 30 19:25:08.660: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-l
Mar 30 19:25:09.660: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=, Called Number=5553001, Peer Info Type=DIALPEER_INFO_SPEECH
Match Rule=DP_MATCH_DEST; Called Number=5553001
Result=Success (0) after DP_MATCHJDEST
Mar 30 19:25:08.660: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0
Mar 30 19:25:08.664: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=5S5
2; Dial-peer Tag=29
Mar 30 19:25:08.676: //-l/C9701343815D/DPM/dpMatchPeersCore:
Calling Number=, Called Number=5553001, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 30 19:25:08.676: //-!/C9701343815D/DPM/dpMatchPeersCore:
Match Rule=DPJ4ATCH_DEST; Called Number=5553001
Mar 30 19:25:08.676: //-!/C9701343815D/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
...truncated ...
The debug cch323 <option> command shows call control processing for 11.323 call legs.
BR-l#debug cch323 all
All CCH323 call tracing is enabled
Mar 30 19:28:35.189: //-l/xxxxxxxxxxxx/H3 23/cch323_post_call_setup_request:
callID-702
Mar 30 19:28:35.189; //-l/xxxxxxxxxxxx/H323/cch323_post_call_setup_request:
incoming_call=0 incoming_media=0
Mar 30 19:28:35.189: //702/4487748A815E/H323/cch323_call_setup: gw_id=l
Mar 30 19:28:35.189; //-l/xxxxxxxxxxxx/H323/h323_TD_get mlpp_info: callID^=702,
tag=(CC_TAG MLPP_lNFO), Failed AV get
...truncated ...
Mar 30 19:28:35.1B9: //702/44B7748A815E/H323/cch323_get_peer_info: Preferred
codec set to G711_ULAW_64K with Bytes = 160
Mar 30 19:28:35.189: //702/4487748AB15E/H323/cch323_get_peer_info: Flow Mode
set to FLOW_THROUGH
)2010CiscoSystems, Inc Troubleshooting Call SetupIssues 3-161
Mar 3C 19:28:35.189: //702/4487748A815E/H323/cch323_get pcer_inf0: peer:
48FA3E54, peer-avo;ce_peer_tag: 22B, ccb: 4914CFD0
Mar 30 :S:2S:35.189: /,' 702/448 774SAB 15E/H32 3/cchll 23_Set_h.323 parTV.s_frcm dp :
ccd h245addr nits = system [0xG007]
Mar 30 19:28:35.189; //702/4487748A815E/H323/ccll323_update_setup calllnfo:
ca 11ir.gWuT.ber = 55544-14
Mar 3= 19:28:35.193; , / 7C2 /4 487748A3 15L:/K32 3/cch323_store cal J info:
Ca-i setup Flayout Mode: l,mii 60, Min 40, Max 1000, Fax~~300
Mar 30 19:28: 35 .23i: //-1 /xxxxxxxxxxxx/H323/cch.323_ct main: SOCK 2 J-lvent Ox:
Mar 30 19:26:35.233: //-1/xxxxxxxxxxxx/H323/cch323 iev_queue_service: Dispatch
OxB internal event lc H225 3M
Mar 3C 19:28:35.233: //702/4487748A815E/H323/run_h225_sm; Received event
H22s_EV_SETUP while at state H225_TDLE
Mar 30 13:28:35.233: //702/4487748A815E/H323/check_qos and_send setup: Setup
CCD 0X491-1CFCG
Mai -Z -.9:2B :3,.î3 : / =9 2/4437 74 SA8 15E/H32 3/cch.323_I Otary_va 1idate : No
peer ccb available
Mar 3C 19:28:35.233: /,' -1 /xxxxxxxxxxxx/H323/cch323_i ev_queue service: Dispatch.
OxE interna: event to H225 SM
Mar 30 19:28.35.233: /'7C2/448774SA815E/H323/run h22b_sni: Received event
H22=_EV_FS_E':'UP w.-r. 1e at state H225_TDLE
Mar 30 19:28:35.233: //702/4487748A815E/H323/idle_fsSetup hdlr: Setup ccb
0x4 314CFD0
Mar 30 15:28:35.233: //702 ,' 448774 8A8 15E/H323 /idle fsSeLup_ hd 1r:
send fasr.Start setup called
Mar 30 19:28 :35 .23> : / I702/ 4', 8774BA8 15E/H323/send fast.Start Setup: Entry
Mar 30 19:28 :35 ,233:
// /02/44 8774eAS15E./K3 23/'cc:h323 .build_local_encoded_tasLStart OLCs : srcAddreas -
0xAClL-A6, 11245 lport = 0, ilow mode - 1, minimum_qos-Q
Mar 30 13:23:35.233: //702/4487748A815E/H323/h24b_set_local audio mask: Near
end Pref Codecs = G711_ULAW_64K
Mar 3 C 19:23:35.233:
/: 70244S774BA3'. 5t./H323/;:ch323_generic open_logical_channel : current codec -
5 : 16 Z -. 16 0
Mar 30 19:28:35.233:
/, 707 ' 4437748A315E 'H3 23, ,-c'r,3 23_ gener ic open_logical channel : Codec is
C-711 ULAW_64K
...truncated ...
The debug ccsip command is like debug cch323 but isused for SIP call legs.
The show voice translation-rule and test voicetranslation-rule commands are used for
voice translation rule troubleshooting. Thefirst shows the configuration and the latter
simulates digit manipulationby using the selected translation rule.
BR-lffshow voice translation-rule
Translation-ruie taq: 21
Match pattern: ".
Replace pattern: 408555&
Match type: none Replace type: none
Match plan: none Replace plan; none
Trans1at ion -ruie :cj : 22
Rule 1.
Match pattern; "4.5
Hepiace pattern: 1212555S
Match type: none Replace type: none
Match plan; none Replace plan; none
BR-l#test voice translation-rule 21 4001
Matched with rule 1
Original number: 4001 Translated number: 4085554001
Original number type; none Translated number type; none
Original number plan: none Translated number plan: none
The show dial-peer cor command shows the configuration ofCOR at the Cisco IOS
gateway.
)2010 Cisco Systems, Inc. Troubleshooting Call Setup Issues 3-163
Troubleshooting Common Voice Call Issues
1 his topic describes various scenarios and common voice issues that can occur in a Cisci
Unified Communicalions system.
Common voice Call Issues
Commonly experienced voice call issues:
' Caller ID missing or incorrect
* No ringbaek heard
8 Dead air after call setup
= One-way audio
* Call dropped in the middle of the conversation
* Call setup failure to or from the PSTN
Second dial tone issues
These common voice issues will be covered as troubleshootingscenarios
Caller ID missing or incorrect
No ringbaek heard
Dead air after call setup
One-way audio
Call that is dropped in die middle of the conversation
(.'all setup failure to or from the PSTN
Second dial tone
3-164 Troubleshooting Cisco Untied Communications (TVOICE) v8 0
r*
Caller ID Issues
This section describes common caller ID issues and how to resolve them.
Caller ID Issues
Problem report:
Caller IDis missing or incorrect.
Analog ex digital port configuration for caller IO
Station name and number definitions
Caller ID blocked by the called-party or calling-party facilities
Caller IDlocalization issues if globalized routing plan implemented
Cisco Unified
Communications
Manager
PSTN
Caller ID is an analog service that is offered by a central office (CO) that supplies calling-party
information to subscribers. Typically, the calling-party number, and sometimes the calling-
party name, appears on an endpoint device such as an IP phone, PC telephony software
application screen, or the display on an analog telephone. Type I caller ID provides the calling-
party information whilethe call is ringing,and Type 2 caller IDprovides theadditional
convenience of calling-numberdisplay while the recipient is on another call. Currently, Cisco
provides only Type 1 caller ID support.
The caller ID feature supports the sending of calling-party information from FXS loop-start and
ground-stan ports into a caller ID-equipped telephone device. The FXS port emulates the
extension interface of a PBX or the subscriber interface for a CO switch,
The caller ID feature supports receiving calling-party information at FXO loop-start and
ground-start ports. The FXOport emulates a connection to a telephone and allows connection
to a PBX extension interface or (where regulations permit) a CO subscriber line.
Cisco Unified Communications Manager supports caller ID and Calling Name Delivery
(CNAM) features on intercluster trunks (ICTs) and PRI interfaces through the display
information element of the Q.931 PRI protocol. This option is available if the PBX or PSTN
link understands and processes text messages within the display information element and is
connected by way of H.323 ICTs or MGCP gateways.
When troubleshooting caller ID or caller name (CNAM) issues, keep in mind these things:
Caller ID is not supported on MGCP-controlled FXS ports.
On nontoll-free numbers, the calling-party can block caller ID.
Toll-free numbers will have caller ID delivered by the carrier even if the calling party is
attempting to block the call (because the tolls are reversed, it cannot be blocked).
>2010 Cisco Systems, Inc. Troubleshooting Call Setup issues 3-165
Settings in Cisco UnifiedCommunications Manager can be set on the ICT or 11.323
gatewav to block caller ID and CNAM.
FXS ports have caller IDdisabled by default. To enable caller ID, use the caller-id enable
command in the voice port configuration mode.
You can block caller IDon the originating FXS port with the caller-id block command in
voice port configuration mode.
You configure theCNAM under the voice port with thestation-id namesting command. If
no caller IDis received on calls that arrive at the FXOport, the string that is specified with
this command will be used as the calling name.
The station-id number string command provides a number to use as the callingnumber
that is associated withthe voiceport. If the numberstringis provided, it is passedas the
callingnumberwhen a call originates fromthis voiceport. If this parameter is not
specified, the calling number that is attained froma reverse dial-peer search is used. If no
caller IDis received on an FXOvoice port, this parameter is usedas thecallingnumber.
3-166 Troubleshooting Cisco Unified Communicalions (TVOICE) v8 0 2010 Cisco Systems, Inc
It
m,
No-Ringback Issues: Scenario 1
This subtopic discusses common ringbaek issues, provides examples of these issues, and
demonstrates how to resolve these issues.
lo-Kinabacl
Problem report:
No nngback on an IP pnone when calling the PSTN.
- PSTN is not providing ringbaek.
The H.323 gateway is not cutting through the audio.
Call Seltio Direct:;--!"
No Ringbaek!
Cisco Unified
Communications
Manager
Alerting PI = 0
'T1/E1 PRI '~\b
H323Gateway '"
-x
AudioNot Cut Through In-BandRinging
PSTN
^
During theestablishment of an ISDN call, thecall mayor maynot leavetheISDN network
because of some potential issues. These issues could bedue tointernetworking issues between
networks, issues with a non-ISDN user, or issues with non-ISDN switching equipment within
the premises of the user. When such a situation occurs and ringbaek is notheard, a progress
indication is returned to the calling user either ina call control message when a state change is
required, or in the progress message whenno statechangeis appropriate. This progress
message is called a progress indicator (PI).
The PI tells the gateway, the Cisco Unified Communications Manager server, or both, if the
call is end-to-end ISDN and signals if setupmessages are available in-band. Issuescan arise
when the setup messages arenot present or are present but thegateway or theCiscoUnified
Communications Manager doesnot seethem. Oneexample of this is an IPphone or PSTN
phone that does not hear ringbaek when making a call.
Alack of Pisina message assumes thatthe originating device will provide the appropriate tone
signaling to the calling party. Analog and digital channel associated signaling (CAS) PSTN
circuits will usually carry theinformation as in-band information. Conversely, with ISDN PRI,
signaling is out-of-band (OOB) on the D channel.
Toseta specific PI incall setup, progress, or connect messages from anH.323 VoIP gateway,
use the progressed command in dial-peerconfiguration mode. To restorethe default
condition, use the no or disable forms of this command.
When a setup message arrives to the originating gateway with PI=3, this message indicates to
the gateway that in-band messages are available. Pis indicate that tones and announcements are
available as signaled byanalerting, call proceeding, progress, connect, setup acknowledgment,
or disconnect message that is contained in a PI of 1 or 8.
Troubleshooting Call Setup issues 3-167
PI Meaning Message
0 No progress indicator is included. Setup
1 Call is not end-to-end ISDN; further call progress information
may be available m-band
Alert, setup, progress, connect
2 Destination address is non-ISDN. Alert, progress, connect
3 Origination address is non-ISDN. Setup
8 In-band information or appropriate pattern is now available. Alert, progress, connect
For setup messages from a VoIP dial peer, the values are 0, I, or 3. For progress, connect, or
alert messages froma POTS dial peeT. the values are 1. 2, or 8.
Problem Report
A Cisco IP phone user makes an outbound call to the PSTNor PliX through a Cisco IOS 11.323
gateway and does not hear a ringbaek tone.
In this situation, the originating device expects in-band ringbaek tones but one of the following
may be happening:
The PSTN or PBX switch is not providing the ringbaek tone.
The CiscoIOSvoicegateway is not cuttingthrough the audioto the originating device.
The problem in this scenario is that CiscoUnified Communications Manager automatically cuts
through the audioto the 11.323 gateway as soonas the logical channel signal is complete. The
IPphone does not hear a ringbaek tone because thealertingmessage that is sent by the PSTN
did not contain a PI that indicates that in-band (the far end is being alerted) information is
available. Typically, in an end-to-end PSTN ISDNnetwork, the enddevice is responsible for
locally generating ringbaek uponreceiving an alert message. CiscoUnified Communications
Manager relies on in-band ringbaekwhen calling out an 11.323 gateway.
Gather the Facts and Formulate an Action Plan
To fix this problem, use the progressjnd alert enable 8 command underthe POTS dial peer
that matches the outbound call leg of the specific call. As soon as this command is issued, the
Cisco IOS Software treats an alerting message as if it came in with a PI of 8. A PI of 8 means
that m-band information is available and that the gateway is supposed to cut through the audio
when it receives an alerting message.
3-K
Note The progress_ind alert and progressjnd setup commands are hidden in some versions
of Cisco IOS Software and may not be visible within the help parser. However, if the
progressjnd progress command is available in the help parser, the progressjnd alert
and progressjnd setup commands will also be available and can be entered intothe dial
peer intheir entirety. These commands will then appear inthe running configuration.
Troubleshootinq Cisco UnifiedCommunications (TVOICE) v8 0
)2010Cisco Systems. Inc
In this scenario, a user is disconnected froma call.
NoR'tngback issues: Scenario 2
Problem report:
- An IP phone user does not heara busy signal when calling a busy number on the
PSTN. Instead, the call is disconnected.
Might beanindication that thenumber being called isa non-working number.
- The H323 gateway is not cutting through theaudio when PSTN provides in-band
progress tones.
Otsii Rtitiip DsrfiUiari
No Busy Tone
or PSTN
Message Heard
Cisco Unified
Communicalions
Manager D|SCOnnec, p| =8
'T1/E1 PRI
H.323 Gateway
Busy!
PSTN
/
Problem Report
There isan IP phone oranalog phone user (VoIP toll-bypass scenario) who is disconnected
when calling abusy number or who does not hear the announcement from the PSTN that
indicates that anonworking number has been dialed (or both ofthese things happen).
When an IP phone or analog phone places acall to abusy or nonworking number, the call is
disconnected as opposed to the caller receiving abusy tone or amessage from the PSTN that
indicates that the number being called isa nonworking number. In this case, this behavior is
occurring under normal call-clearing conditions.
To resolve this issue, configure the command voice call convert-discpi-to-prog in global
configuration mode (Cisco IOS Release 12.2[ I]or later) on the Cisco IOS voice gateway. This
command causes the gateway toconvert adisconnect message with a PI toa H.323 progress
message with the same value. This ensures that audio iscut through and that the IP phone or
analog phone can hear the busy tones, announcements, orboth, from the PSTN.
In the VoIP toll-bypass scenario, you can resolve most ofthese issues by upgrading the
gateway or gateways to aCisco IOS Release 12.1(3a)XI5 or 12.2(1) or later. However, ifthe
originating device or originating ISDN switch does not keep the call active when aH.225 ISDN
disconnect message isreceived, try using the voice call convert-discpi-to-prog command.
This section describes another possible scenario when no ringbaek is heard.
Problem report.
No nngback ona PSTN phonewhen calling IPphone.
Consider the possibilities.
The H323 gateway assumes that Ihe PSTN isend-lo-end ISDN and assumes that
the far end will generale the nngback
Problem Report
Cisco Unif ed
Communications
Manager
Setup PI = 0
Alerting
T1/E1 PRI
H 323 Gateway
In-band ringing
No Ringbaek!
PSTN
4j/
Anon-ISDN PSTN or PBX user places acall to aCisco IP phone through aCisco IOS H.323
gateway anddoes not heara ringbaek tone before thecall is answered.
As indicated in the previous nngback problem, a nngback issue can occur for calls that come
into the IP network from the PSTN, When the Cisco IOS voice gateway receives asetup
message without a PI. the gateway does not play ringbaek toward the PSTN. This behavior is
because the gateway assumes that the PSTN is end-to-end ISDN and, therefore, relies on the tar
end to play ringbaek to the originating device upon receiving the alerting message. Ifthe
network is not end-to-end ISDN, the setup message should arrive with aPI of 3, indicating that
the origination address is non-ISDN. In this scenario, the device on the PSTN does not send a
PI. Ioresolve this issue, you need to configure the gateway to play ringbaek toward the PSTN
regardless ofthe PI in the setup message. Note that even though you arc trying to resolve the
issue with ringbaek on the POTS side, you must configure the listed parameters under the VoIP
dial peers. The commands that are listed in the action plan force the Cisco gateway to treat
every setup as ifit has a PI of3. This forces the gateway to play in-band ringbaek toward the
PSTN.
Use oneof these solutions tosolve thisproblem:
Use the command progressjnd setup enable 3under the voice dial-peer #voip
command in the Cisco IOS gateway. This command forces the gateway totreat the inbound
ISDN setup message as ifit came in with aPI of 3. This command also causes the gateway
to generate an in-band ringbaek tone toward the calling party if the 11.225 alert message
does not contain a PI of 1. 2, or 8.
An alternate to the progressjnd setup command is the command tone ringbaek alert-no-
pi under the dial-peer voice #voip command. This command causes the gateway to
generate ringbaek toward the calling party ifan alert is received on the IP call leg with no
PI present. Unlike the progressjnd setup command, the tone ringbaek command will not
contain aPI of3. Some devices may not accept setup messages when a PI isincluded. The
tone ringbaek command would be useful in those situations.
No-Ringback issues: Scenario 4
1his scenano illustrates an issue when no ringbaek is heard by the PSTN partv
Problem Report
*4^
Problem report;
- No ringbaek on a PSTN when transferred from an IP phone.
*H225 user information isnot sentto theH.323 gateway by Cisco
Cisco Unified
Communicalions
Manager
H225
User Info!
T1/E1 PRI
H323 Gateway
PSTN
No Rmgbacki
-4?/
APS I"N or PBX user hears no ringbaek when aCisco IP phone initiates acall transfer through
aCisco IOS H.323 gateway. Acommon problem isthe lack ofringbaek toward the PSTN user
when a Cisco IP phone user transfers a call toanother phone.
Ihis problem occurs because the signaling is not passed through by the gaieway when the call
istransferred. This issue occurs when the logical channel for that call istorn down between the
gateway and the transferring device. With the progression signaling torn down, the PSTN user
does not hear a ringbaek tone.
In Cisco Unitied Communications Manager, to get the signaling to send ringbaek upon transfer,
make sure that the Send H225 User Info Message service parameter of the Cisco CallManaiier
sen. ice is set to User Infofor Call Progress Tone.
3-172 Troubleshooting Cisco Unified Communications (TVOICE) v8=
One-Way Audio Issues
This section describes one-way audio and examines the possible reasons why this issue may
occur.
One-Way Audio Issues
Problem report:
* One end cannot hear the other end after a call is set up.
Consider the possibilities;
IP reachability between theendpoints (access list, NAT issue, VPN issue with Cisco
IP soft phones |
H.323 or MGCP binding
Sendingand receiving answer supervision
- cRTP settings on point-to-point links
Cisco Unified
Communications
Manager
Gateway
One-Way Audio
(Either Direction)
Problem Report
Auser reports that calls from an IP phone at the branch office have one-way audio; the branch
user cannot hear the user atthe headquarters but the user at the headquarters can hear the
branch user.
One-way audio is defined as aphone call that is established from an IP station through aCisco
IOS voice gateway or router, where only one of the parties receives audio. The causes of one
way audio in aCisco Unified Communications system can vary, but the root of the problem
usually involves IP routing issues.
Youshould consider these things if youareexperiencing one-way audio:
IP reachability
H.323 binding
MGCP binding
Answer supervision
Compressed Real-Time Transport Protocol (cRTP) settings on point-to-point links
Network Address Translation (NAT) support
VPN issues
Firewall settings
Gather the Facts and Formulate an Action Plan: IP Reachability
When one-way audio occurs, always check basic IP network connectivity first. Because RTP
streams are connectionless (Transported over User Datagram Protocol [UDP]), traffic may
travel successfully in one direction but get lost in the opposite direction.
To verify Layer 3routing, use an extended ping from the first router, setting the source of the
packets to the IP address local to the IP phones, and ping the address of the other IP phone.
Gather the Facts and Formulate an Action Plan: H.323 Binding
When the Cisco IOS voice gateway has multiple active IP interfaces, you can source the II 323
signaling and RTP streams from multiple IP addresses. This behavior can generate various
kinds ot problems, One such problem is one-way audio.
To avoid this problem, you can bind the 11.323 signaling to aspecific source address. 'I he
source address can belong to aphysical or virtual interface, such as aloopback interface Use
the h323-gateway %oip bind srcaddr ip-address command in interface configuration mode
Contigure this command under the interface with the IP address to which the Cisco Unified
Communications Manager refers.
Note The best practice is to use aloopback interface for the H.323 address binding.
On the H.323 gateways, verity' that the loopback interface is defined and that the loopback
interface ,s used as the source IP address for all H.323 messages. Fix any contiguration errors
that you tmd.
Gather the Facts and Formulate an Action Plan: MGCP Binding
One-way voice can occur in MGCP gateways ifyou do not specify the source interface for
signaling and media packets. You can bind the MGCP media to the source interface ifvou issue
the mgcp bind media source-interface interface-id command and then the mgcp bind
control source-interface interface-id command. Reset the MGCP gateway in Cisco Unified
Communications Manager after you issue these commands.
Note The best practice is to use a loopback interface for the MGCP address binding.
On the MGCP gateways, verify that the loopback interface is defined and that the loopback
interface is used as the source IP address for all MGCP messages. Fix any configuration errors
that you find.
Gather the Facts and Formulate an Action Plan: Sending and Receiving Answer
Supervision
Ifyou have aCisco IOS voice gateway that connects to atelephone company (telco) or PBX
switch, verify that answer supervision is sent correctly when the called device behind the telco
or switch answers the call. Ifthe Cisco IOS voice gateway does not receive answer supervision,
the gateway fails tocut through (open) the audio path in a forward direction. This failure causes
one-way audio including DIMF issues. The issue with DTMF will occur when calling an
interactive voice response (IVR) or voice-mail system. Because ofthe one-way audio, DTMF
digits will not bereceived bythe remote end. Aworkaround tothis issue is toenter the voice
rtp send-recv oncommand. This command causes the gateway to establish a two-way audio
path as soon as the Real-Time Transport Protocol (RTP) channel isopened, before the connect
message being received.
If you suspect that this issue could be causing one-way audio, enter the voice rtp send-recv on
command and test to verify that the results are correct. If this does not fix the problem, back out
the command.
Gather the Facts and Formulate an Action Plan: cRTP Settings on Point-to-Point
Links
This issue applies to scenarios, such as toll bypass, in which more than one Cisco IOS router or
gateway is involved in the voice path and cRTP is used. The cRTP, or RTP header compression
method, makes the VoIP packet headers smaller to regain bandwidth. This method takes the 40-
byte (B) IP. UDP, or RTP header on a VoIP packet and compresses it to 2 to 4 B per packet.
This compression yields approximately 12 kb/s of bandwidth for a G.729 encoded call using
cRTP.
The cRTP method is performed on a hop-by-hop basis, with decompression and recompression
on every hop. Each packet header must be examined for routing. Therefore, you need to make
sure that cRTP is enabled on both sides of an IP point-to-point link.
Check the configuration on both sides of the point-to-point link. If you cannot check the remote
end, change the local settings to be the opposite of the current setting for cRTP and test the
results. If this does not fix the problem, back out any configuration changes.
Gather the Facts and Formulate an Action Plan: Minimum NAT Cisco IOS Version
If you use NAT, you must run Cisco IOS Release 12.1(5)T or later for Cisco IOS voice
gateways to support Skinny Client Control Protocol (SCCP) and H.323 version 2 (H.323v2)
with NAT simultaneously. Earlier versions of NAT do not support SCCP protocol translation.
These earlier versions lead to one-way voice issues.
To fix this problem, upgrade the Cisco IOS Software on the gateway.
Gather the Facts and Formulate an Action Plan: VPN Issues
Cisco IP Communicator allows a PC to work like a Cisco Unified IP Phone 7970G. Remote
users who connect back to their company network through a VPN must configure additional
settings to avoid a one-way voice problem. This is because the media stream needs to know the
endpoint of the connection. The solution is to select the VPN IP address, instead of the IP
address of the network adapter in the Cisco IP Communicator settings.
Gather the Facts and Formulate an Action Plan: Firewall Issues
Firewalls, if present in the RTP path, might be configured to block RTP in one direction but not
the other. Verify that the firewall configuration is correct. Change any settings that are blocking
RTP.
Dead Air Issues
This section discusses common issues that can cause dead air and how to resolve these issues.
Dead Air Issues
Problem report:
- PSTN caller sets up call and hears ringbaek. Call is answered by the
IP phone user and dead air is heard by both parties.
Routing issues
- IP access lists filtering RTP
> Firewall fixup issues
Cisco Unified
WAN
Gaieway
Problem Report
A PSTN that is calling dials the direct inward dialing (DID) for an IP phone. Ringbaek is heard
and the IP phone user answers the phone. Immediately, both parties that are involved in the call
hear dead air.
Dead air is a condition in which a call is made but the caller can neither hear the voice nor sec
the video. Because the call setup signaling is independent of the RTP stream, it is possible to
have the call set up but not have RTP flow in either direction. There are several possible causes
of this problem:
There is a codec mismatch.
Access control lists (ACT.s) on the router are preventing connectivity of the RTP streams.
The proper fixup on the firewalls is not enabled.
To resolve this issue, verify these settings:
Check the codecs that are involved in the call to determine if a codec mismatch could be
causing the issue. Install transcoding resources if necessary, or address the codec mismatch.
The gateway does not have any ACLs that would allow SCCP packets and block RTP
packets. If you suspect that an ACL is the problem, you can temporarily remove the ACL
from the interface and test to verify that this is the source of the problem. If you determine
that the ACL is the problem, fix it and reapply it.
Check that the firewall is not blocking the RTP traffic. If the firewall is blocking the RIP
traffic, make any required changes to fix the problem.
3-176 Troubleshooting Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems, Inc.
Dropped Calls
This section describes the common issues that can cause dropped calls and how to resolve these
issues.
Dropped Calls
Problem report:
IP phone has been calling adestination on the PSTN and the call was successfully
set up. The call isthen dropped in themiddle ofthecall.
Agateway using older Cisco IOS Software lost connectivity to its primary Cisco
Unified Communications Manager server.
The remote end accidentally hung up, orwas using a cell phone andlost signal.
The network hasa connectivity eventthatoccurred thataffected the RTP path.
Asystem or phone software enor occuned. Aearner equipment or network issue
occurred.
S'-'Jcs>33t'ul Call Seuip, E.yistsr.j Csii
Manager
Call Suddenly Dropped
T1/E1 PRI
H.323 Gaieway
PSTN
Problem Report
The IP phone calls adestination on the PSTN and the call successfully sets up. The call then
drops inthemiddle of thecall.
The biggest challenge with isolating issues that are related to dropped calls is to quickly isolate
the cause. Determine whether the dropped calls are isolated to one phone orto agroup of
phones. Perhaps you will find that the affected phones are all on aparticular subnetwork or at
the same geographical location. Some possible causes for dropped calls arc as follows:
An II 3">3 gateway lost connectivity to its first choice Cisco Unified Communications
Manager server. If this release is comes before Cisco IOS Release 12.4(4)9T, after the loss
of connectivity, any active calls will bedropped.
The remote end accidentally hung up, orwas using a cell phone and lost signal.
The network has aconnectivity event that occurred, which affected the RTP path.
Asystem orphone software error occurred and adevice needs restart.
Acarrier equipment or network issueoccurred.
To resolve this issue, verify these items:
Check the logs todetermine if a gateway reset occurred.
Ask with whom the caller was talking and if the caller knows if the remote-end device was
a cell phone.
Use ping ortraceroute Io verify connectivity.
Check for alerts that can indicate that asystem problem orerror occurred.
Check with the carrier to determine if any issues are occurring in the PSTN.
Use the Cisco Unified Communications Manager Real-Time Monitoring Tool (RTMT) to view
the system log for phone orgateway resets.
You see one warning and one error message for each IP phone that resets. These messages
indicate that the IP phone cannot keep its TCP connection to the Cisco Unified
Communications Manager server alive, so the Cisco Unified Communications Manager server
resets the connection. 1his behavior can occur because aphone was turned off or aproblem
might exist mthe network. If the call is going out of agateway to the PSTN, you can use the
Call Detail Record (CDR) to determine which side is hanging up the call. You can obtain much
ot the same information by enabling trace generation on Cisco Unified Communications
Manager. In the trace output, look for the entries that indicate phone reset events.
Stationlnit: TCPP1CU [ 1.100.9.10] Keepalive
timeout.|1,100,49,1.2463*10.1.4.14'SEP0024C4455561
Closing Station connection DeviceName=SEP0024C4455561 TCPPid
- [1.100.9.10], IPAddr=10.1.4.14, Port=50083, Device
Controllers [1,50,3] |1,100,49,1.2463*10.1.4.14*SEP0024C4455561
EndPomtUnregistered - An endpoint has unregistered Device
name:SEP0024C4455561 Device IP address.10.1.4.14 Protocol:SCCP
Device type:436 Device description:BRl Phone 1 Reason Code'13
Device MAC address:0Q24C4455561 IPAddressAttributes:3 App
ID:Cisco CallManager Cluster ID:CID10.1.1.1 Node ID-CUCM1-
lJAlarmSEP0024C4455561"**SEP0024C4455561
LineControl::sendSNFNotifylndForPresenceWithAlerting
mPrecenceWithAlertingChangeNotifySubscribed=0,
calllist#=0|1,100,49,1.2463*10.1.4.14ASEP0024C4455561
DeviceManager:star_DeviceStop NameSEPO0.24C4455561
Key=9a6ddea5-6cc8-0606-eeea-3e31c2217976 RegisterDevice=12
DualModeFlag=0|1,100,49,1.2463*10.1.4.14*SEP0024C4455561
21:54:16,102 |LineControl - Delete index
=1|1,10 0,4 9,1.24 63"10.1.4.14*SEP0 024C4455561
3-178 Troubleshooting Cisco Unified Communications (TVOICE] v8 0 2010 Cisco Systems Inc
Perform these tasks to troubleshoot this call setup problem:
Verify that a route pattern is configured that points ultimately totheproper gateway.
Verify thatthe partition of theroute pattern is included in the CSS settings onthe phone.
Verify that the MGCPconfigurationon the gateway is correct.
Use Cisco Unified Communications Manager RTMTto monitor trunk usage.
Call backseveral timesto determine if the problemis intermittent and couldbe causedby
carrier issues.
'2010 Cisco Systems, inc. Troubleshooting Call Setup Issues 3-181
Tracing PSTN Call SetupDigit Analysis
Ihis section covers the Cisco Unified Communications Manager trace for anoutbound PSTN'
call.
Stationlnit: (
linelnstance=t
)7) SoftKeyE\
Digit Analysis: star DaBeq: daReq . parti tionSearchSpace (Bf 7febb9 -
aS51-3cl7-3ab7-edc57[)b7e6cfl;d4cc61ba-3fl7-8286-eBa3-3e2all3ae33b),
f ll t eredFarti tionSearchSpaceString(HQ Emergency Pt : HQ Intl Pt:HQ L
D Pt:HQ Local Pt:lnternal Pt) ,
partitionSearchSpaceStringlHQ Emergency Pt :HQ Intl Pt;HQ LD Pt:HQ
Local Pt:Internal Pt) 1,100 ,4 9, 1. 2160*10.1.2 .13 "SEP002 4C4454AD8
Digit analysis : match (pi = "2 , - ',,". , -i <
plv="5",
pss="HQ Emergency Pt:HQ Intl Pt:HO LD Pt:HQ Local Pt: Internal Pt"
TodPilteredPss =uHC; Emergency Pt:HQ Intl Pt:HQ LD Pt:HQ Local Pt In
ternal Pt" ,
dd=9",dac-"l"l 1,100, 49,1. 2162 "10 .1.2 .13 "SEP00 24C4454ADB
Digit analysis: potentialMatches =PotentialMatchesExist |
Digit analysis: match (pi =-2 , qcn =- 5215553001", cn="3001",
plv="5",
pss ="HQ_Emergency Pt:HO. Intl Pt:HQ LD PtiHQ Local Pt: Internal Pt",
TodPilteredPss=-HQ_Emergency Pt:HQ Intl Pt:HQ LD Pt:HQ Local Pt:In
ternal Pt" , """"--
dd="91s,dac ="l") 1,100,49,1.2164~10.1.2.13"SEP0 024C44 54AD8
Digit analysis: pc-tentialMatches =Patent ialMatchesExist I
Cisco CallManager service parameter Digit Analysis Complexity changed from itsdefault
value StandardAnalysis (shown as dae-"0" in trace output) to
TranslationAndAltematePattcmAnalysis (shown later as dac-'T' intrace output).
Only the relevant trace output is shown; the rest was truncated for clarity.
The figure starts with the IP phone event: the caller pressed NewCall sofikey button. Then,
digit analysis starts. The first digit analysis output shows the CSS for the IP phone.
Then, digit collection starts. Each collected digit isshowing up and the "dd" field fills up
gradually asthe digits are collected from the IPphone. The CSS isalso repeated per each digit
collected. Until potential matches exist in thedial plan, thedigit analysis continues.
2010 Cisco Systems. Ire
Second Dial Tone Issues
This section discusses the common reasons why you might experience second dial tone issues
and ways to resolve these issues.
Second Dial Tone issues
Route Patterns
User dials string:
9211
The call is routed
User dials string:
92065. . .
^/_XX
Outside Dial Tone
9[2-9]XXXXXX f_:>_._._ .V.-"::?;.]
9.[2-9]XX[2-91XXXXXX f'-?J>-. ST.~. .--..a
il'i'H '''. ->iv'"'" :; "'.I
y :?.;;' .
ummum ,irl ^'. ;"":-- '. -j
The second dial tone heard after the fifth digit is entered
Outside dial tone providedwhen all of the route patterns
that can match after 9XXX have the Provide Outside Dial
Tone option checked.
Problem Report
A user dials 92065551212 and hears the second dial tone after entering the fifth digit.
The only time that Cisco Unified Communications Manager provides an outside dial tone is
when all ofthe route patterns that can match after you dial "9"have the Provide Outside Dial
Tone option checked. Ifsome ofthe patterns do not have this option checked, you will not get
the outside dial tone.
Run aRoute Plan Report for all patterns that begin with 92 (the three matches in this figure
display). This scenario exemplifies poor dial plan design; route patterns that do not need a
seconddial tone shouldnot start withthe samedigits as route patternsthat do needa second
dial tone. To fix the problem, you could change the access code from 9to8, if the dial plan
permits, orchange the 92XX pattern sothat there isno problem.
PSTN Call Setup Failure, MGCP Gateway
fhis section discusses the common reasons why you might experience call setup issues and
ways to resolve these issues.
Problem report.
- Acall from an IP phone fails locomplele tothePSTN destination through an MGCP
gateway, andthecaller hearsa reorder tone orannunciator message
The route patternthat defines the destinationhas a partition that is not inthe CSS
of the IP phone
The dial plan onCisco Unified Communicalions Manager ismisconfigured
- TheMGCP gateway is misconfigured or cannotregister due tonetwork issues.
All circuits to the PSTN are in use
PSTN has an issue
Cisco Unified
1CommunicationsManager
I 'WCall SetupFailure
JH4
>a-1
MGCP Gateway
PSTN
1/
Problem Report
Acall from an IP phone fails to set up to the PSTN destination through an MGCP gateway and
the caller hears a fast-busy tone.
There are several possible reasons for this call failing to setup:
Theroute patient that defines thedestination has a partition that is not intheCSS of the IP
phone. Acaller hears an annunciator message.
The dial plan on Cisco Unified Communications Manager is missing or misconfigured.
The gateway MGCP configuration is missing, incorrect, or incomplete.
The gateway that is defined in Cisco Unified Communications Manager ismissing,
incorrect, or incomplete.
All circuits to the PS'I N are in use.
All circuits at the carrier are busy (rare).
TroubleshootingCisco Unified Communicalions (TVOICE) v8 0
Tracing PSTN Call Setup-
(Cont)
Digit analysis: match (pi=-2-, f qcn=-5215553001",
cn="3001",plv="5",
pBE ='HQ_Biriergency_Pt:HO_Iiitl Pt: HOLD Pt: HQLocalPt: InternalPt",
TodFllteredPHB="HQ_EmBrgency^Pt:HQ_Intl_Pt:HQ_LD Pt:HQ_Loeal_Pt:In
ternalPt",
dd=-91212 55 50120-,dac=-l") | 1, 10 0 , 49 , 1. 2174*10 .1. 2. 13*SEPDD24C44 54A
8
Digit a
results
lalysls: analysis
1, 100. 49, 1.217 4*10.1.2. 13*SEP0 024C44 54AD8
PretransforraCallingPartyNumber-3 001
CellingPartyNuinber = 52 155 53001
DialingPartition=HQ._LD_Pt
,Diali.ngPat tern* 9.1 [2- 9] XX[2- 9] XXXXXX
,FullyOualifiedCollodPartyNumber912125 55 012 0
PotentlalMatchesrHoPotentialMatchesExist
PretransformDigitString=912125 5 5 012 0
Pre tranaonnTagBList=ACCBSS-CODE tSPBSCRIBEH
PretransformPoBitionallfetchLiHt = 9: 1212 55 50120
CollectedDlgits=12125 5 5012 0
One on sumedDi gi t s =
TagsList = SUBSCRIBER
PositionalMatchList-1212 555012 0
When no more potential matches are reported, the digit analysis stops and summarizes the
collected number. The "dd'" field now contains the complete called number as it was originally
dialed. The summary shows PretransformCallingPartyNumber=3001, which is the directory
number of the dialing phone and the CallingPartyNumber=52I5553001, which is the calling
number after an external number mask was applied.
Then, DialingPartitionÎIQ__LD_Pt is the route partition of the matched route pattern
Dia!ingPattern=9. t [2-9]XX[2-9]XXXXXX. The FullyQualifiedCalIedPartyNumbcr=
912125550120 and the PretransformDigitString=912125550120 show the called number as it
was originally dialed. The ColIectedDigits=I2I25550120 and PositionalMatchList=
12125550120 show the called number after the route pattern digit analysis (DDI in this case)
was applied.
i 2010 Cisco Systems, Inc. Troubleshooting Call Setup Issues 3-183
Tracing PSTN Call SetupPath Selection
The trace output continues with the path selection.
.<...*>,* lZnM*,i
SMDMSharedData: : findLoealDevice - Naine=PSTN rl Key =02dla37b- 2706 -
fbe7-d635-71S08a41d45c isftc Lvie = 1 Pid= (1 , 73, 3 ) found
RouteListControi : sidle CcSetupReq - RouteLi st (PSTN rl) .
numberSetup=0 , . , -
vmEnahled=0 1, 100 , 49 ,1 . 2174 *1Q . 1 . 2 . 13"SEP0024C4454ADB
RouteListCdrc - FouteGroup count; = 1
RouteListCdrc :: . , ;-,'':-.-':-; "-' ,';.-." IH- ."RI BUT ION
type =l 1, 100, 4S, 1.2174 "10. 1.2.13*SEP00I4C4454ADB
SMDMSharedData: : findLoealDevice - Name=>'
Key =49b090db-3e5 9-faGB-64 59-d383e8'=ebfOO
found
RouteListCdrc ! iexecuteRouteAction : '.,'' :,>'
Pid= (1,132, 3)
MGCPpnSd:restartO CcSetupReq - S0/SU0/DS1-08HQGW '-.y
Locations reserveBandwidth -- cdcoPID= (1.194 .12 ] Orig=0=Dest = C
At the stage uhen route pattern is matched, the path selection starts. The first row in this output
shows the route list that the route pattern has pointed to. The route list PSTN rl is active and
operational and. therefore, included in the path selection.
The next trace output lines show numberMember 2. This is the number of devices (in this case
gateways) that are at the leaves of the path selection tree, not the number of route groups as one
might think. The number of the route groups within the route list PSTN_rl is 1 (one) and its
name is not show in the output.
The route group distribution algorithm is "lop Down, which is shown as
CDRC SERIAL DISTRIBUTION in the trace. Out of the two route group members, the
gateway SO SUO DSl-Ofa 1IQGW was selected for the routing. Since it is active, the Cisco
Unified Communications Manager immediately extends the call to this gateway
(EXTEND_CALL_TO_CURRENT_MEMBER).
At the bottom of this trace output, regions and locations are shown. Because the IP phone and
the gateway belong to the same region, no bandwidth reservation is needed.
3-184 Troubleshooting Cisco Unified Communications (TVOICE) v8.0 2010 Cisco Systems. Inc
Tracing PSTN Call Setup-ISDN Call Setup, Media Setup
This section shows the trace output that is related tomedia setup.
Tracing PSTN Call Setup-
Media Setup
MGCPHandler send mag SUCCESSFULLY to: 10.1.250.102
-. -,: i- ; ' ':; ;rs> - O/I''ESGW MGCP 0.1
M: recvonly
output omitted ...
e =IN IP4 10.1.250.102
t = 0 0
m=audio 16776 RTP/AVP 18 100
a =rtpoiap:lB G.729/BOO0
a =f mtp : IS anneitb=yes
MfCPHandler -'' rf-'Y ?":'< with RTP PortNum:
16776 1,100,149, 1.793*10.1.250.102*SO/SUO/DS1-0HQGW
Out Message -- ;-'. .-re:
In Message - - 1-e li. ^l'
RouteListCdrc: :s
s.-,,;Ma .. protocol- PrlEuroProtocol
r is<?3 ingK.-.g
!ÛtlT,g
StationD- (0000007) OpenReceiveChannel conferenceID=23303305
paSSThruPartyII>=1677722B milliBeCondPcl-etSize=.2 0
c00preS3ionTypa=15(Miia_Payl0ad_G729AnnexB) P.FC283 3PaylOadType=0
qualifierln=? sourcelpAddr=Ipftddr .type: 0
ipAddr-0x0a01fa660O00O00OO00OOOOOOOOOOOOO(10.1.250.102) . mylP._
IpAddr . type;Q lpv4Addr i00a01020d (10 .1. 2- 131 I1,100, 211.12-1 * *
When the path selection completes and the gateway is chosen, the MGCP CreateConnection
(CRCX) message is originated toward that gateway to set up the media. The message contains
the codec for the media setup (G.729B) and the RTP port number for the gateway side of the
media (16776). The gateway acknowledges the receipt of the MGCP CRCX. Note that this
MGCP message exchange opens the media channel in the receive-only mode (M:recvonly),
fromthe gateway perspective.
This figure shows the media setup being interleaved with the ISDN call setup. Remember that
the Cisco Unified Communications Manager terminates the ISDN Q.931 channel using the
backhauling mechanism. The ISDN setup was sent and was followed by the receipt of the
ISDN proceeding message. Based on this success, the hunting for other members of the route
list is stopped.
The last rows show how the IP phone is instructed to open amedia channel for receiving. The
IP address of the gateway is 10.1.250.102 and the IP address of the IP phone is 10.1.2.13.
StationD: 10000007) startMediaTransmission
conferencelD-23303305 passThruPartyID=1677722a
remote IpAddress* IpAddr. type :0
ipAddr:OxOa01fa'6000000000000000000000000110 1 2in msi
remotePortNumber,16776 mllliSecondPacketSize-20 '
qualifier0ut-?. mylP: IpAddr . type :0 ipv4Addr, OxOa01Q20d (loTi . 2.13)
Message --
-- Protocol- PriEuroProtocol
ipAd^Âddr'.ê-O7' PenKeCei-"lAOt âtua=0,
ipAddr;0:(Oa01020d0000000000000O0000000000 (10. 1.2.13), Port=2396 2
MGCPHandler send msg SUCCESSFULLY to: 10.1.250.102
C: DOO00OO0016394Ba001)600F500000006':l
rj; f^dre-vG'729b' S;0"' tlb8' fx^x:"B
. output omitted .
.-audio 23962 RTP/AVP 18
-IN IP4 10.1.2. 13
In Message - -
Out Message --
PriEuroProtocol
-- Protocol- PriEuroProtocol
.-:,<>ii'; -- Protocol-
The IP phone ,s instructed to use the gateway IP address 10.1.250.102, the gateway RTP port
number 16776, and the codec G.729B to start the media transmission. Remember the previous
media setup. From the gateway perspective, it was for receiving only. The media side being set
up in this figure, transmits toward the gateway.
The IP phone returns its IP address and the RTP port number 23962. This information needs to
be delivered to the gateway to open its media channel for transmission as well So far the
gateway has been terminating the receive-only channel. The gateway is instrucled about the IP
phone parameters (10.2,13 and 23962) using the MGCP ModifyConnectiou (MDCX) message.
The call setup is completed by the receipt of the ISDN connect and by sending the ISDN
connectacknowledge messages. Nowthe call is active.
3-1
86 Troubleshooting Cisco Unified Communications (TVOICEl v6 0
PSTN Call Setup Failure, H.323 Gateway
This section explains what to consider if acall setup fails when you use an 11.323 gateway.
PSTN Call Setup Failure, H.323 Gateway
Problem report:
Acall from an IP phone fails to complete to the PSTN
destmation through an H.323 gateway, and the caller hears a
reorder tone orannunciator message.
The dial plan on H.323 gateway is misconfigured, including
OORand digit manipulation.
The rest is similar to the MGCP gateway scenario.
Cisco Unified
PSTN
H323 Gateway
Problem Report
Acall from an IP phone to the PSTN fails to set up through an H.323 gateway and the call
hears areorder tone or annunciator message. g<eway, ana the call
There are several possible reasons for this call failing to set up:
' phoned Pattem th3t dCfmCS tHiS deStinatin h3S 3Partiti0" that is not in the CSS of the IP
The dial plan on Csco Unified Communications Manager is missing or misconfigured
' incoSy ^C'SC0 Unified Cmmunicatlons Meager is miss.ng, incorrect, or
The Csco IOS H.323 gateway dial plan is misconfigured in terms ofPSTN dial peers dieit
manipulation, or calling privileges (COR). P ' g
All circuits tothePSTN arein use.
All circuits at the carrier are busy (rare).
To troubleshoot call setup problems, check these items:
Verify the partition and CSS settings on the phone and route pattern.
- Verify that aroute pattern is configured that points ultimately to the proper gateway.
Verify that the dial peers are correctly configured on the H.323 gateway.
er
2010 CiscoSystems, Inc
View the trunk usage fromthe gateway.
Call back several times to determine if the problem is intermittent and could be caused by
carrier issues.
3-188 Troubleshooting Cisco Unifiea
Communications (TVOICE) v8.0
Debugging the H.323 Gateway Dial Plan
This section describes the Cisco IOS H.323 gateway dial plan.
Debugging H.323 Gateway Dial Plan
HQGW#debug voice ccapi inout
cc api call setup Ind common:
clsco-usernan"ie = :;'* I r-*-~;2'. . I
cccalllnfo IE gubfields
ciscc-anitype-0
cisco-aniplan=0
cisco-anipi-0
cisco-aniei-l
dest=Q114523365900
cisco-desttype=0
cisco-deatplan-O
.. output omitted ...
/CCTlPI/ccapi call setup ind_counion.!
Interface=6x4B3C493C, Call Infol
'ill inc sj.''j =; =3:i!;5iJJ-:i, [Calling Name-) (TON-unknown,
NPI=tJnknowi, Screenlng=nBer, Passed, Presentation=Allowed)
Called Number==011452336590Q(TON-On)mown, NPI=linknowill
/CCftPI/cc lr it if call volumei
.1.1.1,
Hwidb-SerislO/1/Q
Because the Cisco Unified Communications Manager trace would be much like the one in the
previous scenario with the MGCP gateway, examine the dial plan at the Cisco IOS H.323
gateway. Use the debug voice ccapi inout command to activate the debug. The outbound call
that is placed to the PSTN traverses this H.323 gateway and the summary of the received H.225
call setup is shown at the beginning. The calling number (after applying the external number
mask) is 5215SS3001 and the called number (after applying all digit manipulation at the Cisco
Unified Communications Manager) is0114523365900. The call setup was received from the
H.323 peer with the IP address 10.1.1.1 (Cisco Unified Communications Manager) over the
serial interface 0/1 '0.
3-190
Outgoing Dial-peer=9011, ParamS=0x4BF4EADC, Progress
Indicatinn=NCLL(0) a
.output omitted .
/ CCAPI/ ccCallSetupRequest:
Destination Patterns 117. Called Number^ 0114523365900, Digit
Strip=TBUE H
. output oioitted .
/CCAPI/ccCallSetupRequest:
Passed, Pre.entaticnÛowedK^"1""0"11' NPI =Unk Screening.User,
Called Number=0114523365900(TOfcHnknown, NPI =Un]<nown)
Redirect Number-, Display lnfo=
Account Number,5215553001, Final Destination Flag=TRUE
Guid,802986El-Ea2D-91B9-0200-02010A01020D, Outgoing Dial.peer,9011
/CCAPI/ccIFCallSetupRequest Private:
SPI call setup Hequest Is Success; Interface Type,6, PloMode=l
/CCAPi/cc api .... . - .
Interface=0x4BBlB53e, Progress Indication=NtlLL(0 I
The 11.323 gateway ,s preeonfigured with its own dial plan. The first row shows the tag of the
dial peer that was matched against the called number (901 1). Immediately below the
destination pattern and the called number are shown to summarize the dial-peer matching
process. 6
Then the call setup request is progressed toward the PSTN. The type of number (TON) and the
numbering plan identifier (NPI) are sen, with the call setup request, even though they arc-
unknown (set to unknown by the Cisco Unified Communications Manager).
The call setup was successfully originated via the service provider interface (SPI), which is
ISDN PR! in this case. The call setup request is followed by the receipt of the ISDN call
proceeding.
Troubleshooting Cisco Unified Communications (TVOICE) vS 0
Debugging H.323 Gateway Dial Plan (Cont
/CCAPI/ccCallAlert!
Program mdication.HULMO ), Signal Indie ation-NOT PRESENT (255)
*;SDN-6-CQNNECT: Interlace SerialO/C/0,1 ie nov connected to 0114523365900
/CCAPI/cc apl cap ind:
Destination Interface0x483C493C, Destination Call Id-69, Source Call
Id-670,
Caps (Codec.0x10000000, Pax Rate-0x2, Fax Varsion;-0, Vad-0x2,
Modem-OxO. Codec Bytes-160. Signal Typ-2)
/CCAPI,'ccCallConnacti
Progress Indi cation.MULL 10) . Data Bitmask-Oxl
/CCAPI/cc api caps ack;
Destination Interface-0x4SBlB538, :ai.ii.= !
Id.669,
Capu(Codec.g722-<S4[0xl0000000) . Fax Rate=MX_RATB_VOICE 10x2) , Fa
venionÔ, Vad-ON10x2),
Modem-OFFIOxO) , Codec Bytes-ISO. Signal Type-2. Soq Hum Start-0)
/CCAPI/cc api_- ..- ="-:" '':"
/CCAPI/cc_api_voice_oode event:
Call Entry(Context.0x48P4EA8C)
.11 i :
;, Source Call
The ISDN alerting message was received, immediately followed by the ISDN connect message.
The media were set up between the 11.323 gateway and the IP phone by using the G.722 codec
inthis case. The call identifier for this call is670. The last rows declare the media setup
completion for this activecall.
Globalized Call-Routing Issues
3-192
This topic describes the issues ofglobalized call routing and plus <+) dialing and their related
digit manipulation.
i~ ^^'-t^l^M^M,^MâMeA%t^"^^^Ûa'ia,
uiini
Call Ingress
Call Egress
trom internal.
Calling Localized E164 ^f^^
Called' Localized E.164.
trom External.
Calling. Localized E.164
Called: Localized E 164
Calling Localized E 164
Called Localized E 16-1
Io External
Calling ON
Called according
to local dial rules
(PSTN access
tortes mternahonal
access codes etc )
orE 164 (click io
dial call lists, etc
Gsco Unified
Commuracabons
to Eaiemai Globaizod
CaBing. E.164 Cat! Routing
Called: E.164
~'Jr \o Interna
Calling DN
Called DN
from Externa).
Calling Localized E 1(
Called ON
Since Cisco Unitied Communications Manager v7.0, you can place calls to E. I64 numbers with
a - prefix. Support for - dialing is implemented by recognizing +as adialablc pattern that can
be part otcall-routing entries such as route patterns and translation patterns.
Cisco Unified IP phones can place calls to PSTN destinations by using destination numbers in
H, 164 format with a+prefix. The f sign, however, cannot be dialed from the phone keypad
that is. it cannot be manually entered by the user at the IP phone. Plus dialing from Cisco
Unified IP phones is supported from call lists, directories, speed dials, and applications (such as
click to dial).
To allow t dialing also to be used for callbacks to PSTN destinations, the calling-party number
of incoming PSTN calls has to be in E.164 format with a+prefix. This, however, is usually not
well accepted by end users, because they, typically, do not want to see all PSTN callers with
their complete, international number in E. 164 format. Further, the calling-party number
generally is not provided inE.164 format bythe PSTN itself.
There is asolution to these problems that are implemented by calling-party localization.
Together with localization, the new dial plan approach implements normalization. Calling pan\
normalization means that all PSTN caller IDs are changed to awell-defined (normalized)
format: in this case, this is E.164 format with the +prefix at call ingress. On incoming PSTN
calls, call ingress occurs when Cisco Unified Communications Manager receives the PSTN call
from the PSTN gateway. As the caller ID provided by the PSTN usually is only in international
format, if the call comes from aninternational source, there isthe need tonormalize the caller
ID.
Normalization at call ingress isachieved by globalization {because the normalized format isin
the global E.164 format) in this case; consequently, the calling number ofincoming PSTN calls
needs to be globalized atcall ingress (that is, when the call is received at the gateway).
*m
After the call-routing decision, that is, after the destination phone was determined (based on the
called number), the calling-party number should belocalized at call egress to the IPphone. Call
egress is the process of routing the call toitsdestination (inthis case toanIPphone).
Localization at call egress tothephone ensures thatthepreviously globalized calling-party
number is reduced to a locally used format. Typically, endusersat an IPphone want to seethe
calling-party number in the shortest possible format: A PSTNcaller from the same area code
should be signaled with thesubscriber number only; a PSTN national caller(from another area
code, but the same country code) should beindicated with the national number; only an
international PSTN caller should beshown with thecomplete E.164 number.
Thelocalization, therefore, can bedifferent perphone that receives thecall. Aphone that is
located in the United States should show national (U.S.) callers with 10 digits. Ifthe same
callerplaces a call toa phone that is located inGermany, thecalling number should bein
international (E.164) format.
Such animplementation allows end users tosee only the required digits instead ofseeing all
caller IDs inlong, international format. However, if an end user places anoutgoing PSTN call
bycalling back anearlier PSTN caller, the outgoing call would not use +dialing. This can lead
tosuboptimal results; for example, if the local PSTN gateway is busy andthecall should be
placed using a different gateway (maybe located in another area code or even in a different
country). The advantages of using +dialing foroutgoing calls would not beapplicable tosuch
callbacks.
Toavoid such issues, thelocalization of the calling number onincoming PSTN calls at call
egress tothephone only applies tothe display of thephone (inalerting andconnected state).
Internally, both, the localized calling number aswell asthe globalized calling number are
stored at the phone. When you place acallback, Cisco Unified Communications Manager uses
the globalized number for the outgoing call. This combines both advantages: incoming calls are
shown insimple, short format at thephone display, but complete E.164 numbers with+
prefixes are used when such callers are called back.
The figure provides anoverview about call-routing thatis based on normalized numbers.
Onthe left of the figure, two types ofcall sources illustrate call ingress:
External callers
Their calls are received by Cisco Unified Communications Manager through agateway or
trunk. Ina PSTN gateway, calling and called-party numbers areusually provided in
localized E. 164 format.
Internal callers
Their calls arc received from internal phones. In case ofcalls tointernal destinations (for
example, phone tophone) calling and called-party numbers are typically provided as
internal directory numbers. Incase ofcalls toexternal destinations (for example, phone to
PSTN). thecalling number is thedirectory number (at call ingress time), andthecalled
number depends on the local dial rules for PSTNaccess. These dial rules can differ
significantly per location.
The center ofthe figure illustrates the standards that are defined for normalized call routing. As
mentioned earlier, most calls use theglobal E.164 format, whichis alsoreferredto as
globalized call routing. These are the defined standards:
External to Internal
Calling Party Number: E.164
CalledPartyNumber: Directorynumber
External to External (if applicable)
)2010 Cisco Systems, Inc Troubleshooting Call Setup Issues 3-193
Called Party Number: E.164
Internal to Internal
Calling Party Number: DN
Called Party Number: DN
Internal to External
Called Party Number: E.164
At the right of the figure, twotypes of call targets illustrate call egress:
Gateways
When sending calls tothePSTN, the localized E.!64 format is used for both - the calling
andthe called-party number. Theformat of these numbers (especially of the called-party
number) can significantly differ basedon the location of the gateway (for example,
different international access codes in the United States [011] versus Europe [00]).
Phones
When a call froman internal phone is sent to another internal phone, the call should be
received at the phonewithboththe callingand callednumber-by usinginternal
directory numbers. Because this isalso the fomiat that is used byglobalized call routing,
there is no need for localized call egress in thiscase. When a call from anexternal caller is
sent toaninternal phone, most users (especially users inthe United States) prefer toseethe
calling number in localized format (for example, acall from the local area code should be
displayed with seven digits). Thecalled number is the directory number and usually not
displayed at the phone.
It is evident from the figure that there areseveral situations where thenumbers provided at call
ingress do not conform tothe normalized format tobeused for call routing. The same applies
tocall egress, where the normalized format is notalways used when delivering the call.
Therefore, localized call ingress hasto benormalized (that is, globalized), andtheglobalized
format has to be localized at call egress.
3-194 Troubleshooting Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems inc
"" Inbound Call with Globalized Call Routing
mm
Inbound Call with Globalized Call Routing
Callback number is not shown on display (only internally stored at phone)
Allows users to see simple (short) numbers but enables +dialing to be
used for callbacks
Globalization at
Call Ingress
Incoming CallingParty
PSTN Cpll 4085551234
'- i ifffMA
Calling Party:
+14085551234
_ Pni'V
nispidV'40iV555 1?j4
Atlanta
C?i!'!.-=;j Part)
Display H4085%; 234
C-Jback -140S"^r>1234
Germany
The figure illustrates the process of globalizing the calling-party number at call mgres (that >s,
when me call is received by Ctsco Unified Communications Manager from agateway and th
process of localizing the called-party number at call egress (that is, when the call is extended to
an internal IP phone).
In the example, acall from aSan Jose user is received through agateway that is located in San
Jose. The PSTN provider delivers the call with a10-digit caller ID.
Note
In the United States, it is common to provide the caller ID in 10-digit format on both local and
national calls. Sometimes the caller ID of local calls is provided with seven digits only.
Cisco Unified Communications Manager is configured to normalize the caller_ID to global
(E.164) format; therefore, the caller ID 4085551234 is globalized to +14085551234.
After the destination phone has been determined, based on the called number, the calling-party
number is localized (depending on the location of the phone): Aphone ,n San Jose w.ll see a
localized caller ID of 5551234, aphone in Atlanta wdl see alocalized caller ID of 4085551234,
and aphone in Germany will see alocalized caller ID of +14085551234. AM three phones,
however, will also store the globalized number (+14085551234) in their call lists so that
callbacks arc placedusing+ dialing.
Normalization of Localized Call Ingress on Gateways
This section explains the process of number normalization.
3-196
lizat.'.n *f L'.alrecirji inn^^ ,
on batew.^M
Calling Localized C 164
Called Located E 164
Localized call ingress j
/
Calling number:
mcoming calling-party
sellings dl GW, DP or SP
prefiit. digits stripping or
transformation CSS
Catled number;
Significant digits (GW).
prefix ON or incoming
called-partysettings al
GW DP or SP
prefi* digitsstrippingor
transformation CSS
| GW Gateway. DP Device pool SP Service
parameter
Cisco Untftect
Communications
CallRouting
The figure illustrates how localized call ingress on gateways ,s normalized.
Therequirements arc as follows:
Changing the calling number from localized E.164 format to global F164 fomiat
" S^SnUmbCT from l0Ca"ZCd E'64 taat t0 dMy *' ^
calhnZ" thlngUK: thc "or7al1 of the calling number can be achieved by incoming
mampulat on by using transtormation CSS. As discussed earlier in the lesson these^ttmgs are
configured at the gateway, thc dev.ee pool, or as Cisco CallManager scrv.ee parameter A
example tor agateway in San Jose is given: Parameters. An
Prefix for incoming called-party numbers with number type subscriber: ,-1408
Prefix for incoming called-party numbers with number type national: i 1
Prefix for incoming called-party numbers with number type international: +
S?f ,hC Ca"e,d -mber can be achieved by significant digits that are
conhgur da. the gateway (only applicable ifno calls to other external destinations are
permitted and afixed-length number plan is used) or by incoming called-party settings that can
add apr, ,, smp |gi or pcrform more comp|cx djg. man.puiat,on byPJ (ran j^^.-;
<- -^. In the example, the gateway is configured with significant digits "4;'
Troubleshooting Cisco Unified Communications (TVOICE)
v8 0
2010 Cisco Systems
~ Outbound Call Using Callback from Call Lists
This section describes how Cisco Unified Communications Manager and gateways process
E.164 calls. This section illustrates an outgoing PSTN call.
r>
W^
Outbound Call Using Callback from Call Lists
lRP:u!h[Rn^_PSTNJ m^rQute pattern is matched
TEHO routes the call via France
Called and calling numbersmodified
based on selected gateway
+19 89 555 020
Called-Number Localization
a1 Call Egress
Cisco Unified
Communicalions
Manager
New York
Calling Number
Localization at Call Egress
New York
5550100
12125550100. international
RP: Route pattern
RL: Route list
RG: Route group
In the example, an end user in New York places acall to a^\TN destination thatis.located in
Germany by using aspeed dial, call list, or directory entry, which is in E. 164 format with a+
prefix.
Cisco Unified Communications Manager is configured with one or more route patterns starting
with \+ (typically W). This route pattern refers to aroute list, whiclv -in this case -refers to
two different route groups (New York and Paris). If the call is sent through the New York
gateway the called number is manipulated in away that it results man international NANP
call- 011 4989555020. If the call is sent through the Pans gateway, the called number ot
+4989555020 is changed to 00 4989555020, which indicates an international call (from France
to Germany).
Also calline number is transformed into the international format, independent of the outgoing
gateway. To"the number 5550100 in the local format, 1212 prefix is added as the call leaves the
Cisco Unified Communications Manager toward a selected gateway.
Instead of configuring digit manipulation by using different digit manipulation settings per
route group at the route lists, as previously described, you can use anew dial plan feature ot
Cisco Unified Communications Manager-global transformations.
3-K
Localized Call Egress at Gateways
The figure reviews how localized call cress at gateways is implemented
Internal to
Ejdwnaf:
Caning:E.164
Called: E.164
Cisco Unified
Communications
Manager
Globalized
CaiiRouting
GW Gaieway DP Device
'jEii/*,, ,
Calling- Localized E.164
Called: Localized E 164
Called number:
Called-party
TRANSFORMATION
CSS (GW. DP)
Calling number:
Calling-parly
ira reformation CSS
(GW DP)
fiJiiKtl I
^:tZ;l^T** [kC Cal"!,e ^ nura*' ** *<*' E- '64 format
mmmCrakC t^ Chan8C b>' C0,,fieuri,,S cal,cd aild filing-party transformation patterns,
puttmg them into pamnons, and assigning thc appropriate called- and calling-parry
transformation CSS to gateways. As mentioned earlier in this lesson, you can configure called-
and callmg-party transformation CSS at the device (gateway) and at the device pooh
In the example the called-party transformation patterns that are applicable to the New York
gateway (based on partition and called-party transformation CSS) arc configured as follows;
Transformation Pattern
\+.\
\+.1XXXXXXXXXX
I+1212.XXXXXXX
Performed Transformation
DDI PreDot, Prefix 011
DDI PreDot
DDI PreDot
Note
In this example, the San Jose gateway does not use number types. Therefore 011 has to
be prefixed on international calls and the "1" of national calls isconserved.
The called-party transformation patterns that are applicable to agateway in Paris, France (based
on partmon and called-party transformation CSS) are configured as follows:
>2010Cisco Systems,'
mm
Transformation Pattern Performed Transformation
\+.! DDI PreDot, type international
\+33.! DDI PreDot, type national
\+3340.l DDI PreDot, type subscriber
Note In this example, the Paris gateway is using number types instead of international (00) or
national (0) access codes (in contrast to the New York gateway that does not use number
types).
In the example, the calling-partytransformation patterns that are applicable to the New York
gateway(basedon partition and calling-party transformation CSS) are configured as follows:
\+.! DDI PreDot, type international
VH.XXXXXXXXXX DDI PreDot, type national
\+1212.XXXXXXX DDI PreDot, type subscriber
Note In thisexample, numbertypes are used at the NewYork gatewayforthe calling-party
number. Ifno number types were used, typically, 7-digit numbers (localsource of the call),
10-digit numbers (national source of the call), or more than 10 digits (international source of
the call) are used.
Having nonlocal calling-partynumbers, however, impliesthe use of tail-end hop-off (TEHO)
or PSTNbackup over the IPWAN. This is not permitted insome countries and bysome
PSTN providers. Some providers verifythat the calling-partynumber on PSTN calls that
they receive matches the locally configured PSTN number. Ifa different PSTN number is set
for the caller ID, either the call is rejected or the calling-party number is removedor replaced
by the locally assigned PSTN number.
Thecalling-party transformation patterns that areapplicable tothegateway inParis, France
(based onpartition and calling-party transformation CSS) areconfigured as follows:
H.l DDI PreDot, type international
\+33.! DDI PreDot, type national
V-3340.!
DDI PreDot, type subscriber
Outbound Call Using Local Dialing Rules
This section describes how a regularly placed call that follows local dialing rules is processed
by Cisco Unified Communications Manager and gateways that arc enabled for globalized
routiim.
n-amrnffa imtSmm:, ^.m^hM^^nrnM^:
iuim:
IP Phones in France
TP. 000 i
DDI PreDot, Prefi* +
Called-Number
Globalization at Call Ingress
Cisco Unified
Communicalions
Manager
New York
5550100
12125550100, international
TP: Translation pattern
RP Route pattern
RL" Route lis!
RG Route yroup
A caller places an international call to Germany and dials 90114989555020. Thc number
includes both PSTN access code 9 and the international dialing prefix 011. To use the
globalized routing, thc numberhas to be transformed intothe normalized E.164 format. This is
performed at the call ingress. Thc called number is globalized based on the two translation
patterns:
The translation pattern 901 1.! is used for IP phones in New York.
The translation pattern 000,! is used for IP phones in Paris.
The number is dialed from the phone in New York, therefore, the 9011.! translation pattern is
used. The translation pattern removes the PSTN access code 9 and the international dialing
prefix 011 by usingPreDot DDI and adds thc prefix +. The final globalized callednumber is
-4989555020 and this number can be processed by the globalized routing plan.
The globalized called number -4989555020 matches thc \+! route pattern that extends thc call
by using the route list and route group. Thc TEHO is implemented in this enterprise, and the
call to Germany is routed primarily via thc Paris gateway. To enter the PSTN in France, the
called number has to be localized. Hence, the called number is transformed into 00 4989555020
at the call egress. If the primary' path is not available, the call is routed via the NewYork
gateway and the number must be localized to the corresponding PSTN format as
0114989555020.
Also, the calling number is transformed into the international format, independent of the
outgoing gateway. To thc number5550100 inthe local format, thc 1212 prefixis addedas the
call leaves the Cisco Unified Communications Manager toward the selected gateway.
*
lm
ms
w
Normalization of Localized Call Ingress on Phones
Thefigure illustrates how localized call ingress on phones is normalized.
Normalization of Localized Cal
on Phones
Normalization
Localized Call Ingress
Calling number for
external
destinations:
External phone
number mask
(phone)
Example hr,
=1 =[2-Hl
' in N(-
] '
1 -il .
to V
TL
to External.
Calling- DN
Called according
to local dial rules
(PSTN access
codes, international
access codes etc
Called number for
external
destinations:
Translation patterns
Psampfe (phones in p
"::'--DniPr-j[.<:t o-
>: f-U: s-rfetv-t pr'.-i -
i ' -> UO=P.u'.ki [j-'.-'k
Cisco Unified
Commiinirations)
Manager
toExternal: Globalised
SBEE CaUR^
The requirements are as follows:
" numbed to FT/^^^?*D*a* ** ^ nu,nber from an intema' **y
number to E.164 format. Changing the called number to E.164 format ifany other format
was used (according tolocal dial rules).
For calls to internal destinations: No normalization is required.
As shown in the example, you can achieve normalization of the calling-party number for calls
to ternal destinations by configuring an external phone number mask (inll64 form e
phone. You can achieve normalization of the called-party number by using 1^^^
*the figure gives examples for phones that are located in Hamburg, Germany, and SanIT'
>2010CiscoSysiems,lnc.
Globalized Call-Routing Issues
This section presents the most common issues that you can experience while using globalized
caii routing.
3-202
Common globalized call-routing issues:
*All ofthe regular call-routing issues, plus
Unreachable internal number when calling inbound
Callback not possible
Unreachable PSTN number when calling outbound
Called PSTN party sees a wrong calling number
Called internal party sees a wrong calling number
Interdigit timeout experienced when calling outbound
Because the elobahzcd call routing is arather complex configuration setup, there arc many
potential points of misconfiguration. The number normalization and local,zation processes use
digit manipulation mechanisms and CSS and partition concept extensively to interconnect
various configuration components.
These are the most common issues that arc seen with globalized call routing in addition to the
regular call-routing issues:
Unreachable internal number when calling inbound
Callback not possible
Unreachable PSTN number when calling outbound
Called PSTN party sees a wrong calling number
Called internal party sees awrong calling number
. Interdigit timeout that is experienced when calling outbound; for example, if thc route
pattern +! is set to nonurgent
Verify the Configuration That Affects the Globalized Routing
This section describes which configuration elements toverify when selected globalized routing
issuesare experienced.
Verify the Configuration that Affects the
Globalized Routing
Verify globalized call-routing configuration elements:
If the PSTN destination is unreachable:
Outboundcall called-numberglobalization: translationpatterns for a
phone
- Route pattern \+and its reachability using CSS, partition
Outbound call called-number localization: called-party transformation
CSS (GW,DP)
If the PSTN cannot reach the internal destination:
Inboundcall called-number normalization: significant digits (GW) or
translation patterns, incoming called-party prefix, digit stripping and
transformation using CSS (GW, DP. SP)
If callback to the PSTN is nol possible:
Inbound callcalling normalization: incoming calling-partysellings
(GW, DP, SP): prefix, digit stripping and transformation using CSS
If, for an outbound call, a PSTN destination isnot reachable, verify the following:
Called-number localization at thegateway; that is, check whether theCalled Party
Transformation CSS gives an access to proper digit manipulation elements. The Called
Party Transformation CSS can be configured at the gateway itself or can inherit the CSS
from the gateway device pool. Ifboth are checked, then itwill inherit the device pool,
which may not be desirable.
Called-number globalization ata calling phone; that is, check that translation patterns
realizing globalization are correct and reachable tothe phone via CSS.
Check whether thc route pattern \+isproperly configured and whether it is reachable tothe
translation patterns that perform called number globalization.
If for aninbound PSTN call, thePSTN cannot reach aninternal destination, verify thecalled-
number normalization at the inbound gateway; that is, if significant digits or translation patterns
normalize thenumber tothecorrect directory number format. If theincoming called-party
settings arc used instead ofsignificant digits, check the following:
The correctprefixis applied for the numbertype.
Digit stripping produces thecorrect number.
IfaCSS issetup, the correct transformation configuration is reached.
If a callback to the PSTN isnotpossible, verify aninbound calling normalization byusing
incoming calling-party settings at the gateway, the device pool, orCisco CallManager service
parameters (depending on what is used). Check that the coned prefixes are assigned for an
inbound calling number, that digit stripping isconfigured as expected, orthat the correct
transformationis applied using a CSS.
)2010 Cisco Systems, Inc
3-204
Verify globalized call-routing configuration elements:
Called PSTN party sees anincorrect calling number:
Outbound call calling-number globalization: external phone
number mask on a phone
Outbound call calling-number localization: calling-party
transformation CSS (GW, DP)
Called internal party sees an incorrect calling PSTN number:
Inbound call calling normalization: incoming calling-party
settings (GW, DP, SP): prefix, digit stripping and transformation
usingCSS (isalso unable tocall back)
Inbound call calling localization: calling-party transformation
CSS (phone, DP)
If, for an outbound call, acalled PSTN party sees an incorrect calling number, verify the
following:
Calling number localization: Verify by using calling-party transformation CSS at the
gateway or device pool. Also, check that thc calling number transformation patterns are
reachable to the gateway byusing CSS and partition.
Calling number globalization at the phone: Verify ifthe external phone number mask is
correct andis reused byother dial plancomponents.
If tor an inbound call, and ifan internal called party sees an incorrect calling PSTN number,
verify the following:
Calling number normalization: Verify by using incoming calling-party settings at cither
gateway, device pool, or Cisco CallManager service parameters (depending on what is
used). Check that thc correct pretixes areassigned for an inbound called number or that
digits are stripped properly as the Cisco Unitied Communications Manager is receiving die
call. In this case, thc phone isalso unable to call back due toanincorrect received callmtr
number,
Calling localization at the phone or its associated device pool: Check the calling-party
transformation CSS. and check if the correct digit manipulation elements are reachable
using the CSS and partition.
Troubleshooting CiscoUnified Communications [TVOICE] v8.0
Verifying Globalized Call-Routing Configuration Elements
This section covers a globalized call-routing contiguration example that also includes CoS. It
explains thc mutual relationships between various globalized call-routing configuration
elements.
Verifying Globalized Cal
Configuration Elements
Cisco Unified Communications Manager MGCP Gateway
HQGW
Globalization at Ingress
DN 3001. lnternal_Pt
Ext mask:+1521555XXXX
CSS. Manageress
Manager ess.
HQEmergencyPl,
HQ_Local_Pt,
HQ_LD_Pt.
HQ Intl PI
PSTN Pt
Internal ess
Internal Pt
TP: 9.911, HQ_Emergency_Pt
CSS: PSm<if.s
DDI PreDot, Prefix: +
TP:9.[2-9]XXXXXX, HQ_Local_Pt
CSS: PSJNjrss
DDI PreDot, Prefix:+1521
TP:91.[2-9]XX[2-9]XXXXXX, HQ_LD_Pt
CSS: PSUJ ess
DDI PreDot, Prefix:+1
TP:9011.,#,HQ_lntl_Pt
CSS'SNfJS
DDI PreDot-#, Prefix: +
TP: W,lntemal_Pt
CSS: PSTN ess
The first figure shows howa called and a calling number are globalized at the phone ingress.
For the calling number globalization, the phone external number mask+1521555XXXX is
used.
The called number globalization is a bit more complex. First, the CSS Manager.css is
configured, it includes the HQ_Emergency Pt, HQ Local Pt, HQ_LD_Pt, HQJntl Pt
partitions that are assigned to the translation patterns. The phone uses the Manager.css to access
the proper translation patterns.
The translation patterns are as follows:
9.911 strips 9 and adds the prefix +
9 .[2-9JXXXXXX strips 9 and adds the prefix +1521
91,[2-9]XX[2-9]XXXXXX strips 91 and adds the prefix +1
9011.!* strips 9011 and # and adds the prefix +
Pattern \+! is used for callback and does not performany digit manipulation because the calling
number was stored at the phone in globalized format already.
All translation patternsuse PSTN.css that includes PSTN_Pt partition. This way, the translation
patterns gain access to the globalized call-routing route patterns that are explained next.
Verifying Globalized Cj\\ Uouh
>onfiauration El
Cisco UnifierJ
Communications
Manager
RP. VH.PSTN.Pt
[RL^PSTN rl i
4
[RG PSTN_rg j
! GW' HQGW I
Localization MGCP Gateway
at Egress hqgw
CldPlyTP \+1521 XXXXXXX HQ eld ply Pt
DDI PreDot
CldPlyTP- H 1XXXXXXXXXX, HQ eld ply Pt
DDI PreDot
CldPlyTP H911 HQ eld pty PI
DDI PreDot
ClngPtyTP U'.HQ cinq ply Pt
DDI PreDot TON International
ClngPtyTP \+1 XXXXXXXXXX, HQ cinq ply Pt
DDI PreDot. TON National
ClngPtyTP U1521.XXXXXXX. HQ dng pty,PI
DDI PreDot. TON" Subscnber
Cisco Unified Communications Manager performs the globalized call routing that is based on
the single route pattern \+! that is in the PSTN Pt partition. Thc route pattern points to the route
list, route group, and the gateway f IQOW that is used for the outbound calling.
The figure also illustrates thc localization of thc called and calling numbers at the gateway
egress. For this purpose, the called and calling transformation patterns are used.
The called number is localized by using these called number transformation patterns:
.+.! strips - and adds Oil. It is used for localization of international called numbers.
.-i-1521.XXXXXXX strips -1521. It is used for localization of local called numbers,
-. 1XXXXXXXXXX strips -. It is used for localization of national called numbers.
-.91 1 strips -<-, it is used for localization of emergency numbers.
All thc called number transformation patterns are connected to thc gateway by using the CSS
HQcldpty.css (seen in the next figure) and the partition HQ_cld_pty_Pt.
The calling number is localized by using these catling number transformation patterns:
.+ .! strips < and sets the type of number international. It is used for localization of
international calling numbers.
*1521.XXXXXXX strips -152 I and sets the type of number subscriber. It is used for
localization of local called numbers.
+1.XXXXXXXXXX strips -+-1 and sets thc type of number national. It is used for
localization of national called numbers.
All thc calling number transfomiation patterns arc connected to the gateway by using the CSS
HQ_clng_pty.css (seen in thc next figure) and the partition NQ_elng_pty_Pt.
3-206 Troubleshooting Cisco Unified Communications (TVOICE] v8 I
Verifying Globalized Cal
Cisco Unitied
Communications
Manager
MGCP Gateway
HQGW
DN: 3001, Internal Pt
GW: HQGW
Device Pool: HQ^dp
Significant Digits: 4
Inbound CSS: Internal ess
Called': .. =.!' .= :'
Calling Use Device Pool CSS
Device Pool HQ__dp
ClngPty Transformation CSS: HQ cinq ptvess
. : HQ eld ptv.css
Incoming Calling-Party Settings (Prefixes)
National +1
International +
Unknown +
Subscriber: +1521
HQ eld Dtv.css:
HQ_cld_pty_Pt
HQ clna Dtv.css:
HQ_clng_pty_Pt
This figure illustrates the globalization at gateway ingress. The called and calling numbers that
are received from PSTN need to be reformatted to suit the requirements of the globalized
routing plan.
Internal extensions that represent the phones are four-digit directory numbers in Internal_Pt
partition. The inbound called number is truncated to four digits by thc configured "significant
digits=4" at the gateway 1IQGWin this example. For more options, use the calling-party
transformation settings by adding a prefix, removing leading digits, or by using calling-party
transformation patterns.
The inbound calling number is manipulatedby using incomingcalling-party settings at either
device pool, gateway, or service parameters. This example uses the device pool configured
incoming calling-party prefixes.
The gateway configurationalso sets the CSS Internal.css for the inbound calling to reach the
phone.
The figure also highlights how the called- and calling-party transformation patterns are
associated with the gateway. In this particular example, in the gateway settings, the two check
boxes Use Device Pool CSSfor calling and Use Device Pool CSSfor called are activated. The
correct CSSs are then inherited from the device pool. Another alternative is to configure the
CSSs directly in the gateway configuration, but the two check boxes have to be deactivated;
otherwise, they have a precedence.
Tracing Inbound Globalized Call Setup
This section describes thc trace output for globalized inbound call setup. It especially focuses
on those pans that can be examined to identify potential globalized call-routing issues.
TYacinq Inboun
HQGW* debug sdn q931
ISDN Se0/0/Q:15
Bearer
... trunca ted ...
Q931: RX
Capability
Standard
Transfer
Transfer
Transfer
<- SETUP pd = 8 callref = 0x00 07
i = 0x8090A3
= CCITT
Capability = Speech
Mode = Circuit
Rate = 64 kbit/s
Plan I SDI. , Type;Subscriber(local)
Called Party
Plan
Number i = OxCl, '5553001'
ISDN, Type:Subscriber(local)
ISDN SeO/0/0
truncated
15 Q931 TX -> CALL PROC pd = B callref = 0x8007
ISDN SeO/0/0 15 Q931 TX -> ALERTING pd = B callref 0x8007
This section covers an inbound call setup by using globalized routing. The figure shows the
inbound ISDN call from 5554444 to 5553001. The following figures show Cisco Unified
Communications Manager tracing the output that is related to this inbound call.
Troubleshooting Cisco Unified Communicalions (TVOICE) v8 0 i 2010 Cisco Systems Inc
Tracing Inbound Globalized Call S*
MGCPpn9d::processPriSetup - vlprCgpnS164= [5554444],
viprCdpnE164=[5553001] ... truncated...
SPROCPrl; : giobalizelacamingCgpn - fiddisig presfixj +1521,
Di-j'rs! to -trip; 0, Cgpn Trans format" Ion CSS: |"**
19:06:29.080 |SPR0C :: stripAndPrapondDigitS- The number
5554444 is prepead*d with prefix +1521, updated
".'j:-*?!---15 2It5 5444 4 )***
19i06;29.080 |SPROCPri;:globli*InconiingCgpn - Global!a?
-;;-^r, =- -112:;?5-i-144 !*"**
The Cisco CallManager service parameter Digit Analysis Complexity changed from its default
value StandardAnalysis (dac-"0" in the trace output) to
TranslationAndAltematePattemAnalysis (shown as dac="l" in the trace output in the next
figure).
Thc tracing output starts with the received ISDN PRI setup from the MGCP gateway. The
numbers that are received from the gateway are the calling number 5554444 and the called
number 5553001.
Then Cisco Unified Communications Manager continues with calling number normalization,
and the prefix +1521 is added to the calling number. This makes up the globalized calling
number+15215554444.
>2010 Cisco Systems. Inc Troubleshooting Call Setup Issues 3-209
Tracinq Inbound Gl
Digit analysis: match(pi="2", fqcn="H,
:"..":; -. s 5, plv="5" , pss="Internal_Pt" ,
TodFilteredPss="Internal Pt*,
dd ="3001",dac="l-) .1,100,"150,1.69*10.1.250.102"*
19:06:29.081 .Digit analysis: analysis
results 1,100,150,1.69*10.1.250.102"*
19:06:29.0B1 | | Pre trans formCallingPartyNumber=-( :- ;.,!-_><; j 4-, j
CalllngPartyHumbers-i- r,', : S!-i i-;
DialingPartition=Internal Pt
DialingPattern=3001
FullyQualifledCalledPartyNumber=+15215553001
... truncated ...
IPre transformDigitString=3001
ICollectedDigits=3001 ... truncated ...
SMDMSharedData::findLoealDevice - Name=3 001:15415 9 8d-0bld-
c9 4e-48al-a00768falcB2 Key=b650f63a-80d7-40ee-leb-
102e045624bc isActvie=l Pid=(1,154,1) found ... truncated...
CRCX 11B SO/SUO/DSl-0/iaHQGW MGCP 0.1 ... truncated ...
The MGCP gateway inbound call processing is set for Significant Digits^4; hence, the Cisco
UnifiedCommunications Manager digit analysis shows thc dd^300l (dialed digits). The digit
analysis also shows that the inbound gateway CSS has an access to the Internal Pt.
The digit analysis has completed and shows the numbers summary:
Globalized calling-party number is+15215554444 .
Globalized called-party number is +15215553001 with original pretransform digit string of
3001.
After the digit analysis, thc Cisco Unified Communications Manager locates the target IP
phone with thc directory number 3001.
When it is found and registered (active), thc Cisco Unified Communications Manager sends the
MGCP CRCX message to the MGCP gateway (SO/SUO/DSl-0/lfgiHQGW) to set up media
between the IP phone and the MGCP gateway.
Troubleshooting Cisco Unified Communicalions (TVOICE) v8 0 2010 Cisco Systems, Inc
Tracing Inbound Globalized Caii Setup (Cont.)
SPRQC DUTransforMCatch - s=*t;-nNi;j.Ler t-i"-:5 55'l'i*4 3 transf ormCSSPkid
[S6eS6 4f-98d4-ffa-be52-7709a7a6 8779] transformationCss
[HQ clng pty Pt] pattarnOsaga 115] paternNodeiD If bca52cS-0 E04- fBld-
ebd0-0c4616c"bldd] OutpulsadNum.iid 15S54444] to [4] pi [1] npi
[1] *-'
LocaliieCgpnAcdaentJOutpulsodNumbur; StatlonCdpc on device
SEPC024C4454ADe , CSS - <S6eaS<S4f-86t)4-afa-be52-
7709e7B6B779 .uaeDevicePoolCgpnCaa -1
Alt ernatecgpn (global 1-* 15215554444
cgpn.55 54444 1,1 00, 150.1.69*10.1. 250. 102"'
... ti-uaca ced ...
Stations 10000003) (1,100,9,1J) Calllnto callingPartyKame-'
callingParty-555 4444 cgpnVaiceMailbo*- alternataCalllngPartys
+15215554*44 calladPartyNamB.' call ?*.* r t y - -'O')! cdpnVoiceKailbajo.
onginalCalledPartyName-' 'r;gj a* Li"all*-li^r =.y=1flCI
originalCdpnVoiceMailbo*- originalCdpnP.adiTHCtRaason.0
las tRedirectingPartyliana-'' laatP.adiractingParty-3001
laatRedirectingVoicemilbox- laatRedirac tlngRaaaon-O
callTypo-1(InBound) linaloatance-1 callReferenca-22651996. version;
85720013:1,100,150,1.69"10.1.250.102**
Stationlnit: (00000031
OffHook. 1,100,4 9,1.2073"10.1.2.13*SKP0024C4454fcD8
This figure starts with the calling number localization. The calling number transformation CSS
has an access to the HQ_clng_pty_Pt and the final localized calling number is 5554444.
Cisco Unified Communications Manager sends the calling-party number of 5554444 and thc
alternate (globalized) calling-party number of+15215554444 to the IP phone. These two
numbers are presented to the called party (the alternate called number in phone call lists only).
Finally, the called IP phone goes off-hook, and the media is established (not shown in the
trace).
2010 Cisco Systems, Inc. Troubleshooting Call Selup Issues 3-211
Tracing Outbound Globalized Call Setup
This section descnbes thc call setup trace output for an outbound globalized call.
"racinq C
Stationlnit: (00000031 SoftKeyEvent softKeyEvent=2(NewCall] ..
truncated ...
Digit analysis: match(pi="2",: -.-!.. - i- ,.; ,- - ;;'", cn="3001",
plv=-5",
pss="HQ_Eoiergency_Pt:HQ IntlPt:HQ_LD_Pt:HQ LocalPt:Internal
_Pt",
dd="9",dac="l")|1,100,49,1.1282*10.1.2.13*SEP0024C4454AD8
Digit analysis: potentialMatches=PoteatlalMatchesExiat
... truncated ...
Digit analysis: match (pi =" 2 " , fqcn=". 15215553001" ,
cn^'SOOl'.plv^-S",
pss="HQ_Bnergency Pt :HQ_Intl_PC:HQ_LD Pt:HQ_Local_Pt:Internal
Pt- ,
>-t . -=-r-..= . :- ,dd="914 0855 53690'>,dac=-l-) j1,100,49, 1.1297*10
.1.2 .13*SEP0 024C4454AD8
The output starts with pressing NewCall button at the IP phone and dialing the number
9140K5553690. The digit analysis starts when the first digit is received, in this case the digit is
9. The digit analysis also shows the CSS as TodFilleredPss string. The IP phone has an access
to the partitions shown. After the first digit receipt, potential matches still exist.
The digit analysis continues until thc last digit is received, in this case "0." At this stage, the
entire dialed number is shown in the "dd" string 914085553690.
Troubleshooting Cisco Unified Communicalions (TVOICE] v8 0 2010 Cisco Systems. Inc
Tracing Outbound Globalized Ci
(Cont.)
Digit analysis: analysis results
PretransormCallingPartyNuoibet =].f'. 215^33 001
!CallingPartyHumber=+15215SS3001
DialingPartition=PSTN_Pt
DlallngPattern=\*I
FullyQualifiedCalledPartyNumber=*14085553690 ... truncated
PretransformDigltStrlng+14085553690 ... truncated ...
PretraosIormCalll
CalllngPartyHumbe
DialingPartition.
DialingPattern=91
FullyQualifiedCal
PatteraType=Trana
PotentialHatches
PretraasformDigit
PretransformPosit
CollectedDigits=i-
DnconsumedDigits=
P os i ti onaIMatchLi
. truncated ...
ngPartyNumber=io11
r=+lS~lSS5J"01
HQ LDPt
. [2-9JXX [2-91XXXXXX
ledPartyNujnber=914085553690
lation
NoPotentialMatohesExist
String914085553 690
ionalMatChLlst=91:408 555 3 690
14085553690
Bt-40B555369D
When the digit analysis is completed, you see a summary of the calling and called numbers.
The first part of the output shows route pattern V*! and its partition (PSTN_Pt) with the
globalized called number as PretransformDigitString and the globalized calling number as
PretransformCalhngPartyNumber.
Although, duringthe processing, the translation patterncomes first, in the output, it is shown
after the route pattern. The translation pattern shows howthe called number was globalized.
The translation pattern 91.[2-9]XX[2-9]XXXXXX in thc partition HQ_LD_Pt was matched
against the number that was dialed. In this case, the FullyQualifiedCalIedPartyNumber or
PretransformDigitString is the called number before the translation 914085553690. The
PretransformPositionalMatchList and PositionalMatchList showhowDDI was performed.
Although the translation panemprefix+ is not explicitly seenin thc output, the CollectedDigits
shows the final globalized called number +14085553690.
>20!0Cisco Systems, Inc.
"racing Outbound Globalized Ci
Cont.)
SKDMSbaredData::findLocalDevlce - Name = PSTH rl ... truncated
RouteListControl: :idleCcSetupReq - RouteList(PSTN rll ,
numberSetup =0 :...,=-:...;.= ;;.'-; - j. ... truncated ...
RouteListCdrc::executeRouteAction:
GCPpn9d;;restart0 CcSetupReq ViPR Learning Routes:
viprCgpnE164=[+15215553001], viprCdpnE164-[+14085553690],
vcrDploadNeeded=[t] |1,100,49,1.1297*10.1.2.13*SEP0024C44 5 4A
D8
>"::". . :.- . jC. => . :-i^G,; mgcp 0.1 ... truncated ...
Then the tracing continues with the route list selection. PSTN rl was found. This route list has
only a single route group member and, therefore, thc call is immediately extended to this only
member. Thc Toute group name is not shown in the trace, only its descriptor string (truncated,
not in the figure).
The route group contains the MGCP gateway. During this routing stage, which is globalized,
both the calling and called-party numbers are shown as viprCgpnE 164-[ +15215553001] and
viprCdpnt: 164[-^14085553690], Cisco Unified Communications Manager progresses the call
to thc MGCP gateway (SO/SUO/DSl-O/lfaHQGW) by sending the CreateConnection MGCP
message.
Tracing Outbound Globalized C;
(Cont.)
MGCPHandler rccv CHC'X M* with RTF Portû*;.. L""S|
1,100,149,1.445*10.1.250.102*SO/SDO/DS1-0HQGW
DATransformHatch - matohNumber [ +140B5553690J
transformCSSPkid [66dfd8cd-f03 9-98 57-le9d-aecd7 48f2ed2]
transformationCss [HQ_cld_pty_Pt] patternDsago [20]
paternNodeiD [aa2Q42df-7b3c-ld01-4335-2115DB21ded3]
OutpulsedNum.nd [14085553690J tn [0] pi [0] npi [0J|***'
DATransformMatch - matehNumber [+15215553001]
transformCSSPkid [66ea664f-BBd4-affa-b52-7709a786B779]
" i-::^ t-;r:r.^t i-iri";FS [HO dlna j>ty Pt] pattemllsage [15]
paternNodeiD [fbcaS2c6-0f04-f81d-bd0-0c4 616c2bldd]
Jijl-3-.:if=5.iP.:.t;.:.d [55:0001] tn [4] pi tl] npi [1]|-****
... truncated ... [codec, RTP exchange, media setup follows]
The MGCP gateway returns MGCP ACKwiththe RTPport number. At this stage, thc trace
output showsthecalledand the callingnumber transformations. The first is the callednumber
transformation that uses the CSS with the access to the HQ_cld_pty_Pt to transform from the
globalized +14085553690to the localized 14085553690number format.
Similarly, the calling number transformationuses the CSS with the access to the
HQ_clng_pty__Pt to transform from the globalized +15215553001 to the localized 5553001
number format.
Then the call setup completes with the codec and RTP ports exchange and the media setup
(truncated).
>2010 Cisco Systems, Inc. Troubleshooting Call Setup Issues 3-215
Outbound ISDN Call Debug
This figure shows the debug of thc outbound ISDN call.
HQGW* debug isdn q931
ISDN Se0/0/0:15
Sending
Bearer
... truncated ...
Q.931: TX -> SETUP pd = 8 ca
Complete
Capability i * 0xB090A3
Standard - CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/a
Href - 0x0009
Called Party Number i = OxBO, '14085 553690'
Plan:Unknown, Type:Unknown
ISDN SeO/0/0
... truncated .
15 Q931: RX <- CALL PROC pd = 8 callref = 0x8009
ISDN SeO/0/0 IS Q931: RX <- ALERTING pd = 8 callref = 0x6009
The MGCP gateway progresses thc call setup to the ISDN by using the localized calling
number 5553001 and thc localized called number 14085553690.
3-216 Troubleshooting Cisco Unified Communicalions (TVOICEl v8.Q )2010 Cisco Systems. Inc.
Summary
Summary
Common issues that can occur when calling the PSTN or
across an intracluster WAN link include, dial plan problems,
digit manipulation issues, dial-peer issues, protocol problems,
gateway configuration, and regions- and locations-based CAC
settings.
When troubleshooting gateways, a methodical and structured
approach is best.
Cisco IOS voice gateways analyze digits one at a time until an
explicit match is found. If more than one match is found, the
preference command is used to break the tie. If the preference
is equal, the router randomly chooses.
DDI is one type of digit manipulation that can cause issues if
incorrectly used.
Cisco Unified Communications Manager dial plans use route
patterns that point to a route list that has one or more route
groups. Each route group then has one or more gateways.
Summary (Cont.)
Commonly experienced voice call issues include missing or
incorrect caller ID, ringbaek problems, dead air, one-way
audio, dropped calls, and various call setup failures.
Globalized call routing uses single-route pattern and plus (+)
dialing. Extensive digit manipulation mechanisms, if they are
misconfigured, can cause issues.
Uponcompleting this lesson, you are able to explain the common calling issues that can occur
with off-net calls and identify the most likely causes of these issues.
References
For additional information, refer to these resources;
Cisco Systems. Inc. Cisco Unified Communicalions Manager Administration Guide
Release 8.0(11, February 2010.
http: uuw.cisco.com en l.'S docs voice_ip_eomm.ciicm.;admin,'8 II 1 cemefg
bcem-801-cm.html
Cisco Systems. Inc. C'isco Unified Communications Manager Features and Services Guide.
Release fi.O(l), April 2009 and updated April 2010.
hup. \\ wu .ciêo com en US doc-, voice ip comm cuenv'admin 8 ll_! cemfeiit
t<gd-S0|-em html
Cisco Systems. Inc. Troubleshooting Guide for Cisco Unified Communications Manager.
Release H0(1}. February 2010.
hup: u \\\\.csi.eo.com en US'does \oice ip cornnraicin/trouble/S 0 lrrhliS01.html
3-218 TroubleshootingCisco Unified Communications (TVOICE) v6 0 2010 Cisco Systems, Inc
Module Summary
Module Summary
Commonly experienced call setup issues include call setup
failure, caller ID issues, inefficient routing, and redirecting
number issues.
Call setup failure within a single site is often caused by either
misconfigured CoS settings, invalid directory numbers, or an
unregistered directory number.
Unique issues that can arise between multiple sites or with
intercluster calls include overlapping dial plans, a call routing
over the PSTN when it should go over the WAN first, and call
failure immediately after call setup at a remote site.
When troubleshooting off-net Cisco Unified Communications
Manager dial plans, remember that they use route patterns that
point to a route list. A route list can have one or more route
groups as options. Each route group contains one or more
gateways.
In this module, you learned to diagnose call setup issues and resolve the issues as you discover
or reveal them, given a trouble call for which the source of the problem is unknown.
References
Cisco Systems, Inc. Cisco UnifiedCommunications ManagerAdministration Guide
hup: 'v\\\\v,cisco.cLmieii/L!S/docs/'voicejp_comnV'ciicm/admin,'8 0 l.'ccmcfg/
hccm-H01-cm.html
Cisco Systems, Inc. Cisco Unified Communications Manager FeaturesandServicesGuide,
Release8.0(1), April 2009 and updated April 2010.
http: www.eisco.com enUS/docs/voice ip comnv'cuem''admin/8_0_l/ccmfcat
fsgd-801-cm.html
Cisco Systems, Inc. Troubleshooting Guidefor Cisco Unified Communications Manager.
hup: www.ciseo.comcn'US/docs/voice ip comm/eucm/trouble/K 0 l/trbl801.hlml
Cisco Systems, Inc. Cisco Unified Border Element Configuration Guide, Release 15.1,
March 2010.
http: wwH.ci^co.conv'en/US/docs;ios.'voice';cubc.'conftguration/guide/15_l/
vb 15 1 Book.htm!
Cisco Systems. Inc. Troubleshooting Gatekeeper Endpoint Call Admission Issues, February
2006.
http: ûu.ci-cu.coin e:i U'S tech tkl0~7. technologies tech notc0l S6aOf)K00e654c.shmil
3-220 Troubleshooting Cisco UnifiedCommunications (TVOICE) v8 0 2010 Cisco Systems Inc
%
Module Self-Check
Ql) What is the most typical issue that causes a call setup to fail within a single site?
(Source: Examining Call Setup Issues and Causes)
A) a translation pattern is configured as urgent
B) caller ID is missing or incorrect
C) incorrect CoS settings such as CSSs and route partitions
D) destination not registered
Q2) Which problem would never cause a multisite intracluster call to fail? (Source:
Examining Call Setup Issues and Causes)
A) a codec mismatch between the two endpoints
B) CoS settings on thc endpoints
C) incorrect digit manipulation
D) MTP not available
E) RSVP-based CAC blocks the call
F) incorrect gatekeeper configuration
Q3) What are four common call setup failure causes when call setup fails between Cisco
Unified Communications Manager clusters? (Choose four.) (Source: ExaminingCall
Setup Issues and Causes)
A) incorrect CSS configuration at local end or remote end
B) location-based CAC preventing the call
C) cRTP mismatch between the two sites
D) gatekeeper CAC mechanismthat might be preventing the call
E) PSTN having problems
F) MGCP gateway that is not registered with any of Cisco Unified
Communications Managers
G) call control discovery-related partition thatmight not be included inthe calling
device CSS
H) Cisco UnifiedCommunications Manager database replication errors
Q4) What are fourof the most common issueswhilecallingon-premises? (Choosefour.)
(Source: TroubleshootingOn-Premises Single-Site Calling Issues)
A) transcoder allocation issues
B) CoSmisconfigured in CiscoUnifiedCommunications Manager
C) H.323 gateway not routing a call
D) digit manipulation misconfigured
E) target that is unknown or unregistered
F) high CPU utilization
G) wrongconfiguration of a called-party numbertransformation pattern
H) voicemail or computer telephony integration portsthat are not registered with
2010Cisco Systems, Inc. Troubleshooting Call Setup Issues 3-221
05) What are three options that perform digil-by-digit collection? (Choose three.) (Source:
Troubleshooting On-Premises Single-Site Calling Issues)
A) ISDN SETUP
B) Cisco Unified IP phones using SCCP
C) Cisco Unified IP phones using SIP dial rules
D) Cisco Unitied IP phones using SIP KPML
E) Cisco Unified IP phones using overlap sending and receiving
F) overlap sending and receiving on trunks or gateways
(i) 11.245 on trunks or gateways
Q6) Entities that are in route partition <None> are always accessible, regardless of whether
thc calling entity has a CSS. (Source: Troubleshooting On-Premises Single-Site Calling
Issues)
A) true
B) false
Q7) Given thc following trace output, what is the list of partitions that the caller 2002 can
reach? (Source: Troubleshooting On-Premises Single-Site Calling Issues)
CCM|Digit analysis: match(pi="2",fqcn="4 085 552 00 2", cn="2 002",
plv="5", pss= riBR_managers_pt :HQ_managers_pt: ITSP_pt" ,
TodFilteredPss="BR_managers_pt:HQ_managers_pt ",
dd="2001",dac="0")
A) BR managers pt
B) BRmanagerspt. HQmanagers pt, ITSPpt
C) ITSP_pt
D) BR_managers_pt:llQ managerspt
Q8) Thc following problem has been reported. Phone Acan initiate calls to phone B, and
the audio works in both directions; however, phone B cannot initiate calls to phone A,
What is the most likely possibility to consider? (Source: Troubleshooting On-Premise.s
Single-Site Calling Issues)
A) This issue typically occurs when cRTP is enabled on the serial line interface at
thc phone A side, but it is disabled on the serial line interface at thc phone B
side.
B) This issue typically occurs when the CSS of phone B includes the partition of
the phone A line, but thc CSS of phone Adocs not include the partition of thc
phone B line.
C) This issue typically occurs when the CSS of phone A includes thc partition of
thc phone B line, but the CSS of phone B docs not include the partition of the
phone A line.
Q9) What are threereasons whyan on-premises call can fail to be forwarded? (Choose
three.) (Source: Troubleshooting On-Premises Single-Site Calling Issues)
A) The CSS on thc IP phone that has originated the call docs not have the partit
of the forwardingdestinationdirectory number included.
B) None of route patterns that are configured matches with the call.
C) The specified destination is invalid.
D) The specified destination is busy.
L) The specified destination is not registered.
F) Cisco Unified Communications Manager servers that register the calling and
the called IP phones cannot talk to each other.
ion
3-222 Troubleshooting CiscoUnified Communicalions (TVOICE) vS0 2010CiscoSystems. Inc
mm
-"W
QIO) Thefollowing problem has been reported. Phone Acalls phone B. Phone Bdoes not
answer. Thecall is supposed tobeforwarded to voice mail but, instead, a busy signal is
heard. What are the three most likely possibilities to consider? (Choose three.) (Source:
Troubleshooting On-Premises Single-Site Calling Issues)
A) There is a codecmismatch between phoneAand the voice-mail server.
B) TheCSSon theCiscoIPphone configuration is lacking thepartition of thc
voice-mail pilot number.
C) The CSSon the CiscoIPphoneconfiguration is lackingthe partitionof the
voice-mail pattern.
D) The specified destination CFURis invalid or unspecified.
E) The specified destination CFNAis invalidor unspecified.
F) The voice-mail server or ports are currently not registered or all voice-mail
ports are currently in use.
Qll) How does Cisco Unified Communications Manager behave when 0112 isdialed and
the following dial planis used? (Source: Troubleshooting On-Net Multisite Calling
Issues)
Route Pattern 1: 0.112
Route Pattern 2: 0.1XXXXXXXXX
A) Route pattern 1 is used to route the call.
B) Route pattern 2 is used to route the call.
C) Cisco Unified Communications Manager triggers an interdigit timeout.
D) A reorder tone is played to the caller.
Q12) Which three called-number digit manipulationoptions exist at inbound when setting up
the call from one cluster to another cluster via an ICT trunk? (Choose three.) (Source:
Troubleshooting On-Net Multisite Calling Issues)
A) discard digits instructions
B) incoming called-party number setting
C) called-party transformation
D) external number mask
E) significant digits
F) route pattern
G) prefix
Q13) Given the following traceoutput for an intercluster call, whichtwostatements are
correct'? (Choose two.) (Source: Troubleshooting On-Net Multisite Calling Issues)
StationD: (0000019) startMediaTransmission
conference!D=20476773 passThruPartyID=16777237
remoteIpAddress=IpAddr.type:0
ipAddr:0x0a020211000000000000000000000000(10.2.2.17)
remotePortNumber=2 34 02 milliSecondPacketSize=2 0
compressType=6(Media__Payload_G72 2_64k) RFC2833PayloadType=101
qualifierOut^?. myIP: IpAddr.type:0
ipv4Addr:0xOa010211 (10.1.2.17) j1, 100,56,1.22*10.2.1.1**
A) Indicates thestart of themedia transmission from thecalling phone 10.2.2.1 7
to the target phone 10.2.1.1.
B) indicates thestartof themedia transmission from thccalling phone 10.2.2.17
to the target phone 10.1.2.17.
C) Indicates the start of the media transmission fromthe callingphone 10.1.2.17
to the target phone 10.2,2.17.
D) I he conference bridge with the ID 20476773 is involved in the call.
E) The remote RTP port is 23402, and the selected codec is G.722
QI4) Gatekeepers have mandatory' and optional responsibilities. Match the mandatory
responsibilities with the tasks. (Source: Troubleshooting On-Net Multisite Calling
Issues!
A) address translation
B) admission control
C) bandwidth control
D) zone management
1. The gatekeeper manages endpoint bandwidthrequirements. When
registering with a gatekeeper, an endpoint will specify its preferred codec.
During11.245 negotiation, a different codecmightbe required. These RAS
messages arc used to control this codec negotiation.
2. A gatekeeper is required to provide address translation, admission control,
and bandwidth control for terminals, gateways, and multipoint control
units that are located within its zone of control.
3. Calls originating within an 11.323 network can use an alias to address the
destination terminal. Calls originating outside the 11.323 network and
received by a gateway can use an E.164 telephone number to address the
destination terminal. The gatekeeper must be able to translate the alias or
the F. 164 telephone number into the network address for the destination
terminal.
4. The gatekeeper can control the admission of thc endpoints into the 11.323
network. It uses these RAS messages to achieve the ARQ, ACT, and ARJ.
3-224 TroubleshootingCisco Unified Communications (TVOICEl v8.0 2010 Cisco Systems. Inc
1
Q15) The H.323 gatekeeper is commonly used to perform CAC between multiple Cisco
Unified Communications Manager clusters. Match the common codecs with the
amount of pcr-call bandwidth that a gatekeeper considers when thc call is being set up.
(Source: Troubleshooting On-Net Multisite Calling Issues)
A) 32 kb/s
B) 128 kb/s
C) 30 kb/s
D) 128 kb/s
E) 16 kb/s
I. G.711
2. G.729
3. G.722
4. G.728
5. iLBC
Q16) Cisco Unified Communications Manager is required to define a unified
communications SIP trunk to the Cisco Unified Border Element. When is an MTP
required for a SIP trunk that connects Cisco Unified Communications Manager and
Cisco Unified Border Element? (Source: Troubleshooting On-Net Multisite Calling
Issues)
A) if delayed offer or invite with no SDP is acceptable for trunk outbound calls
B) if delayed offer or invite with no SDP is acceptable for trunk inbound calls
C) if early offer or invite with SDP is a requirement for trunk inbound calls
D) if early offer or invite with SDP is a requirement for trunk outbound calls
Q17) Which two statements about the issue of immediate call drops are true? (Choose two.)
(Source: Troubleshooting On-Net Multisite Calling Issues)
A) If a call to a phone at the remote site immediately drops after the remote user
picks up the handset, it is a dial plan problem.
B) If a call to a phone at the remote site immediately drops after the remote user
picks up the handset, it is not a dial plan problem.
C) You can resolve the issue of an immediate remote call drop by modifying CSS
and partitions at the calling phone.
D) You can resolve the issue of an immediate remote call drop by adding an MTP.
E) A codec mismatch is the most common problem with a call that sets up but
then drops right away.
Q18) Whichcommand verifies that the MGCP gateway is registered and pointing to the
correct Cisco Unified Communications Manager server? (Source: Troubleshooting Off-
Net Calling Issues)
A) show mgcp connection
B) show mgcp endpoint
C) show mgcp state
D) show ccm-manager
2010Cisco Systems, Inc Troubleshooting Call SetupIssues 3-225
Q19) Which two of the following would you typically verify when troubleshooting inbound
call issues with a Cisco IOS H.323 gateway? (Choose two.) (Source: Troubleshooting
Off-Net Calling Issues)
A) If the H.323 gateway is set for early offer.
B) 1hat the commands under the dial peer are correct for Cisco Unitied
Communications Manager redundancy.
C) That the voice class H.225 timeout for TCP is set to 3 seconds or more.
D) That the preference command has been used to determine the Cisco Unified
Communications Manager server order. The higher thc number, the more
preferred the server is,
E) That the H.323 binding is configured.
Q20) Identify the inbound dial-peer matching priority for digital POTS or VoIP dial peers.
(Source: Troubleshooting Off-Net Calling Issues)
A) answer-address
B) destination-pattern
C) incoming called-number
D) voice-port
1. First priority
1. Second priority
2. Third priority
3. Fourth priority
Q21) When the called number matches a directory number of the phone, no further called-
party transformation is possible. (Source: TroubleshootingOff-Net Calling Issues)
A) true
B) false
Q22) When the called number matches a translation pattern, hunt pilot, or route pattern, you
can configure a set of digit-manipulation methods for both the called- and thc calling-
party numbers. What are three options that exist for calling-party transformations?
(Choose three.) (Source: Troubleshooting Off-Net Calling Issues)
A) discard digit instruction
B) external phone number mask
C) numbering mode
D) significant digits
E) prefix digits
F) translation pattem
G) transformation mask
Q23) If any of thcavailable called-party transformation methods is configured, thenthe
complete set of called-party transformationmethods that are configured at the route
pattern is ignored and thc complete called-party transformationconfigurationof the
selected route group is applied. (Source: Troubleshooting Off-Net Calling Issues)
A) true
B) false
3-226 Troubleshooting CiscoUnified Communications (TVOICE) v80 2010CiscoSystems Inc
m*
5m
mm
Q24)
Q25)
Q26)
What arc three DDIs that can be used if a route pattern does not contain the (a); sign?
(Choose three.) (Source: Troubleshooting Off-Net Calling Issues)
A) PreDot
B)
PreDot-Leading-#
C) PreDot-Trailing-#
D) PreDot-InttTollBypass
E) PreDot-10-10-Dialing
F) NoDigits
G) PrePlus
Which are the three main characteristics of a local route group? (Choose three.)
(Source: Troubleshooting Off-Net Calling Issues)
A) Local route groups decouple the selection of the egress device fromthe route
patterns that are used to access the gateway.
B) Standard Local Route Group is a new entry in the list of route groups that can
be added to a device pool.
C) Calls that are routed by route patterns that use local route groups will route thc
call to thc same gateway, no matter what device originated the call.
D) With local route groups, the egress device (gateway or trunk) is selected based
on thc matched route pattem.
E) Local routegroups cangreatlyreducethe complexity of dial plans in Cisco
Unified Communications Manager, but its size will slightly increase.
F) Local route groups can greatly reduce the complexity and size of dial plans in
G) Local route groups have been introduced with Cisco Unified Communications
Manager Version 6.0.
A PSTN that is called dials the DID for an IP phone. Ringbaek is heard and the IP
phone user answers the phone. Immediately, both parties that are involved in the call
hear dead air. What are three possible causes of this problem? (Choose three.) (Source:
Troubleshooting Off-Net Calling Issues)
A) PSTN switch that cannot manage the call
B) codec mismatch
C) RTP port number mismatch
D) voice packets being dropped because of insufficient resources
E) ACLs blocking RTP streams
F) ACLs blocking voice signaling
G) proper fixup on the firewalls not enabled
Q27) What are threerequirements for numbernormalization whenan ingress call comes to a
gateway? (Choose three.) (Source: TroubleshootingOff-Net Calling Issues)
A) changing the calling number from localized L.I64 fomiat to global E.164
format
B) changing the calling number from global E.164 format to local E.164 format
C) changing thc callednumberfromglobal F..164 format to directory numbers for
calls to internal destinations
D) changing the callednumberfrom localized li.164 format to directory numbers
for calls to internal destinations
E) changing both calling and called numbers to directory numbers
F) changing the called number from globalized E.164 format to localized E.164
format for calls to external destinations
G) changing the called number from localized E.164 format to global E.164
format for calls to external destinations
Q28) Which two of the following would you verify if, for an outbound call, a called PSTN
pany sees an incorrect calling number? (Choose two.) (Source: TroubleshootingOff-
Net Calling Issues)
A) Calling number localization: Verify by using calling-party transformationCSS
at the route pattern or route list. Also check that thc calling number
transformationpatterns arc reachable to the gateway by using CSS and
partition,
B) Calling number localization: Verify by using calling-party transformationCSS
at the gateway or device pool. Also check that the calling number
transformation patterns are reachable to the gateway by using CSS and
partition.
C) Calling number localization at thc phone: Verify that the external phone
number mask is correct and is reused by other dial plan components.
D) Calling number globalization at the gateway: Verify that the external phone
E) Calling number globalization at the phone: Verify that thc external phone
Troubleshooting Cisco Unified Communications (TVOICE) v8.0 2010 Cisco Systems. Inc
^
QD
C
Q2) F
Q3) A. 0. D. G
04) B. D. R.H
Q5) B.D, F
Q6) A
Q7> D
QS) C
Og>
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QIO) B.E. F
Qll) C
Q12) B. F. d
Q13) C. R
014) 3-A.4-B. 1-C.2-D
Q15) 1-B.2-E. 3-D. 4-A. 5-C
Q16) D
QI71 B.K
Q18) D
Q19> B. L
Q20) 1-C.2-A. 3-B.4-D
02!) A
022) B, 11. G
023) A
Q24) A. C. F
025) A. D. F
026) B.F.G
Q27| A. D. G
Q2S) B.F
2010 Cisco Systems, Inc Troubleshooling Call Setup Issues 3-229
3-230 Troubleshooting Cisco UnifiedCommunications (TVOICE) v8.0 2010 Cisco Systems. Inc
Table of Contents
Volume 2
SAF and CCD Issues 4-1
Overview 4-1
Troubleshootinq SAF 4-3
Objectives 4-3
Service Advertisement Framework 4-4
SAF Client Types 4-6
SAF Methods 4-8
SAF client Protocol Basic Flow 4-10
SAF Routing Characteristics 4-11
SAF Components Configuration Review 4-13
Troubleshooting SAF 4-15
SAF Client Statistics in Cisco Unified RTMT 4-16
Troubleshooting SAF Forwarder Adjacency Issues 4-17
Troubleshooting SAF External Client Interactions 4-21
SAF External Client Integrity Failure 4-24
Tracing Registration to a SAF Forwarder 4-26
Troubleshooting SAF Information Exchange 4-28
Summary 4-33
References 4-33
Troubleshooting CCD 4-35
Objectives 4-35
Call Control Discovery 4-36
CCD Operation During IP Connectivity Failure Review 4-38
CCD Configuration Elements on SAF External Client 4-39
Relationship of Configuration Elements on the External SAF Client 4-41
CCD Configuration Elements on the Internal SAF Client 4-42
Relationship of Configuration Elements on the Internal SAF Client 4-43
Troubleshooting the SAF Client in CCD 4-45
Verifying Learned Patterns at the Internal Cisco IOS Client 4-46
Troubleshooting Patterns Learning Process 4-47
Verifying Advertising and Requesting Services 4-49
Tracing the CCD Advertising Patterns Process 4-51
Tracing the CCD Patterns Requesting Process 4-52
Tracing CCD Pattern Blocking 4-53
Dialing to Learned Patterns Fails 4-54
Troubleshooting an Incorrect Learned Pattern Prefix 4-55
Troubleshooting CCD PSTN Failover 4-58
CCD Feature Parameters for the PSTN Failover 4-60
Ensure that Correct PSTN Number Is Made for CCD PSTN Failover 4-62
Tracing a Successful PSTN Failover Call 4-63
Troubleshooting an Incorrect PSTN Prefix 4-67
CCD PSTN Failover Unconditionally Applies 4-68
Cisco Unified SRST CCD Considerations 4-69
Summary 4-70
References 4-70
Module Summary 4-71
References 4-71
Troubleshooting Cisco Unified Communications Manager Features and Application
Issues 5^1
Overview 5-1
Troubleshootinq Device Mobility Issues 5;3
Objectives 5-3
Device Mobility and Its General Issues 5-4
Dynamic Phone-Configuration Parameters 5-5
Device Mobility Configuration Elements 5-7
Relationship of Device Mobility Configuration Elements 5-8
When and How Phone Configuration Is Modified 5-10
Device Mobility Considerations 5-12
Device Mobility and CSSs 5-14
Device Mobility Example with Different Device Mobility Groups 5-16
Device Mobility Example with Same Device Mobility Group 5-17
Interaction of Local Route Groups and Device Mobility 5-18
Device Mobility Example with Local Route Groups 5-19
Device Mobility General Problems 5-20
Troubleshooting IP Infrastructure Problems 5-21
Troubleshooting Device Mobility Configuration Mismatches 5-24
Troubleshooting Device Mobility Call-Routing Problems 5-26
Troubleshooting Device Mobility Call-Routing Problems Summary 5-32
Troubleshooting Device Mobility Call Privilege Problems 5-33
Summary 5-36
References 5-36
Troubleshootinq Cisco Extension Mobility Issues 5-37
Objectives 5-37
Cisco Extension Mobility and Its General Issues 5-38
Cisco Extension Mobility Configuration Elements Review 5-40
Relationship of Cisco Extension Mobility Configuration Elements 5-42
How Does Cisco Extension Mobility Manage Phone Model Differences? 5-43
Cisco Extension Mobility General Issues 5-45
Troubleshooting Cisco Extension Mobility Error Messages and Login and Logout Issues 5-46
Troubleshooting Cisco Extension Mobility Login Problems 5-49
Troubleshooting Cisco Extension Mobility Logout Problems 5-51
Troubleshooting Cisco Extension Mobility Phone Button Problems 5-52
Troubleshooting Cisco Extension Mobility Call Privilege Problems 5-53
Cisco Extension Mobility and CSSs 5-54
CSSs Example: Device CSS Only 5-55
CSSs Example: Line CSS Only 5-56
CSSs Example: Line CSS with Local Route Group 5-57
CSSs Example: Line or Device CSS 5-58
Troubleshooting Cisco Extension Mobility Call-Routing Problems 5-59
Summary 5-60
References 5-60
Troubleshooting Cisco Unified Mobility Issues 5-61
Objectives 5-61
Cisco Unified Mobility and Its General Issues 5-62
Mobile ConnectInternal Calls Placed from Remote Phone 5-64
Mobile Voice Access 5-65
Cisco Unified MobilityConfiguration Elements 5-66
Shared Line Between Phone and Remote Destination Profile 5-68
Relationship of Cisco Unified MobilityConfiguration Elements 5-69
CSS Handling Since Cisco Unified Communications Manager 7.x 5-70
Access Lists Since Cisco Unified Communications Manager 7.0 5-71
Operation of ToD Access Control 5-72
Enterprise Feature AccessDTMF Directed Call Park 5-73
TroubleshootingCisco Unified Communications(TVOICE) v8 0 2010 Cisco Systems, Inc
DTMF-Directed Call Park Considerations 5-75
Enterprise Feature AccessDusting Feature 5-76
Cisco Unified Mobility General Issues 5-78
Troubleshooting Cisco Unified Mobility Mobile Connect 5-79
Troubleshooting Inbound Internal and PSTN Calling Problems 5-80
Tracing Inbound Call to Mobile Connect 5-85
Troubleshooting Remote-Phone Calling Problems 5-89
Troubleshooting Problems with Call Redirection to Remote Phone 5-91
Troubleshooting Cisco Unified Mobility Mobile Voice Access 5-93
Troubleshooting Mobile Voice Access Outgoing Call Problems 5-94
Troubleshooting Enterprise Feature Access and Dusting Feature Problems 5-95
Troubleshooting Dusting Feature Problems 5-97
Summary 5-98
References 5-98
Troubleshooting Cisco Unified Communications Manager Native Presence Issues 5-99
Objectives 5-99
Native Cisco Unified Communications Manager Presence General Issues 5-100
Native Cisco Unified Communications Manager Presence Configuration Review 5-101
Cisco Unified Communications Manager Call History Presence 5-102
Cisco IP Phones that Support Viewing Presence Status 5-103
Native Cisco Unified Communications Manager Presence General Issues 5-104
Troubleshooting Line Presence Indications 5-105
Troubleshooting Line Presence Indications with Subscribe CSS 5-107
Presence Policies Using Presence Groups 5-108
Troubleshooting Line Presence Indications with Presence Groups 5-110
Troubleshooting Trunk Presence Indications 5-111
Verify Presence Policies Are Set on SIP Trunks 5-112
Troubleshooting Historical Presence Indications 5-113
Summary 5-114
References 5-114
References 5-115
Voice Quality and Media Resources Issues 6-1
Overview 6-1
Troubleshooting MOH Issues 6-3
Objectives 6-3
MOH Review 6-4
Unicast MOH 6-5
Multicast MOH 6-6
MOH Sources 6-7
MOH Audio Source and Media Resource Selection 6-8
MOH Service Parameters 6-9
MOH Operation 6-10
Common Issues of MOH 6-11
Typical Issue: Calls Disconnected When Placed on Hold 6-12
MOH Performance 6-13
MOH Device Flapping Issue 6-14
Troubleshooting MOH Registration and Nonresponsive Software Issues 6-15
Tuning MOH Loudness 6-17
TOH Instead of MOH 6-18
Verify MOH Configuration 6-19
TOH to PSTN Caller 6-21
Troubleshooting Multicast MOH 6-22
Troubleshooting IP Multicast Routing 6-23
Multicast MOH Trace 6-25
2010 Cisco Systems, Inc. Troubleshooting Cisco Unified Communications (TVOICE) v8.0 iii
Troubleshooting Multicast MOH from Branch Router Flash 6-27
Considerations When Using Multicast MOH from Branch Router Flash 6-28
Multicast MOHfrom Branch Router Flash Example 6-29
Multicast MOH from Branch Router Flash Operation 6-31
Troubleshooting Multicast MOH from Branch Router 6-33
Summary 6-34
References 6-35
Troubleshootinq MTP Issues 6-37
Objectives 6-37
MTP Review 6-38
MTP for NTEs 6-40
MTP Requirement Review 6-41
General MTP Issues 6-42
Troubleshooting MTP Registration and Nonresponsive Software Issues 6-43
MTP Registration Issues 6-46
MTP Allocation 6-47
MTP Allocation Failure Decision Tree 6-48
MTP Allocation Issues 6-49
Verify MTP Utilization 6-50
MTP Allocation Trace 6-52
MTP Allocation Debug 6-56
Hardware MTP Allocated 6-58
Summary 6-59
References 6-59
Troubleshootinq Issues with Conferences 6-61
Objectives 6-61
Conferencing in Cisco Unified Communications 6-62
Meet Me and Ad Hoc Conferencing 6-63
Linear Ad Hoc Conferencing 6-64
Nonlinear Ad Hoc Conferencing 6-65
Hardware and Software Conference Media Resource 6-66
General Issues Related to Conference Bridges 6-67
Troubleshooting Conference Bridge Registration and Nonresponsive Software Issues 6-68
Troubleshooting Ad Hoc Conferencing 6-70
Ad Hoc Conferencing Issues 6-72
Verify Conference Bridge Performance 6-76
Tracing Ad Hoc Conference Setup 6-77
Troubleshooting Meet-Me Conferencing 6-83
Meet-Me Conferencing Issues 6-85
Tracing a Meet-Me Conference Setup 6-88
Meet-Me Conference Setup Completed 6-92
Tracing an Unsuccessful Meet-Me Conference Setup 6-93
Summary 6-94
References 6-94
Troubleshootinq Transcoder Issues 6-95
Objectives 6-95
Transcoder Review 6-96
Codec Configuration and Selection Review 6-97
Transcoder Decision Tree 6-98
General Transcoder Issues 6-99
Troubleshooting Transcoder Registration Issues 6-100
Transcoder Registration Issues 6-103
Troubleshooting Transcoder Allocation 6-105
Verify Transcoder Utilization 6-106
Transcoder Allocation Trace 6-108
Transcoder Allocation Debug 6-113
Transcoder Allocated 6-115
Summary 6-116
References 6-116
Troubleshooting Issues with RSVP Agents 6-117
Objectives 6-117
RSVP CAC 6-118
Three Call Legs with RSVP-Enabled Locations 6-119
Characteristics of Phone-to-RSVP Agent Call Legs 6-120
Agent-to-RSVP Agent Call Leg 6-121
RSVP Agent Operation 6-122
General RSVP CAC Issues 6-124
Troubleshooting the RSVP Agent Registration 6-125
Troubleshooting the RSVP CAC Operation 6-130
RSVP CAC Operation Issues 6-132
Verify RSVP CAC Performance 6-135
RSVP CAC Debug 6-136
RSVP Reservation Installed 6-140
Troubleshooting Intercluster RSVP with SIP Preconditions 6-141
Issues of Intercluster RSVP with SIP Preconditions 6-144
Tracing Successful Call Setup with SIP Preconditions 6-145
Tracing Unsuccessful Call Setup with SIP Preconditions 6-151
Tracing Call Setup with Fallback to Local RSVP 6-153
Summary 6-155
References 6-155
Troubleshooting Voice Quality Issues 6-157
Objectives 6-157
Voice Quality Issues in Cisco Unified Communications Systems 6-158
Lack of Bandwidth 6-160
End-to-End Delay 6-162
Jitter 6-163
Echo 6-164
QoS Requirements and QoS Policy 6-165
Typical QoS Policy Definition 6-167
QoS Policy Implementation Options 6-168
Identifying and Isolating Voice Quality Problems 6-170
Cisco IP Phone RTP Statistics 6-175
Troubleshooting Layer 2 Quality Problems 6-176
Layer 2 Voice Quality Considerations 6-177
Queuing and Scheduling on Cisco Catalyst 3750 Series Switches 6-178
WTD on Catalyst 3570 Series Switches 6-180
Verify Ingress Classification 6-181
Verify Egress Classification 6-183
Verify Mapping of Traffic to Egress Queues 6-184
Monitor Packet Drops 6-185
Troubleshooting Voice Quality Issues on a Gateway 6-186
Monitor Interface Load and Congestion 6-188
Monitor Congestion at the Policy Map 6-189
Verify cRTP and LFI 6-190
Sample Troubleshooting Scenarios 6-193
Delay Budget Calculation 6-195
Synthetic Voice 6-196
Identifying Issues with Jitter 6-198
Choppy Voice 6-200
Echo 6-202
Locating and Eliminating Echo 6-203
Identifying a Type of Echo 6-205
Eliminating the Echo 6-207
One-Way Audio Issue 6-209
Policy Map Attached to One End of the Link Only 6-211
Summary 6-212
References 6-213
12010 Cisco Systems. Inc. Troubleshooting CiscoUnified Communications (TVOICE) v8.0 v
Module Summary
References
Module Self-Check
Troubleshooting Cisco UnifiedCommunications (TVOICE) vB0
6-215
6-216
6-217
6-223
Module 4
SAF and CCD Issues
^ Overview
Cisco Service Advertisement Framework(SAF) provides a framework that allows applications
to discover the existence, IP address, port, and configuration of networked resources within
networks. Cisco SAF allows a timely and reliable awareness of the services within networks, as
applications advertise and discover services on networks. Service infonnation distributes
though a network of CiscoSAFcooperative nodes that assumespecific functions to efficiently
distribute knowledge of services and facilitate their discovery.
The Call Control Discovery (CCD) feature leverages the SAFnetworkservice, a proprietary
Cisco service, to facilitate dynamic provisioning of intercall agent information. Byadopting the
SAFnetwork service, theCCDfeature allowsCiscoUnifiedCommunications Managerto
advertise itself along with other key attributes such as directory number patterns that are
configured in Cisco UnifiedCommunications Manager Administration. As a result, other call
control entities that also use the SAF network can use the advertised information to
dynamically configure and adapt their routing behaviors.
Module Objectives
Upon completing this module, you will be able to solve the common issues of an SAF-enabled
network and CCD. Thisabilityincludes beingable to meet these objectives:
Explain the common issues that relate to the SAF Client and SAF Forwarder in an
environment with CCDand identifythe most likely causes of these issues
Explain the common issues that relate to the CCD as an application of SAF and identify the
most likely causes of these issues
Lesson 1
Troubleshooting SAF
L Overview
Today, system configuration between application clients and servers is primarily done
statically. Static configurationcreates deployment and reconfiguration barriers in any typical
medium to large-scale enterprise application. Typically, when a service is established and all
the clients and servers are staticallyconfigured, the effort and cost to move, add, change, or
delete any portion of the system create barriers to the business.
The Service Advertisement Framework(SAF) is a dynamic communications frameworkfor
network applications that allows servers and clients to advertise and discover services. The
network-based SAF application propagates information inthesame waythatrouting propagates
information andthusallows customers greater scale, availability, andadaptability todeploy and
manage applications across the enterprise.
Objectives
Upon completing this lesson, you will be able to explain the common issues that relate to the
SAFclient and the SAFforwarder in an environment withCall Control Discovery (CCD)and
toidentify themost likely causes of these issues. This ability includes being able tomeetthese
objectives:
DescribeSAFand thecomponents that it comprises and explainthe roles of the SAFclient
and the SAF forwarder
Describe the common issues that relate to the SAF client and the SAF forwarder and
explain their potential causes
Service Advertisement Framework
This topic describes SAF and its components. It also explains the roles of the SAF client and
the SAF forwarder.
CCD-enaoied call agenls
advertise to and learn from
the network
SAF is used to distribute
information within the
network
SAF forwarders interact
with CCD-enabled call
agents (SAF clients)
SAF forwarder learns
information from Ihe
SAF client
SAF forwarders
distribute information
among each other
SAF forwarder
advertises all learned
information to the SAF
client
Call Agent
Call Age "IN,
Call ftgen
Call Agent
Call Agtin!
|j|| CallAger.l
\\\
SAF-Enabled
IP Network
Call Agent
Call Agent
^1
CaJIAgeril
^1
IIAgsril
The CCD feature leverages thc SAF network service, a proprietary Cisco service, to facilitate
dynamic provisioning of intercall agent information. By adopting the SAF network service, the
CCD feature allows Cisco Unified Communications Manager to advertise itself along with
other key attributes. The attributes, for example, are directory number patterns that arc
configured in Cisco Unified Communications Manager Administrationso that other call control
entities that also use the SAF network can use the advertised information to dynamically
configure and adapt their routing behaviors.
SAF provides a framework that allows applications to discover the existence, IP address, port,
and configuration of networked resources within networks. SAF allows a timely and reliable
awareness of the services within networks as applications advertise and discover services on
networks. Service information distributes though a network of SAF cooperative nodes that
assume specific functions to efficiently distribute knowledge of services and facilitate their
disc oven,'.
A non-SAF node is any node in a network that does not understand SAF. Non-SAF nodes are
called "dark nets" and are required to traverse ISPs. Thc SAF messages are IP-based and,
therefore, are unaffected by dark nets.
In theory, any sen ice can be advertised through SAF. The first service to use SAF is CCD.
CCD uses SAF to distribute and maintain information about the availability of internal
directory numbers that arc hosted by call control agents such as Cisco UnifiedCommunications
Manager and Cisco UnifiedCommunications Manager Express. CCDalso distributes the
corresponding number prefixes that allow these internal directory numbers to be reached from
the public switched telephone network (PSTN) ToDID prefixes.
Troubleshooting Cisco Unified Communications (TVOICEl v8 0 ) 2010 Cisco Systems, Inc
The dynamic nature of SAF and the ability of call agents to advertise the availability of their
hosted directory number ranges and ToDID prefixes to other call agents in a SAF network
provide distinct advantages over other static and more labor-intensive methods of dial plan
distribution.
These SAF-cooperative network nodes are grouped into two major functional responsibilities:
Thc SAF forwarder: A SAF forwarder receives services that arc advertised by SAF
clients, distributes thc services reliably through the network, and makes services available
for SAF clients to use. Here are some SAF forwarder features:
Ensures reliable delivery of service advertisements
Maintains knowledge of path redundancy
Is scalable
Minimizes the use of network bandwidth by using targeted multicast and unicast
messages
The SAF forwarder can propagate service advertisements to other SAF forwarders across a
LAN, campus network, WAN, or ISP.
SAF client: A SAF client is a producer (advertises to the network) or consumer of services
(requests a service from the network), or both. When a SAF client sends a register message
to a SAF forwarder, it establishes a relationship with the SAF forwarder. The SAF
forwarder uses this register message to obtain a unique handle that distinctly identifies this
SAF client from others that are connected to it. Only after a SAF client registers is it able to
advertise (publish) to or request (subscribe) services.
2010 Cisco Systems, Inc. SAF and CCD Issues 4-5
SAF Client Types
This section reviews the twotypes of SAFclients and the protocols that are usedwithinthe
SAF network.
SAF forwarder is always a Cisco IOS
device
Two types of SAF clients'
Exiemal SAF clients.
SAF client and SAF forwarder
are different devices
SAF client is Cisco Unified
' SAF-CPisused
Irtemal SAF clients.
< SAF client and SAF forwarder
are coiocated functions within
Ihe same device
SAF client is Cisco Unified
Express, Cisco Unified SRST,
or Cisco Unified Border
Element
Internal API is used
Cisco Unified
Communicator
Manager
turf
Abasic SAF forwarder provides the relationship between SAF clients and thc framework. A
SAF forwarder is normally located at the edges or boundaries of a network. Thc SAF forwarder
receives sen ice advertisements and stores a copy before forwardingthe advertisement to its
neighbor SAF nodes. The SAF client and the SAF forwarder relationship is to maintain the
advertisement. If a SAF client removes a service or disconnects from the SAF forwarder node,
the node will inform the framework about the services that arc no longer available. When the
SAF forwarder node receives advertisements from other SAF forwarder nodes, il will keep a
copy of the entire advertisement (header and opaque data) and forward it to the other SAh
pecrs.
You can configure a SAF forwarder on a LAN to automatically allow dynamic discover,' of
services to all enabled interfaces, and at the same time, specify interfaces (static configuration)
that you want blocked to other interfaces that attempt to discover their services.
Two general types of SAF clients can interact with the SAF forwarder:
The external SAF client is implemented apart from the SAF forwarder on a separate
platform. Currently, thc only implemented external SAF client is the Cisco Unitied
Communications Manager cluster. Between the SAF forwarder and the external SAF client,
the SAF control protocol is used. SAF currently supports a maximum of 50 external SAF
clients.
Thc internal SAF client is coiocated with a SAF forwarder on the same Cisco IOS platform.
These are typical internal SAF clients:
Cisco Unified Communications Manager Express on a Cisco Integrated Services
Router (ISR)
Cisco Unified Border Element on Cisco ISR
Cisco Unified Sur\ivable Remote Site Telephony (SRST) on Cisco ISR
Because both SAF network components exist on the same system, internal application
programming interface (API) isused.
SAF forwarders use SAF forwarding protocol (SAF-FP) topropagate advertisements that are
received from clients. SAF-FP defines metrics that areusedto avoid loops.
2010 Cisco Systems. Inc SAF and CCD lssues
SAF Methods
TheSAF client maintains theconnection tothc SAF forwarder.
4-8
SAF Methods
SAFclients advertise or request a service.
Each SAF client transaction consists ofa request
message followed bya response message.
Responses are of two types:
* Success responses
a Error responses
These SAF methods are defined:
Register, Unregister, RegisterRevoke
Publish/Notify, UnpublishAA/ithdraw, PublishRevoke
' Subscribe, Unsubscribe
Each SAF client performs two functions in a SAF network:
Advertising asenice: When advertising aservice, aSAF client publishes (sends)
ad\ertiscments to the SAF forwarder that contain information about the service that it
offers. The SAF client can send multiple publish requests, each advertising adistinct
sen-ice. The SAF forwarder advertises all services that are published by the SAF client.
Requesting aservice: When requesting aservice, the SAF client sends arequest
notification ofsen ices by using asubscribe request. The subscribe request contains afilter
that describes the set ofservices in which the SAF client is interested. In response to this
request, the SAF forwarder sends the current setof services that match the filter tothc SAF
client in aseries ofnotify requests. Multiple notify requests are sent to provide flow
control; thc SAF client must respond to each notify request before the SAF forwarder sends
the next request. As with apublish request, thc SAF client can generate multiple subscribe
requests, each with a different filter. The SAF client can also generate anunsubscribe
request, which removes oneof itsexisting subscriptions.
SAFdefines nvo message types:
Request messages
Response messages that can besuccess or error responses
In most situations, you would configure a SAF client application with oneor more IP addresses
andpons thatarc used toconnect toa SAF forwarder. ASAF client initiates a TCPconnection
to aSAF forwarder. When the TCP connection is established, aSAF client sends aregister
message tothe SAF forwarder. This register message uniquely identifies the SAF client from
all other SAF clients that are connected to the SAF forwarder.
When aSAF client registers, it advertises aservice by sending apublish request to aSAF
forwarder CCD is the first SAF service that can be advertised. SAF services are identified to a
network of SAF forwarders and clients by their SAF Service ID. CCD for Cisco Unified
Communications uses a SAF service IDof 101:2.x.x.x.x:
Service ID 101 = CiscoUnifiedCommunications
Subservice ID 2 = CCD
Instance ID x.x.x.x - ID of aCisco Unified Communications Manager cluster (primary key
ID[PKID]) or a CiscoIOSdevice
ASAF forwarder advertises all services that are published by aSAF client. Similarly, aSAF
client can request notification of services by using asubscribe request. The subscribe request
contains afilter that describes the set ofservices in which the SAF client is interested. In
response to this request, the SAF forwarder sends to the SAF client, in aseries of notify
requests, the current set of services that match the filter. Multiple notify requests are to provide
flow control. The SAF client must respond to each notify request before the SAF forwarder will
send the next request.
Like apublish request, the SAF client can generate multiple subscribe requests, each with a
different filter. The SAF client can also generate an unsubscribe request, which removes one of
its existing subscriptions.
>2010 Cisco Systems, Inc SAF and CCD Issues 4-9
SAF client Protocol Basic Flow
The figure shows the basic message flow between the SAF client and the SAF forwarder.
4-10
SAF CHent Prot
SAF Client
v - ^
SAF Forwarder
SAF-Enabled
. .IP Network
' '' ; . Service 1
-..;i,= Service 1
-C;,(.:.
I- .. . Service 2
A=t;r.- Service 2
?' - .V.J. -
Page 1
Page 2
Page 1
Page?
ASAF client registers uith the SAF forwarder. Thc SAF client publishes services by using a
publish request; in the case of CCD, that publish request comprises aset of hosted patterns A
SAF forwarder advertises all services that were published by the SAF client to the SAF-enabled
IP network,
Similarly, aSAF client can request notification of services by using asubscribe request The
subscribe request contains afilter that describes the set ofservices in which thc SAF client is
interested. In response to tins request, thc SAF forwarder sends to the SAF client in aseries of
notify requests, thc current set of services that match the filter. Multiple notify requests are sent
to provide fiow control. The SAF client must respond to each notify request before the SAF
forwarder u ill sendthe next request.
Like apublish request, the SAF client can generate multiple subscribe requests, each with a
different filter. The SAF client can also generate an unsubscribe request, which removes one of
its existing subscriptions.
SAF Routing Characteristics
The Cisco SAF is a Layer 4 communications framework that is built into Cisco IOS Software
that distributes service messages around theLAN andWAN much likerouting protocols
propagate routes.
SAF Routing Characteristics
SAF-FP leverages features and functions of EIGRP:
Split horizon
- Incremental updates (only when changes occur)
Bandwidth-percent
- Hello interval
- Hold time
- Authenticated updates
Independent of IP routing protocol
Static
Dynamic (EIGRP, OSPF, BGR and so on)
SAF leverages features of Enhanced Interior Gateway Routing Protocol (EIGRP) to distribute
service information within a SAF-enabled network.
SAF is a Service Advertisement Protocol (SAP) thatruns independently of anyIProuting
protocol. SAF uses EIGRP for service message transport enablement, but no EIGRP IProuting
needs to berunning in thenetwork. Any Interior Gateway Protocol (IGP)for example,
Routing Information Protocol (RIP), Interior Gateway Routing Protocol (IGRP), Open Shortest
Path First (OSPF), or Intermediate System-to-Intermediatc System (IS-IS) Protocol- canbe
used.
SAFhas to bedeployedonly near theedges of your network closeto the SAFclientsthat
advertise or subscribe tothe SAF messages. TheSAF forwarders musthave reachability toone
another as well as to route service advertisement messages.
SAFshares manyof thc features of EIGRP. LikeEIGRP, SAFuses reliable transport (EIGRP
uses IP protocol port number 88).
SAF prevents advertisement loops byusing the EIGRP Diffusing Update Algorithm (DUAL)
andsplithorizon. Split horizon controls thesending of SAFupdate andquery packets. When
split horizonis enabledon an interface, updateand querypacketsare not sent for destinations
for which thisinterface is thenext hop. Controlling update andquery packets in thismanner
reduces the possibility of routing loops. By default, split horizon is enabled on all interfaces.
SAF and CCD Issues
SAF leverages EIGRP features and functions, including these:
Split horizon: Split horizon blocks route information from being advertised by a router out
of any interface from which that information originated. This behavior usually optimizes
communications among multiple routing devices, particularly when links are broken.
However, with nonbroadcast networks, situations can arise for which this behavior is less
than ideal. For these situations, including networks in which you have EIGRP configured,
you may want to disable split hori/on.
Incremental updates: SAF sends incremental updates when the state of a destination
changes, instead of sending the entire contents of the routing table. This feature minimizes
the bandwidth that is required for SAF packets.
Bandnidth-percent: By default, SAF packets consume a maximum of 50 percent of the
link bandwidth, as configured with the bandwidth interface configuration command for
autonomous system configurations and with the bandwidth-percent command for named
configurations. Change that value if a different level of link usage is required or if the
configured bandwidth does not match the actual link bandwidth.
Hello: Use this simple SAF neighbor discovery mechanism to learn about neighboring
routers. You can adjust the interval between hello packets and the hold time.
By default, hello packets are sent every 5 seconds. Thc exception is on low-speed,
nonbroadcast multiaccess (NBMA) media, in which thc default hello interval is 60 seconds.
Low speed is a rate of Tl or slower, as specified with the bandwidth interface
configuration command. The default hello interval remains 5 seconds for high-speed
NBMA networks.
Hold time: You can configure the hold time on a specified interface for a particular
EIGRP SAF routing process that is designated by the autonomous systemnumber. The
hold time is advertised in hello packets and indicates to neighbors the length of time that
they should consider the sender valid. The default hold time is three times the hello
interval, or 15 seconds. For slow-speed NBMA networks, the default hold time is 1SO
seconds.
Hops: With EIGRP SAF, the largest possible network width is 100hops, which is the
maximum number of hops.
Authenticated updates: HIGRP SAF route authentication provides Message Digest 5
(MD5) authentication of routing updates from thc EIGRP routing protocol. The MD5
keyed digest in each EIGRPpacket prevents the introductionof unauthorizedor false
routing messages from unapproved sources.
Currently, SAF does not use the metrics that are used by EIGRP (bandwidth, delay, delay
reliability, load, and MTU). "Ihesein-depth enhancements will be supportedwith later releases
of SAF.'
Troubleshooting CiscoUnified Communications (TVOICEl v80 2010CiscoSyslems, Inc
SAF Components Configuration Review
The figure shows the typical basic configurationof the SAF forwarder.
SAF Components Configurati
SAF Forwarder
interface LoopbackO
ip addreas 10.2.250.101 255.255.255.255
router eigrp aaf
I
service-family ipvl autonomous-system 1
External SAF Client
MP *mir*T *ntmm
04*tru>tjim
<.fttMf M'J-?
topology base
eKternal-cllant Cucm2
exit-sf -topology-
exit- service-family
I
service-family extarnal-client listen
lpv4 5050
external-client cucmS
username cucm2
pass-ord ciscol23456
f rocfi4tltr m'n ~
The router eigrp instance command enables an EIGRP virtual instance on a router. The
service-family ipv4autonomous-system command enables a Cisco SAF service family for the
specified autonomous system on the router.
The topology basecommand enables service-family interface topology configuration mode
and creates a topology base for the specified interface on the router. The external-client client-
labelcommand configures a CiscoSAFexternal client withthe specified client label.
To configure external SAF client-specific information, use these commands:
The username command enables external-client label configuration mode and configures
an external SAF client with the specified username.
Thc password commandconfigures a password for an external SAF client. The minimum
password length is 11 characters.
You can specifya keepalive timer for the external SAFclient by usingthe keepalive
command.
The same authentication parameters must be configured at the SAFexternal clientthe Cisco
Unified Communications Manager. InCisco Unified Communications Manager
Administration, choose Advanced Features >SAF>SAF Security Profile. Inthe SAF
Security Profile Configuration window, configure a SAF security profile sothata secure
connection occurs between the SAF forwarder and the Cisco Unified Communications
Manager. Enter the same username and password that you entered on the SAF forwarder. The
username and the password are case-sensitive, so enter themexactlyas you enteredthemon the
SAF forwarder.
Toconfigure SAF forwarder configuration settings inCisco Unified Communications Manager
Administration, choose Advanced Features > SAF > SAF Forwarder.
2010Cisco Systems. Inc.
SAF and CCD Issues
The client label allows the SAF forwarder to identify the Cisco Unified Communications
Manager node. Valid entries include alphanumeric characters, underscore, and (a).. You can
enter up to 50 characters. This label string must match thc one that is configured at the SAF
forwarder.
When you configure a single SAF forwarder for the entire cluster, all nodes in the cluster use
the same SAF forwarder configuration and register to the same SAF forwarder. To create a
unique client label for the nodes in thc cluster, you can append (a) to the SAF client label value.
This ensures that thc registration message includes the bascname (SAF client label) that is
followed by (a<nodeid>. For example, you enter abcde_ny(oj for the client label for a two-node
cluster that connects to one SAF forwarder. The registration message includes abcde_ny(ii 1 for
node I or abede iivCh 2 for node 2. If you do not append the (ai, to the client label value, you do
not need to configure thc bascname on the SAF forwarder. If you append the (a: to the client
label value, you must configure the basenarne on the SAF forwarder.
Enter the IP version 4 (IPv4) address of the SAF forwarder. Enter the port number that Cisco
Unified Communications Manager uses to establish a connection with the SAF forwarder. The
default setting is 5050. The port that you enter must match the port number that you configure
on thc SAF forwarder. The port range on the SAF forwarder is 1024 to 65535.
Check the Enable TCP Keep Alive check box to ensure that Cisco Unified Communications
Manager is notified if the TCP connection between the SAF forwarder and Cisco Unified
Communications Manager fails. If this check box is unchecked, the Cisco Unified
Communications Manager is not notified that the TCP connection fails until the SAF forwarder
keepalive timer expires (configured on the SAF forwarder).
Optionally, you can also enter thc SAF forwarder advanced configuration:
For SAF Reconnect Interval, enter the time (in seconds) that Cisco Unified
Communications Manager allows to pass before it attempts to reconnect to the SAF
forwarder after a connection failure. Enter a value between 0 and 500. The default value is
20.
For SAF Notifications Window Size, enter the number of outstanding notify requests that
the SAF forwarder can maintain at the same time to the Cisco Unified Communications
Manager. The default value is 7. You can enter a number between 0 and 255. If you enter 0
in this field, thc SAF forwarder does not send any notification to this Cisco Unitied
Communications Manager. However, the Cisco Unified Communications Manager can still
publish hosted directory numbers to the SAF network if the CCDadvertising service is
configured and active.
Using thc Selected CiscoUnified Communications Managers pane, youcan prioritize this SAF
forwarder over other configured SAF forwarders.
Contiguration of the internal SAF client for Cisco Unified Communications Manager Express
will be covered in the next lesson.
4-14 Troubleshooting CiscoUnified Communications (TVOICE) v80 2010CiscoSystems. Inc
Troubleshooting SAF
This topic describes the common issues that relate to the SAF client and the SAF forwarder and
explains their potential causes.
ifying Wheth*
mistered
Cisco Unified RTMT > CallManager > SAF Forwarders
EMU*! S
UHIM CUOK-t *
5
SHIimW| wmM PIMDI Port I Ttfa >mdBBg .MMnMWp)B<Bail pnUIIBBl '*
STJt k*-,i r*c.< IB : HOIfJ! 5=150 (=*.,=, EeMDIiiiwI ="X.H 1 Rnmtrhl IW'IflfliO 7
HQ-l#aho eigrp service-family ipv-i clients
EIGRP-SPv4 VE(sof) Clients for AS(1)/IDI1Q.1.250.101)
Client Callback
Handle Name Context/PID/Registered (V/N)
10 cucml 0x00000001/ 218/Y
HQ-ltshD- eigrp service-family ipv* clients detail
EIGRP-SFv4 VRIsaf) Clients for AS (1)/IDUO.1.250.101)
Client Caliban*
Handle Nane Context/PID/Registered (Y/N)
10 cucml 0x00000001/ 218/Y
Subscribed Services/Notifications: 1/33
Published Services/Size OcB): 1/0
Sequence Number Updatad: 0
To exchange call-routing information properly with the SAF network, a SAF client (either
external or internal) has to be properly registered to the SAF forwarder with which it has been
associated.
The figure shows howto verifythat a client is registered with the SAF forwarder.
The upper figure shows howto check the registration from the Cisco Unified Real-Time
Monitoring Tool (RTMT). TheCiscoUnifiedRTMTallowsyou to viewthe report for the SAF
forwarders. In Cisco Unified RTMT, choose CallManager and then SAF Forwarders from
the Reports. The list of SAFforwarders will be shown. Their Registered Statusshoulddisplay
Registered.
The bottom figure shows how to check the registration status from a SAF forwarder. On a SAF
forwarder, issue the show eigrp service-family ipv4 clients command. This command will list
al! registered clients regardless of their type. For eachclient, the registration status Registered
column should display "Y" as yes.
For moreregistration detailsabout an individual client, use the show eigrp service-family ipv4
clients detail client command.
SAF and CCD Issues
SAF Client Statistics in Cisco Unified RTMT
This figure shows how to verily the performance of the SAF client in relation to its SAF
forwarder.
*-!; =.;iE HH'iiJC.
t-O:
T&
5f-..S^.,
Cisco Unified RTMT can display SAF client performance counters for failed and successful
connections, so it can be used as a tool for troubleshooting procedures.
You can also display SAF traffic statistics at thc SAF forwarder platform by using the
following command:
HQ-ISshow eigrp service-family ipv4 traffic
EIGRP-SFv4 VR:saf: Service-Family Traffic Statistics for ASH)
Helios sent/received; 148669 /b1318
Updates sen:; received
Queries sen;., received
Repl.es sent/received
ficky ser.t /received: 8
51/62
22/5
5/22
/0
STA-Quenes sent/received : 0/0
SIA-Replies sent/received: 0/0
Hello Process in: 2 08
PDM Process 1C; 116
Socket Queue. ; 0/20G0/2, 0 (current /max/highest /drops)
Input C'-iet-e : 0/2 C0C/ 2 ,' Z (cut lent /max/highest /drops)
4-16 Troubleshooting Cisco Unified Communicalions (TVOICE) v8.Q 2010 Cisco Systems, Inc
* Troubleshooting SAF Forwarder Adjacency Issues
A SAFforwarder receives services that are advertised by SAFclients,distributes the services
reliably through the network, and makes services available for SAF clients touse.
M,
Troubleshooting SAF Forwarder
>ncy Issues
The problem:
- SAF forwarders cannot discover each other and form an
EIGRP adjacency.
SAF Forwarder Hello SAF Forwarder
X
Consider the following causes:
SAF is not enabled on routers, or AS does not match.
Interface hellos are deactivated (shut down).
SAF forwarders are not Layer 2 adjacent, or wrong static
neighbor configuration is in place.
Network connectivity issues exist.
Hellos are blocked by a firewall.
TheSAF forwarder can propagate service advertisements toother SAF forwarders and can
propagate acrossa LAN, campus network, WAN, or ISP.
You can configure a SAF forwarder ona LAN toallow automatically thedynamic discovery of
services to all enabled interfaces and, at the same time, to specify interfaces (static
configuration) that you want blocked toother interfaces thatarcattempting todiscover their
services.
Two peering SAF forwarders must form EIGRP adjacency toexchange topology information.
Routing devices periodically send hello packets toeach other tolearn dynamically of other
routers on their directly attached networks. Thisinformation is used todiscover neighbors and
to learnwhenneighbors becomeunreachable or inoperative.
Bydefault, hello packets aresent every 5 seconds. Theexception isonlow-speed, NBMA
media on which the default hello interval is 60 seconds. Low speed is a rate of Tl or slower, as
specifiedinthe bandwidth interface configuration command.
Theholdtime is advertised in hello packets andindicates to neighbors thelength of timethat
they should consider the sender valid. The default hold time isthree times the hello interval, or
15seconds. For slow-speed NBMA networks, the defaultholdtime is 180seconds. On
congested andlargenetworks, thedefault holdtime might be insufficient for all routers to
receive hello packets from theirneighbors. In thiscase, youcan increase theholdtime. Donot
adjust the holdtime withoutadvisingyour technical support personnel.
SAF and CCD Issues
IfSAF forwarders do not discover each other and ifthey have issues forming an adjacency,
consider these causes:
SAF might not be enabled on the routers, ortheir autonomous system (AS) does not match.
In the current implementation, all SAF forwarders must be members ofasingle AS. The
following configuration enables SAF andconfigures theASnumber:
router eigrp saf
i
service-family ipv4 autonomous-system 1
Hellos could be deactivated on an interface that is in shut
down state. The following example shows FastEthernetO/1 that
is suppressed from sending hellos.
router ergrp saf
i
sf-rnterface FastEthernetO/1
shutdown
exit-sf-interface
SAF forwarders are not Layer 2adjacent, or the wrong static neighbor configuration might
be in place. The following configuration sets up the static neighbor relationship with a
neighbor 10,10.10.1 over Ethernet 0'0:
router eigrp saf
i
neighbor 10.10.10.1 Ethernet 0/0
Network connectivity issues will prevent neighbors from discovering each other and
establishing their neighbor relationship. Make surethat the SAFnetwork maintains IP
connectivity.
If there is a firewall or access list between the twoSAF forwarders, hellos could be
blocked. SAF EIGRPuses IP protocol port number H8.
4-18 Troubleshooting Cisco Unified Communications (TVOICE) vB 0 2010Cisco Systems, Inc
Jm
Troubleshooting SAF Forwarder
Adjacency Issues (Cont.)
Apr J2 16:49:**.B02= *DOAL-S-NBRCHi\BOE: EIGRP-SFv4 1: Neighbor 10.12.2.101
(Serial0/1/C.211> Is doTOi holding time expired
listiow service -family ipvt neighbors
EIGRP-SFv* VRtsa f) Serv .ce Family Neighbor for AS(1)
H hddres
B
intsrfa
Bold
(sec
uptime
SKIT
(ms)
RTO
10.12. 2.101 SeO/1/0 211 11 13.07 : 18 16 1014
1 10.1.6 .10 2 Se0/1/0 121 12 IdOlh 16 1014
0 Seq
Cut Hum
100
122
HQ-ltuho- eigrp SB cvice -family ipv4 inter f
EIGRP-SFv* VR(baf) Serv ice-Family I
Xmi t Queue
terfac
Mean
Interface P eers tin/ Reliable SRTT
Services
FaO/0.311 0 0/0 0
FaO/0.312 0 0/0 0
FaO/0.313 0 0/0 0
Se0/l/0.121 1 0/0 IE
SeO/1/0.211 1 0/0 16
Se0/1/0.911 0 o/o 0
s for ASH)
Pacing Time
Un/Rellabia
0/1
0/1
0/1
0/169
0/169
0/1
Multicast Pending
Flow Timar
229
237
This figure shows how to troubleshoot potential SAF forwarder adjacency issues. The top
figure shows aconsole message that indicates an adjacency issue; the neighbor is declared
unreachable and down.
You can verify that thc SAF forwarder adjacency isformed by using thc show eigrp service-
family ipv4 neighbors command. The middle figure illustrate when the adjacency is
maintained properly between SAF forwarders. To see thc details about each adjacency that was
formed, use the show eigrp service-family ipv4 neighbors detail command:
HQ-lSshow eigrp service-family ipv4 neighbors detail
EIGRP-SFv4 VR(saf) Service-Family Neighbors for AS(1)
Hold Uptime SRTT RTO Q Seq
(sec) (ms) Cnt Nunt
13 13:07:33 16 1014 0 100
Prefixes: 2
10 IdOlh 16 1014 0 122
Prefixes: 1
Address
Interface
0 10.12.2.101 SeO/170.211
Version 5.0/3.0, Retrans: 1, Retries:
Topology-ids from peer - 0
1 10.1.6.102 Se0/l/0.121:
Version 5.0/3.0, Retrans: 0, Retries:
Topology-ids from peer - 0
Ifneighbors cannot discover each other and establish their adjacency, use the show eigrp
service-family ipv4 interfaces command toverity that aninterface isenabled for SAF. Ifan
interface does not appear ina list ofinterfaces that this command produced, it means that the
interface is not enabledfor SAF. The bottomfigure showsthe two interfaces in which
neighbors have been discovered.
If thc issuesto formthe SAFforwarder adjacency persist, use the debug eigrp service-family
ipv4 neighborcommand to explore theneighbor discovery events:
HQ-l#debug eigrp service-family ipv4 neighbor
HQ-lfr
Apr 22 06:32:19.186: EIGRP: Packet from ourselves ignored
.' Neighbor unreachable during holdtime period, neighbor uuinstalled
Apr 22 06:32:20.466: EIGRP: Holdtime expired
Apr 22 08:32:20.466: %DUAL-5-NBRCHAHGE: EIGRP-SFv4 1:
(SerialO/i/0.211) is down: holding tirae expired
or IP-12.2.101
HC-1S
Apr 22 08:12 =20.466: Going down: Peer 10.12.2.101 total-1 stub 0, iidb Lit-^
lld-a11 = G
Apt: 2 2 0
Apr 22 0
Apr 2 2 C
Apr 22 03:32
Apr 22 ZS
Apr 2 2 0 6
Apr 22 08
IQ IS
KQ-14
I 2 . 4 6 6
iC .4 6 6
2 I .166
21.16 6
21.166
2 2 .01S
EIGK ?
s-: i gr p
EIGRP
EIGRP
EIGRP
EIGR?
Handle deallocation failure {0!
Neighbor 10.12.2.101 went down on Serial0/i/a,2ii
Holdtime expired
Ui-JAL-5-rOBRCKANGE: ElGRP-IPv4 1: Neighboi 10.12.2,101
lSerialC"'l='0.211; is down: holdina time expired
Apr 22 08:32:21 ,I6S: Going down: Peer 10.12.2.101 total-2 stub 0, iidc-stub=0
11 d -a 11 - Z
Handle deallocation failure \)':
Neighbor 10.12.2.101 went down on Serial0/1/0,211
Packet from ourselves ignored
Wei*' adjacency formed, neighbor reachable again
Apr 22 C8:34-',!1 =
Apr 2 2 CS :34
Apr 22 08:34
fSerialO/1/0
Apr 22 DB:34
Apr 2 2 CB:14
HQ-la
Apr 22 08 :34 ;JO .780: %D'JAI,- b-KBRCILANGE : EIGRP-IPv4 1: Neighbor 10,12.2.101
iSerialO.'I/O .21 ": ) is up: new adjacency
The clear eigrp service-family neighbors |soft| | neighbors-address | interface-type
interface-number| command deletes neighbors that were formed by using the IPv4 protocol
family from the neighbor tabic. Optionally, you can resynchronize with apeer without an
adjacency reset (soft). Alternatively, you can delete the interface type and number from the
neighbor table that contains all entries that were learned through this interface. You can also
shut down the SAF EIGRP routing process and bring it back as follows:
router eigrp SAF
shutdown
no shutdown
: EIGRP: Packet from ourselves ignored
: EIGRP: New peer 10.12.2.101
: IDVAL-5-NBRCHAHGE: IGRP-SFv4 1: Neighbor 10.12.2.101
211) is up: new adjacency
29. 90*3: EIGRP: Packet from ourselves ignored
is.730: EIGRP: New peer 10.12.2.101
29 .784
Troubleshooting SAF External Client Interactions
IfSAF forwarders discovered each other and formed their adjacency, the SAF network isready
tosupport SAF external or internal clients.
Troubleshooting SAF E;
Interactions
The problem:
- External SAF client does not interact with SAF forwarder, or a
SAFforwarderdoes not register with external SAF client.
Consider these causes:
$
Credentials or SAF client label do not match.
Wrong port or iPaddress is configured at eitherside.
SAF trunk is misconfigured,
I
Advanced Unified Communications Manager server
to SAF forwarder relationship is misconfigured.
Network connectivity issues exist.
For all information exchanges between the SAF forwarder and the SAF client, a SAF forwarder
has to be registered with theclient.
If the SAF external client does notinteract with the SAF forwarder, or if a SAF forwarder does
not register with an external client, consider thc following possibilities when troubleshooting:
Credentials that are configured for an external client oraclient label might not match
between the SAF forwarder and the SAF external client.
Aport or IP address might be misconfigured at either side. The port that is commonly used
for the SAF forwarder toexternal client interactions isTCP port5050, but it canbe
configured to use anotherport number.
An incorrect external client (Cisco Unified Communications Manager) SAF trunk
configuration might be applied:
A trunk might not be enabled for SAF.
- Atrunk might not beassociated with advertising orrequesting services (CCD).
These services might notexist intheexternal client configuration.
The Activated Feature check box might beunmarked at theadvertising and
requesting service configuration pages.
4-22
You can configure multiple SAF forwarders within aCisco Unified Communications Manager
cluster tor redundancy. The SAF client establishes asecure connection to the primary and the
backup SAF forwarders and registers with the SAF forwarders. By default, an instance ofthe
SAF client is created on every call-processing node within the Cisco Unified Communications
Manager cluster. Using the advanced SAF forwarder configuration option, the administrator
can create thc SAF client on selected call-processing nodes within the cluster. This
configuration option enables the administrator to create the SAF client on specific nodes within
the cluster and to configure SAF CCD with distribution of CCD services for systems that
employ clustering over rhe WAN.
Misconfiguration can result in aCisco Unified Communicalions Manager server being left off
thc Selected Cisco Unified Communications Managers pane on the SAF forwarder
configuration page in Cisco Unified Communications Manager Administration (appears under
the Show Advanced link).
The following pages provide more details on how to configure this SAF forwarder selection in
Cisco Unified Communications Manager Administration.
Network connectivity problems that can prevent the exchange ofSAF traffic can also cause
interaction issues between the SAF forwarder and the SAF client.
Troubleshooting Cisco Unified Communications (TVOICE) v8.0 2010 Cisco Syslems Inc
Troubleshooting SAF External CIh
interactions (Cont.)
I-""' 1
1*-. 1
.nterface LoopbackO
ip address 10.2.250.101 255.255.255.255
juter eigrp sa
I
service-family ipv4 e%ito
1
topology boss
external-client cucm2
exit-sf- topology
exit-service-fafflily
7-aystaxti 1
MP t*r*rtty i>hl*-Tnlo
service-family external-client listen ipv4 5050
external-client cucml
Tjaaaword
This figure shows the critical parameters when establishing aSAF external client to aSAF
forwarder relationship and starting their interaction with each other.
The client label, IP address, and TCP port number have tomatch between the two. The upper
figure shows how to configure the SAF forwarder information at Cisco Unified
Communications Manager. SAF forwarder configuration settings can be modified in Cisco
Unified Communications Manager Administration by choosing Advanced Features >SAF >
SAF Forwarder.
The bottom figure shows how to configure these parameters to match atthc SAF forwarder.
In thc SAF Security Profile configuration window, configure aSAF security profile so that a
secure connection occurs between the SAF forwarder and the Cisco Unified Communications
Manager. When you configure aSAF forwarder in the SAF forwarder Configuration window,
you must choose aSAF security profile to apply to the SAF forwarder.
Toensure thattheCiscoUnified Communications Manager canregister withtheSAF
forwarder, enter the same username that you entered onthe router (SAF forwarder). The
username and the password are case-sensitive, so enter them exactly as you configured them on
the SAF forwarder. Thevalues thatyouenterrepresent theshared secret for message integrity
checks between Cisco Unified Communications Manager andthe SAF forwarder. The
username gets included in any request from Cisco Unified Communications Manager that
contains the MESSAGE-INTEGRITY attribute.
SAF External Client Integrity Failure
This figure shows how to troubleshoot SAF external client interactions.
Hg-l*dabu3 eigrp aervice-1imily external- client meBsaqea
HQ-1*
Apr 22 OB;52:59.649:
Class: Errir Pespor.se Method: Begistei
Apr 22 08:52:59.649= Magic Cooxie: 7F5A9BC7 Transaction ID'
4D54 4A4H54212<H4D514S56
Apr 22 06:52:59.649: Realm: 014: Length: 5: "SAF"
hq-i#
Apr 22 08:52:59.649: Error Class: 4 Error Code- 31
Apr 22 08:52:59.649: Error Reason: Integrity Check Pa,lure
Apr 22 08:52:59.649: Message Integrity: 008: Length- 20-
CF8D96EF50603EDEB9EBS46 02 2EF32FS646C99 0 1
Apr 22 08:52:59.649: 0111 0044 7F5A 9BC7 4D54 4A41 4542 4244 4D51 4856 0014
0005 2253 4146 220C
Apr 22 06:52:59.649: 0000 0009 001B 0000 041F 49SE 7465 6772 6974 7920 4369
6563 6B20 4661 696C
Apr 22 08:52:59.653: 7572 6500 0008 0014 CF8D 96EF 5060 3EDE B9B8 5460 22EF
32F5 646C 9901
Use the debug eigrp service-family external-client messages command to see potential error
events during the SAF forwarder registration process. This output shows that integrity check
failure has occurred. This could be because the username or password does not match, but it
couldalso be because the client label does not match.
If you have enabled debugging but see no output display over aperiod longer than thc
keepalive that is configured at the SAF forwarder external client configuration you have
misconfigured eitherthc IPaddress or theTCP port number.
You can verify that the client label at the SAF forwarder is also using the show eigrp service-
family external-client command:
HC-ldshow eigrp service - family external-client
SAF External Clients
Cleric Label Client API Handle File Dese-iptcr-
cucml i q r
The same command, but shown with details for a particular external client, shows additional
parameters, for instance, keepalive timers:
HQ-ltfshow eigrp service-family external-client cucml
Detail data for client -- cucml
Username: cucml Version: 1.2
Client Contexc: A3 ill- AH: IPv4Tcpology.- Base
Router NaT-e-: saf
Internal Cljer.t AP; Kind:e: 10 Page Size: 7 Allowed to Send- /
Fi-e descriptor: 1 IPv4 Address: 10.2.1.1 Tcp Port: 5060
C_ier.t NaiT.e: 'J CM/ CCCM1-1/NodeId=l/8,0.1.10000-4 0
Keepalive Period(sec!: IS Keepalive Time Remaining(sec): 9
True client has 1 s-jbsci ipriou
The following debug output shows both successful and unsuccessful registrations:
HQ-l#debug eigrp service-family external-client mesBagea
HQ-1#
;Successful registration followed by normal SAF forwarder to SAF client
interaction
Class: Success Response Method: Register
Apr 22 08:56:56.262: Packet LengCh: 52 Not including 20 byte Saf Header
Apr 22 08:56:56.262: Magic Cookie: 7F5A9BC7 Transaction ID:
5349414B544F484253444C4D
Apr 22 08:56:56.262: Keep Alive: 1005: Length: 4: 15000
Apr 22 06:56:56.262: Client Handle: 1002: Length: 4: 12
Apr 22 08:56:56.262: Message Integrity: 008: Length: 20:
4FA363I8E141A73A50BC8FD2BD4 0F8416566064C
Apr 22 08:56:56.262: 0101 0034 7F5A 9BC7 5349 4148 544F 4842 5344 4C4D 0
Apr522208:56:56.262: 0000 1006 0004 0000 3A98 1002 0004 0000 0OOC 0008 0014
4FA3 6318 E141 A73A
Apr 22 08:56:56.262: 50BC 8FD2 BD40 FB41 6566 064C
Keepalives regularly exchanged at configure period
14
Apr 22 0B:52:55.617: 0001 0038 7F5A 9BC7 564F 4C45 5B41 5145 4F59 4555 0006
Apr522%8:52:55.617: 0000 0014 0005 2253 4146 2200 0000 1002 0004 0000 000A
OOOB 0014 AD7F B23B
Apr 22 09:52:55.621: CE1A 643B 322F 28AE B118 1144 B65C D1DE
Apr 22 0B:52 :55.621:
Apr 22 08:52:55.621: - ,
class: success Response Method: Keepalive
Apr 22 08:52:55.621: Packet Length: 56 Not including 20 byte Saf Header
Apr 22 08:52:55.621: Magic Cookie: 7F5A9BC7 Transaction ID:
564F4C4 55841514 54F594555
Apr 22 08:52:55.621: Realm: 014: Length: S: "SAF"
Apr 22 08:52:55.621: Username: 006: Length: 5: cucml
Apr 22 08:52:55.621: Client Handle: 1002: Length: 4: 10
HQ-1#
Apr 22 08:52:55.621: Message Integrity: 0OB: Length: 20:
477B16 317A8BBC56FBFABA9E093 09E9B50E1011C
Apr 22 0B:52:55.621: 0101 0038 7F5A 9BC7 564F 4C45 5B41 5145 4F59 4555 0014
0005 2253 4146 2200
Apr 22 08:52:55.621: 0000 D006 0005 6375 636D 3100 0000 1002 0004 0000 OOOA
0008 0014 477B 1631
Apr 22 08:52:55.621: 7ABB BC56 FBFA BA9E 0930 9E9B 50E1 011C
: integrity failure, credentials misconfigured at either side
Apr 22 08:52 :59.649:
Class: Error Response Method: Register
Apr 22 08:52:59.649: Magic Cookie: 7F5A9BC7
4D544A41454242444D514856
HQ-1#
Apr 22 08:52:59.649: Error Class: 4 Error Code: 31
Apr 22 08:52:59.649: Error Reason; lat'egrity'"ChfedJsFatlltre
Apr 22 08:52:59.649: Message Integrity: 008: Length; 20:
CF8D96EF506 03EDEB9E8546 022EF32F5646C9901
Apr 22 08:52:59.649: 0111 0044 7F5A 9BC7 4D54 4A41 4542 4244 4D51 4B56 0014
0005 2253 4146 2200
Apr 22 08:52:59.649: 0000 0009 001B 0000 041F 496E 7465 6772 6974 7920 4368
6563 6B20 4661 696C
Apr 22 08:52:59.653: 7572 6500 0008 0014 CF8D 96EF 5060 3EDE B9E8 5460 22EF
32F5 646C 9901
Transaction ID:
Tracing Registration to a SAF Forwarder
This figure shows thc trace output ofasuccessful registration with aSAF forwarder.
4-26
WCU.ntConsrol - Begir, sendSaf Register (.., 0,0.0,0.0---.
1PAddc 10.2.25C.101
Prt 5050
connStatug SaEEs tabl ished
clientHandle i6
BdlTCPPid Uj 100i 9- 25)
safClientSettingPkid e18b8da-fb3-d98-3b67-a0a572b60e06
numConnetionReAttempts ... 1
timerReconnectlnterval ... 20
numSafPaeketaSent ... 1344
numOfRegisteredApps 2
forwatderName SAPF- HQ2
=lientName UCM/CUCM1-2/NodeId.l/e.0.1.10000-40
cliencLabel cucm2
teepAlivelcterval 15000
isKATimerRunning 1
designation SafPtimary
iuRegistered 2
enableTgpXeepalive i
safNotificationsHindowSiie 1
To see this trace output, mark the check box Enable Forward &Miscellaneous Trace in the
system diagnostic interface (SDI) trace configuration for Cisco CallManager service.
Ihe figure shows the start ofthc registration process and the formulated SAF connection
information details that Cisco Unified Communications Manager sends to its primary SAF
forwarder. You can sec the registration parameters as the SAF forwarder IP address, TCP port
and the client label in this trace output. The name with which thc SAF forwarder has been
identified at thc Cisco Unified Communications Manager cluster is SAFF-1IQ2.
The registration is successful, and thc connection status is Sailistablished.
Tracing Registration to a SAF
(Cont]
Security Profile Details
uthentlcationHode .... 2
0,0,0.0.0* **
SAFClientCootrol - End B<md3atRegi uter ') |0,P.O.0.0
The trace output continues in this figure in which you can see the security profile that has been
used for all conversations to the SAF forwarder.
The end ofthe registration process is marked with End sendSafRegister.
SAF and CCD Issues
Troubleshooting SAF Information Exchange
4-28
This sect.on describes how to troubleshoot thc issues when aSAF forwarder docs not exchange
call control information.
The problem;
- SAF forwarder does not exchange cadcontrol information.
* Adjacency is not formed.
- Remote SAF client that advertises the missing call control
information does not interact with the SAFnetwork.
' Split horizon is in action.
* Percentage of fink bandwidth used forSAFis set too low.
* Routeauthentication is in place.
- Maximum hop count is set too low.
* Network connectivityissues exist.
[fan external client is registered with its associated SAF forwarder but no call control
information is received, considerthesecauses:
The SAF forwarder might not have established EIGRP adjacency with other SAF
forwarders. Solve this problem as described earlier in the section that covers EIGRP
adjacency.
Aremote SAF client that is supposed to advertise the missing call control information does
not interact with the SAF network on its side. Solve this problem asdescribed earlier in the
section that covers SAF registration.
EIGRP split horizon could be in action (default setting). When split hori?on is enabled on
an interface, it blocks route information (such as update and query packets) from being
advertised by arouter out of any interface from which that information originates.
Controlling update and query packets in this manner reduces the possibility of routing
loops.
This behavior usually optimizes communications among multiple routing devices,
particularly when links are broken. However, with nonbroadcast networks (such as Frame
Relay), situations can arise for which this behavior is less than ideal. For these situations
including networks in which you have Cisco SAF configured, you can disable split horizon.
You can verify if split horizon is active by using the show eigrp service-family ipv4
interfaces detail command:
HQ IWshow eigrp service-family ipv4 interfaces detail SerialO/1/0.211
EIGRP-SFV4 VRi.saf: Service Family interfaces for AS(1 )
Xm;l Queue Mean Peering Time Multicast Fading
1-e"a" Pceis vj.-i/Rel lablc SRTT tin/Reliable Flow Timer
Services
SG0/:'G-2:"- l U/C 16 0/169 ?37 n
Troubleshooting Cisco Unified Communications (TVOICEl v80
Hello-ir.terval is 5, Hold-time is 15
Split-horison is enabled
Next xmit serial <none>
Un/reliable mcasts: 0/0 Un/reliable ucasts: 91/107
Mcast exceptions: 0 CR packets: 0 ACKs suppressed: 0
Retransmissions sent: 7 Out-of-sequence revd: 0
Topology ids or, interface - 0
Authentication mode is not set
The percentage of link bandwidth that is used for SAF, might be set too low. By default,
packets consume amaximum of 50 percent of the link bandwidth, as configured with the
bandwidth interface configuration command. If, during periods ofcongestion, a
percentage is set to avalue close to "0," it could theoretically happen that no bandwidth is
left for SAF traffic. Use the following commands toconfigure the percentage oflink
bandwidth that is used for SAF:
router eigrp saf
I
sf-interface FastEthernetO/0
bandwidth-percent 75
exit-sf-interface
i
exit-service-family
Call control information that isdistributed by using the SAF network could be secured with
MD5 authentication. Ifthe authentication does not work as expected and the process fails,
the information that isdistributed might not reach itsfinal destination and could be
discarded by the SAF forwarder. Use the following command to verify ifauthentication is
enabled:
HQ-i#show eigrp service-family ipv4 interfaces detail SerialO/1/0.211
ElGRP-SFv4 VR(saf) Service-Family Interfaces for AS(l)
Xmit Queue Mean Pacing Time Multicast Pending
interface Peers Un/Reliable SRTT Un/Reliable Flow Timer
Services
SeO/l/0.211 1 0/0 16 0/169 237
Hello-interval is 5, Hold-time is 15
Split-horizon is enabled
Next xmit serial <none?
Un/reliable mcasts: 0/0 Un/reliable ucasts: 91/107
Mcast exceptions: 0 CR packets: 0 ACKs suppressed: 0
Retransmissions sent: 7 Out-of-sequence revd: 0
Topology-ids on interface - 0
Authentication mode ia not Bet
2010 Cisco Systems, Inc SAF and CCD lssues 4"29
4-30
EIGRP maximum hop count could be set too low. Maximum hop count limits the number
ot hops that aservice can propagate to advertise its service. The default number of
maximum hops is 100. Ifthe number ofhops that arc used to advertise aservice is limited
the advertised information might not reach its destination, and it could be discarded when
reaching thc limit. Use the show eigrp protocols command to verify the maximum hop
count limit that is used in EIGRP;
HQ lnshow eigrp protocols
EIGRP-SS--V4 VR.saf: Service-Family Protocol for AS (1 )
Metric weight Kl-i, K2 =0, K3=-l, K4-0, K5=0
NS"-aware route hold n-'.er is 2=10
Routei-ID: 10.i.250.101
lopoiogy ; : ;basej
Active Timer: 3 Tin
distance: internal 92 external 170
Maxi-jrr. path: :
Maximum hopcount 100
Maximum metric variance 1
Network connectivity issues might also hamper thc normal distribution ofthe call control
information within the SAF network.
Troubleshooting Cisco Unified Communications (TVOICE] v8 0
Troubleshooting
Exchange (Cont.)
BQ.1#bI!Cw eigrp service-family ipv4 topology
EIGRP-SFv4 VRIsst) Topology Table for AS(1) /IDUO . 1.250 .101)
Codes: P - Pasalve, A - Active. U - Update. Q - Query, B - Reply,
r - reply Statue, a - aie Statue
p 101:2:gFFC79E6.1EC325Bfc.2SE610F1.16AOO, 1 successors, PD is 2297B5S
via 10.12.2.101 (22978S6/128256), SerialO/I/O.211
P 101:2JBFFC79BS.7EC325BA.26E610F1.ISA01, 1 auccesSOIS. FD is 2297656
via 10.12.2.101 [2297856/12B256I, SerialO/I/O.211
P 101-2iBfiE6S?9.0.0.10200. 1 successors, FD is Si97BS6
via 10.1.6.102 I2297S56/12fl256> , SerialO/1/0.121
P 101:2 .F8AF.19B7 .B41SDP3D.D1I85360.1726D, 1 successors, FD la 12B256
via Connected, HullO
P 101:2:F8AE19B7.B416DF3D.D1EBS360.1726C, 1 euccessore. FD is 1282S6
via Connected, NullO
BO-l#Bhov eigrp service-family ipv4 events
1 13:59;55.323 Poison squashed: 101: 2: FBAB19B7 .B416DF3D.DlBB5360.n26B
reverse
2 13:59:65.323 Ignored route, dup routerid lot: 10.1.250.101
3 9 13;S9=S5.259 Change queue emptied, entries: 1
10 13-59:155.255 Metric set: 101. J i FBAB19B7 . B=U 6DF3 D. D1K8536 0.1726B 128256
11 13:59:55.255 Update reason, delay: new if 4294967295
Ifyou suspect that the call control information is distributed improperly, check thc EIGRP
topology for SAF. The Figure shows the output ofthe show eigrp service-family ipv4
topology command that displays the topology table that is being built from three SAF
forwarders: thelocal SAF forwarder (NullO), 10.12.2.101, and 10.1.6.102.
To verify aparticular topology table entry, supply the command that includes the topology
descriptor in the format Service:Subservice:Instance.Instance.Instance.Instance. The following
output shows the missing topology infonnation;
HQ-l#show eigrp service-family ipv4 topology
101:2:EFFC79E6.7EC325BA.26E610F1.16A01
E1GRP-SFV4 VR{saf) Topology Entry for AS(1)/ID(10.1.250.102)
%Entry 101:2.EFFC79E6.7EC32SBA.26E610F1.16A01 HOC in topology .Sable
Ifyou display the topology infonnation that is currently stored in the topology table, the output
will look like this:
HQ-l#show eigrp service-family ipv4
topologyl01:2:EFFC7 9E6.7EC325BA.26E610F1.105C5
EIGRP-SFv4 VR(saf! Topology Entry for AS(I1/ID(10.1.250.101) for
101:2.EFFC79E6.7EC32 5BA.2 6E610F1.105C5
State is Passive, Query origin flag is 1, 1 Successor
Length: [673] , Sequence No: [4], Owner Client handle: [0]
Originating Address: 10.2.1.1 Port: 5060, Protocol: 6
Subscription Handles: 10
Descriptor Blocks:
10.12.2.101 (Serial0/l/0.21l) , from 10.12.2.101, Send flag is
Composite metric is (2297856/126256), service is Internal
Vector metric:
Minimum bandwidth is 1544 Kbit
Total delay is 25000 microseconds
Reliability is 255/255
Load is 1/255
Minimum MTU is 1500
Hop count is 1
Originating router is 10.2.250.101
FD is 2297856
0x0
201001500 5x5161113, Inc.
4-32
If topology information exchange issues exist, the following command is very useful for
troubleshooting because it displays all topology exchange events that occurred in the past.
HQ l#show eigrp service-family ipv4 events
: 13:55:55.323 Poi sen squashed: 10] :2:F8AE19B7 .B416DF3D.D1E85360 1726E
reverse
2 13 ::;9 :55.32i Ignored route, dup routerid int : 10,1.250.101
J 13:59:55.323 Poison squashed: 101:2:F8AE19B7.B4I6DF3D.DlK85360 1726A
revei=e
4 .3:=9 :5=.323 Ignored route, cup routeria int: 10.1.250.101
5 13:59:55.3:5 Foiscr. Squashed: 101 :2:F8AE19B7 .B41 6DF3D.D1E85360 17263
reverse
6 -3 :-j9 :55 .31 5 Ignored route, dup routerid int: 10.1.250.101
7 13:59:55.315 Poison squashed: 101:2:18AE19B7,B416DF3D.D1E85-60 1726A
reverse
8 13:59:55.315 Ignored route, dup routerid int: 10.1,250.101
9 13:59:5^.2^9 Change queue emptied, entries: 1
: :3:bf' :f:^55 Metli^ set: 101:2:F8AE19B7.B416DF3D.D1E35360.1726B 128256
Ij. .3:i;9 :-53 .255 Update reason, delay: new if. 4294967295
-2 13:59:55.255 Update sent, RD: 101 ;2:FSAFJl 9B7 .B4 16DF3D .Dl K853 60 1726P
4294 96 /29_^
Ij i3:=9:5: .255 Update reason, delay: segno changed 4294967295
14 13:59:55.255 Update sent, RD: 101:2:F8AE19B7.B416DF3D.D1E85360 1726S
4294SS"-'235
]5^ ^-3:59:55.255 Route installed: 101:2:F8AE19B7.B416DF3D.D1E85360.1726B
16 13:53:55.255 Find FS: 101:2:F8AE19B7.B4I6DF3D.D1E85360.1726B 4294967295
-3:=9:5=.255 Serv update met /succmel : 128256 0
ie _3 :59: 5.i .255 Rev jpdate dest/orig: 101:2:F8AE19B7.B416DF3D D1E85360 W26E
Connected
19 13:59:55.255 Metric set: 101:2:F8AE19B7.B416DF3D.D1E85360 1726B
4294967295
2u 13:59:55.255 Local rt change: 101:2:F8AE19B7,B416DF3D.D1E853S0 r?263
Conr.ectea
21 ".3:59:55.255 Change queue- emptied, entries: 1
t2 :-:5!:--'?55 metric set: 101:2:F8AE19B7.B416DF3D.D1E85360.1726A 128256
2i -::53:55.2b5 Update reason, delay: new if 4294967295
24 11:59:55 255 Update sent, RD: 101 :2;F8AE19J37 .B41 6DF3D D?EB5360 1726A
429496729-
^5 -3:3^:^3.2 =5 Update reason, de'.ay: seqr.o changed 4294967295
26 13:53:55.255 Update sent, RD: 101;2:F8AE19B7.B416DF3D.D1E85360 1/16A
4294967295
il :3:=9;55.255 Route installed: 101:2:F8AE19B7.B4]6DF3D.D1E85360 I726A
0 .0 .0 .C
25 13:53:55.255 Find TS : 101 ;2:F6AE19B7 .B4 16DF3D.D1.E853 60.1726A 4294967295
" '3:59:55.255 Rev update met/succmet; 128256 0
3G L3:59:55.255 Rev update dest/orig: 101 :2.F8AE1 9B7 .B41 6DF3D .DIE8S 36C 1';.-.
Connect ed
31 13:59:55.25= Metric Set: 101:2:F8AE1 9B7 .B4 16DF3D .Dl F.8 53 60 n?6A
4294967295
J2 13: 59:55.2:^5 Local r, change: I01:2:F8AE1 9B7 .B4 16DF 3D.D1E8 5i(, 0 1726A
Ccnr.ec. ea
33 :3:59:=5.125 Ignored route, inaccessible:
101:2 :F8AE19B7.B=;i6DF3D.DlE85360.1726 8 4294967295
34 13:59:55.123 Ignored loute, dup routerid int: 10.1.250. 101
35 ii:59:-5.i23 Ignored route, inaccessible:
101:2:F8AF19B7.B416DF3D.DIE85360.17269 4294 9672 95
36 13:59:55.123 Ignored route, dup routerid int; 10.1.250.101
37 13:59:5^.119 Ignored route, inaccessible:
101:2:F8AE19B^.B416DF3D,D1E85360.1726 8 4294 967295
38 13:59:55.119 ignored route, dup routerid int: 10.1.250.lOi
39 '.3:59:55.119 Ignored route, inaccessible:
101:2:F8AL19B7.B416DF3D.D1E85360.17269 4294967295
Troubleshooting Cisco Unified Communicalions (TVOICE) vB 0 2010 Cisco Systems Inc
Summary
Summary
SAF provides a framework that allows applications todiscover
the existence, IPaddress, port, and configuration of networked
resources within networks.
The most common issues of the SAF networkinclude SAF
forwarders that cannot establish their adjacency, a SAFclient
that does not interact with the SAFforwarder, and SAF
forwarders that do not exchange call control information.
In this lesson, vou learned to explain the common issues that relate to the SAF client and
forwarder in an environment with CCD and toidentify the most likely causes ofthese issues.
References
Cisco IOS Service Advertisement Framework Configuration Guide at
http: w\v\v.cisco.comcn'US/docs.'ios'saf-'configuration/giiidc/
saf en psl0591_TSD Products_Configuration_Guide_Chapter.html
Cisco Unified Communications System 8.x SRNDCall Routing and Dial Plan
Distribution Using Call Control Discovery for the Service Advertisement Framework at
http: uww.cisco.com'en !.,S.'docs.'voicc_ip_.comm/cucm/smd''8\/!nodels.lnnil#w|il 1(11849
Lesson 21
Troubleshooting CCD
^ Overview
When multiple call-processing agents are present in the same system, each can be configured
manually to be aware of the others. This configuration can be time-consuming and error-prone.
Call routing between the various call-processing agents requires the configuration of static
** routes on the call agents and their updating when changes occur.
Service Advertisement Framework (SAF) can be used instead to share call routing and dial plan
information automatically between call agents. SAF allows call agents (such^^v,sl
"" multiplexing [TDM] PBXs) that arc not proprietary to Cisco to partake in SAF when they are
interconnected through a CiscoIOSgateway.
* SAF enables networking applications to advertise and discover information about networked
mm serviceswithinan IPnetwork.
The first service to use SAF is Call Control Discovery (CCD). CCD uses SAF to distribute and
maintain information about the availability of internal directory numbers E.164 with | prefix
numbers and route patterns-such as tail-end hop-off (TEHO)-that are hosted by call control
agTnts such as Cisco Unified Communications Manager and Cisco Unified Communicates
# Manager F-xpress.
Objectives
Upon completing this lesson, you will be able to explain the common issues that relate to CCD
as an application of SAF and identify the most likely causes of these issues. This ability
includes being able tomeet these objectives:
Describe thc CCD service and its components and describe how CCD operates
Describe the common issues ofCCD and explain thc causes ofthese issues
Call Control Discovery
This topic describes the CCD sen-ice and its
components andhow CCD operitts
San Jose Unified Communications Manager
Routing Table
S.'12XXXX +121244414 102.11 StP
5442XXXX +442077112/4 10.2.1.1 H.323
101 1.1
New York Unified Communications Manager
Routing Table
84WXXXX +1409555/4 101.1,1 S|P
-1949222/4 10.1.1.1 H.323
102.1.1
Advertising 3212XXXX
8442XXXX
SAF Network
Advert sing B408XXXX
8949XXXX
SAF-Enabled H.323 Trunk
SAF-Enabled StP Trunk
The figure shous rwo Cisco Unified Communications Manager clusters, each having two
naSZ'l hT T: T,am hSled dnCCl0ry numbcr P8"8" di*toiy "*er
patterns that belong to the local call control entity. The San Jose Cisco Unified
949XXXCvatl""S "anaffr.JaVhe tW "USlcd direct0fy 'b" P^terns 8408XXXX and
8442XXXX ' C ' ommunicaiions Manager in New York has 8212XXXX and
alot ^thfSAF?'' T^ ft"Ver Confi*uration and hosted directory number patterns
along *,th the SAF trunk access information to the remote call control entities that use thc SAI
IlLlvV OfK.
Each Cisco Un.tlcd Communications Manager also listens for advertisements from remote call
umberLn"n'CS ?"* ** SAF,*- This ""ures * .earned patters (hosted direct,^
aZo ? U- remte Ca" COmr0' em,ti") arC inscrtcd "lto di& analysis on the local
Cisco Unified Commumcations Manager.
The SAF forwarder .he Csco IOS router -notifies the local Cisco Unified Communications
Manager uhen remote call control entities advert.se their hosted directory number patteniê
SAF forwarder receives publish requests from the local Cisco Unified Communications
Manager cluster so that Cisco Unified Communications Manager can advertise thc hosted
directory numbcr patterns for thc cluster.
SAF-enabled trunks that are assigned to the CCD advertising service II 3-1 or Session
S^Z1 TZt^Vrr n,bUUnd Ca"S frm "^"" "trol "^ 'hat use the
SAF network. SAF-enabled trunks that are requesting service and assigned to thc CCD
requesting service manage outgoing calls tolearned patterns.
CCD Operation Review (Cont)
SanJoseUnified Communications Manager
Routing Table
I 111^ lil IiiIWIi III -WILLLHUHJIM
6212XXXX t12l2444/4 10.2.1.1 SIP
8442XXXX +442077112/4 102.1.1 H.323
10.1.1.1
New York Unified Communications Manager
RoutingTable
8408XXXX
B949XXXX
+1408555/4 1011.1 SIP
+1948222/4 10.11.1 H323
102.1 1
Calling
84422001
SAF-Enabled H.323 Trunk
SAF-Enabled SIP Trunk
This figure shows the call to the hosted directory number pattern that leverages the established
SAF network. When the pattern is learned and the SAF network is reachable, callers can place
calls between the clusters by using the prescribed SAF-enabled trunks.
When connectivity to aremote call control entity is lost, the SAF network notifies Cisco
Unified Communications Manager to mark the learned information as IP unreachable The call
then goes through the public switched telephone network (PSTN) (not shown in the figure)
The same call would go to+442077112001, in this case.
CCD Operation During IP Connectivity Failure Review
This figure explains what happens after an IP connectivity failure when the CCD client cannot
reach the call control component that advertised the pattern being called.
4-38
San Jose Unified Communications Manager
Routing Table
ESEOSOBHH3HH^Si ESS
W/XXXX +1212444)4 '0i1 SjEy
&442XXXX +442077112 >*^Q^ Qy
NewYork UnifiedCommunications Manager
Routing Table
840BXXXX t4408555/4 10.11.1 SIP
8949XXXX +1849222M 10.1.1.1 H.323
102.1.1
Any call that uses SAF-lcarned routes has automatic PSTN failover if the IP path to the called
numbcris unavailable. Thccall is routedin this order:
Take thc selected IP pathtoreach the called numbcr.
a If the IP path is unavailable, use the PSTN failover digit transformation rules (ToDID) to
modify the called number and route the call through the PSTN.
Thc figure shows an IP network connectivity failure that breaks thc integrity of the SAF
network that is overlaid on it. Typically, Cisco Unified Communications Manager has lost thc
connectivity to its SAF forwarder, or SAF forwarder-to-SAF forwarder connectivity has been
broken.
Cisco Unified Communications Manager marks the patterns that are learned from SAF as
unreachable and activates PSTN failover for the patterns. The PSTN failover uses the ToDID
rule that is associated with each pattem to construct avalid PSTN number. For instance, the
ToDID rule in the figure determines to strip four digits from thc learned pattern (sue code in
this particular example) and add the prefix (4420771 12 to form the number +4420771122001
that can be used for placing acall via PSTN instead ofthe IP network.
Cisco Unified Communications Manager Fxpress is also capable ofPSTN failover. It uses thc
same concept as Cisco Unified Communications Manager.
Troubleshooting Cisco Unified Communications (TVOICEl v8.0
CCD Configuration Elements on SAF External Client
This table describes the configurationelements of Cisco Unified Communications Manager that
are used for thc CCD configuration.
CCD Configuration Elements on the
SAF External Client
Hasted DN Group
Hosted ON Pa Hern
CCD Advertising Service
CCD Requesting Service
Stocked Learned Pallerns
One SAF SIP loir* and one SAF H323 frunk can be configured
They are not configured will a destination IPaddress. Rest of
the conf guration is similar Io normal SIP and H323 trunks.
Confined with PSTN failover strip digits and PSTN latover
prepend digits. Applied Io Hosted DN Patterns.
DN ot DN range to be advertised Configured witti PSTN failover
strip digits and PSTN faitover prapend digits; if not configured.
Hosted DNGroup conligiration Is used Raters to Hosted ON
Group.
Refers Io Hosted DN Group, SAF SIP trunk, and SAF H323
trunk
Configued with route partition, learned pattern prefix, and PSTN
prefix. Refers to SAF Trunks.
Configured with remote P. remote call control identity, and
learned pattem or learned prefix.
You must configure a SIP or H.323 (nongatekeeper-controlled) trunk so that it supports SAF.
For SIP trunks, choose Call Control Discovery from the Trunk Service Type drop-down
menu, which displays where you assign the trunk type and trunk protocol. Be aware that you
cannot change the trunk service type after you choose it from the drop-down menu.
For H.323 (nongatekeeper-controlled) trunks, check the Enable SAF check box in the Trunk
Configuration window when you configure the trunk (after you choose the trunk type and trunk
protocol). To disable SAF on the H.323 trunk after you enable it, uncheck the Enable SAF
check box. If a trunk is assigned to a route group or is associated with a route pattern, you
cannot enable SAF on the trunk. Likewise, if you enable SAF on the trunk, you cannot assign
the trunk to a route group or associate the trunk with a route pattern. Verify that the SIP trunk
has a security profile that is set to nonsecure before you enable SAF on the trunk. You cannot
enable SAF on SIP trunks that use authenticated or encrypted security profiles.
Supported with the CCD feature, hosted directory number groups are a collection of hosted
directory number patterns that you group in Cisco Unified Communications Manager
Administration. After you assign a hosted directory number group to the CCD advertising
service in Cisco Unified Communications Manager Administration, the CCD advertising
service advertises all the hosted directory number patterns that are a part of the hosted directory
number group. You can assign only one hosted directory number group per the CCD
advertising service.
The Hosted DN Configuration window supports the CCD feature, which allows Cisco Unified
Communications Manager to use the SAF network to advertise information such as directory
number patterns to other remote call control entities that also use SAF. You associate these
patterns with hosted directory number groups, which allow you to associate multiple patterns to
a CCD advertising service easily.
i 2010 Cisco Systems, Inc SAF and CCD Issues 4-39
The CCDrequesting service, which supports the CCDfeature, allows Cisco Unified
Communications Manager to listen for hosteddirectory numberadvertisements from remote
call control entities that use the SAF network. Inaddition, the CCD requesting service ensures
that learned patterns are inserted intothe digit analystsmasterroutingtable.
Adedicated CCD partition under Call Routing -> Call Control Discovery > Partition supports
only theCCD feature; that is, all learned patterns automatically belong totheCCD partition
that you assignto the CCDrequesting service. The CCDpartition ensuresthat the learned
patterns areinserted into digitanalysis under thispartition for CCD. Thepartition that you
assignto the CCDrequesting servicemust belong to a callingsearchspace(CSS)that thc
devices canusefor calling the learned patterns, soassign thepartition totheCSS that you want
thedevices to use. If youdo not assign a CSSthat contains the SAFpartition to thedevice, thc
device cannot call the learned patterns.
The Blocked Learned Pattern Configuration window supports theCCDfeature by allowing you
to purgeandblock learned patterns, for example, learned patterns that you no longerwant to
use. You can purge learned patterns that you no longer want to use, and you can block the
learned patterns so that CiscoUnified Communications Manager ignores thc patterns when
thev are advertised by remote call control entities.
For example, you might want to block a learned pattern with prefix235 fromremote call
control entityxyz withthe IP address of 111.11.11.! I. Youcan blockthe pattern specifically
for this call control entity by entering the relevant information in the Block Learned Patterns
window. In this example, after you save the configuration, the CCDrequesting service searches
thelocal cache andpurges the learned patterns with a 235 prefix from remote call control entity
xyz with thc IP address of 11 LI 1.11.11. Any subsequent notification with this informationis
blocked and ignored byCiscoUnified Communications Manager.
Blocking andpurging of patterns is based on exact match. Forexample, configuring 235XX
blocks 235XX, not any subsetsof that pattern. Be awarethat if youdo not specifya remote call
control entityand an IPaddress for the entity, CiscoUnified Communications Manager purges
and blocks the panem for all remote call control entities that advertised it.
4-40 Troubleshooting CiscoUnified Communicalions (TVOICE) v80 2010CiscoSystems, Inc.
%r
km
m*
w
Relationship of Configuration Elements on the External SAF
Client
This figure shows how the CCD configuration elements of the Cisco Unified Communications
Manager as an SAF client interact with each other.
Relationship of Configuration Eler
on the External SAF Client
SAF Forwarder
Note Configuration element is called the SAF
forwarder because it points to the SAF
forwarder, the configuration element itself
belongs to Ihe SAF dienl.
SAF Client
Configuration elements relationships can be 1:1, n: 1, or n:n.
2010 Cisco Systems. Inc. SAF and CCD Issues 4-41
CCD Configuration Elements on the Internal SAF Client
This section describes the configuration elements of the internal SAF client.
l 0 ( ( hllpfnlK
hiiei iirfi bar kAla
li' ..ItS OP life
Trunk Profile
DN Block Profile
Call Control Profile
SAFi il Ctii
profit* trunk-routs: Configured wilf . ..
should be used for signaling when setting up SAF calls
profile dn-blocK. Coiiliguredwilh patterns tn be advertised (internal
number an) number used for PSTN backup)
profits callcontiol: Refers to DNblock profiles and trunk profile
channel. Configured wiih SAF client ID and autonomous system
Advertising and requesting services are enabled. Ihe advertising
service refers to the call control profile
dial pew voice: Configured wtlh destination pattern Tend session
target saf This is the ncoming and outgoing dial peer tor a call sent
to or received from SAF trunks.
1he table describes the configurationelements of Cisco UnifiedCommunications Manager
Express or any other internal SAF client that is used wilhin thc CCD environment.
4-42 Troubleshooling Cisco Unified Communications (TVOICE) v8.0 )2010 Cisco Systems, Inc.
m
f
\m
.
I**
nm
Relationship of Configuration Elements on the Internal SAF
Client
This figure shows how the CCD configuration elements of Cisco Unified Communications
Manager Express, as a SAF client, interact with each other.
Relationship of Configuration
on the Internal SAF Client
SAF Forwarder Other SAF Clients
"W%m*
\
SAFaî ;;
Ih'.l'.r. ;
X
DN Block
Profile
DN Block
Profile
SAF Client Channel
CCD Requesting ^H
Service ^^|
\X.;
Call Conlrol Profile
CCDAdvertising ^Q
Service ^^|
^0M^
Trunk Profile | ':L"!!!!!-!!!!(!;:
' c--S:"}.".:.'''"%::\ Dial-Peer SAF ['
The typical internal SAF client configuration looks like this:
interface LoopbackO
ip address 10.1.250.102 255.255.255.255
voice service saf
profile trunk-route 1
session protocol sip interface LoopbackO transport tcp port
5060
i
profile dn-block 1 alias-prefix 521555 strip length 3
pattern 1 type extension 5213XXX
profile callcontrol 2
dn-service
trunk-route 1
dn-block 1
channel 1 vrouter saf asystem l
subscribe callcontrol wildcarded
publish callcontrol 2
destination-pattern .T
session target saf
The SAF forwarder that is configured at ihe same plalform as the CCD internal SAF client
would be configured, for example, like this:
router eigrp saf
topology base
exit-sf-topology
exit-service-family
4.44 Ttoubleshooling Cisco Unified Communicalions (TVOICE) v8 0 2010Cisco Systems, Inc
Troubleshooting the SAF Client in CCD
This topic describes the common issues of CCD and explains the causes of these issues.
"9
IMIn i~
QuntiMi
sumiimi owm-i
i i:>x- ?"hQriii>::rBij 0,^,0!,
FHHRt WD OKmniM
loinocsm ;i;ms
1031 1(33(1117) 3532W5
This figure shows how to display the patterns that are learned from the SAF network.
Because SAF learned routes are dynamic, they arc not held in the Cisco Unified
Communications Manager database, but they are stored in memory. Use the Cisco Unified
Real-Time Monitoring Tool (RTMT) to display SAF learned patterns.
This figure shows four patterns that have been learned. They are coming from two sources:
Cisco Unified Communications Manager cluster with the ID CID 10.2.1.1, the server
10.2.1.1 of this cluster
Cisco Unified Communications Manager Express (ID not shown because it does not exist)
with the IP address 10.1.250.102
The patterns are reachable either via SIP or via an H.323 trunk.
When patterns are unreachable via the IP network, the ToDID field shows the direct inward
dialing (DID) that must be performed to manipulate the pattern to a format that is dialable in
PSTN. All patterns need to strip three leading digits and add a prefix to modify a pattern to
complete the PSTN format.
Verifying Learned Patterns at the Internal Cisco IOS Client
This figure shows how to verily' if patterns are learned at thc Cisco Unified Communications
Manager Express client.
vent
Inlor
L.1..4,*rf t=u t, f <
BR-l#show voice sit dndb all
Total no. of patterns io db/max allowed .
Patterns classified under dial plans (private/global!
Informational/Error stats -
Patterns w/ invalid expr detected while add . 0
Patterns duplicated under the same instance . 0
Patterns rejected overall due to max capacity , 0
Attempts to delete a pattern which is invalid : 2 C
Last successful DB update <f 2010:04:22 05:38:36:130
.,.,., private Dialplan Partition
Pattern - 5 112XXX
Primary Trunli -Bout e Is) IB
Aliae-Routels) Prefix/strip-Len
Pattern - 5122XXX
Primary Trunk-Route <e) IC
Alias-Route 13) Prefix/5trip -Len
n . 5 22iXXX
ary Trunk-Rou
B-Routeia) Pr
;(s) IE
:ix/stnp-Len
The client has learned three patterns from the SAF network. You note that this Cisco Unified
Communications Manager Express client is allowed to learn a maximum of 6000 patterns.
Error information that is related to a pattern learning process is also provided here.
To verify reachability information of each individual pattern, use the following command that
displays information like Cisco Unified RTMT did for the Cisco Unified Communications
Manager client in the previous illustration.
BR-lffshow voice saf dndb detail 5112XXX
Kluxer ci primary t run.-: routes 2
Trun< Rcj;e id - sz
Session target
Session Protoco:
Session 7: ar.spci L
Preioaaed Route set
runK Rotate i.J - -9
Session Target
Session Protocol
Session 7i a:iscoi t
ipv4 : IC . 1 .1 .1 . b()60
SIP
TCP
sip:0058c499-92ar:-2c57-55de-e4cld8b8a7a6-.vCUCMl-:
ipv4:lfi.l .1 . 1 .33031
H323
TCP
NuiTiOsr cf alias routes 2
Alias Pretix/Strip Len - 511555/0
Alias Prefix/Strip I.en - 5:iS5b/3
Troubleshooting Cisco Untied Communications (TVOICE) v8 0 2010 Cisco Systems, Inc
Troubleshooting Patterns Learning Process
Thissection describes howtotroubleshoot an issue when patterns arc not learned from theSAF
forwarder.
Troubleshooting Patterns Learning
Process
The problem:
* Patterns are not being learned at the SAF client.
* Connectivity to the SAF network is not established.
* CCD requesting service is not activated at the local client.
Trunk is not selected for the CCD requesting service.
SAF network does not distribute call control information.
Remote CCD dient does not advertise patterns.
Patterns are blocked at the local CCD client.
Maximum allowed number of learned patterns is reached.
Cisco Unified Communications Manager, as an implementation of the SAFexternal client, runs
two services that are responsibleto exchange patterns with the SAF network:
CCD advertising service: This service makes the local Cisco Unified Communications
Manager cluster advertise the PSTNfailover configuration, the hosted directory number
patterns, and the SAF-enabled trunk access information for its cluster to the remote call
control entities that use thc SAF network.
CCD requesting service: This service makes the local Cisco Unified Communications
Manager listen for advertisements from remote call control entities that use the SAF
network. The CCDrequesting serviceis responsible also for inserting learnedpatternsfrom
the remote call control entities into digit analysis and the local cache.
If you discover that patterns are not being learned at thc Cisco Unified Communications
Manager CCD client, consider the following potential causes:
SAF connectivity between Cisco Unified Communications Manager and thc SAF forwarder
might not be established and functional. Verify that Cisco Unified Communications
Manager is registered with the SAF forwarder, as explained earlier.
The CCDrequesting servicemight be configured improperly beforebeingactivated at
Cisco Unified Communications Manager (Activated Feature check box).
The trunk might not be selected for the CCD requesting service.
The SAF network could be in a state of not distributing call control information (SAF
network connectivity is broken). Verify that the SAFforwarders are receiving Enhanced
InteriorGateway Routing Protocol (EIGRP)topology information, as explained earlier.
The remote CCD external client might not advertise the hosted patterns, its advertising
service might be configured improperly, or it might not be activated (Activated Feature
check box).
If you configured the patterns blocking feature at Cisco Unified Communications Manager,
thc patterns could be blocked and prevented from being entered into digit analysis and the
local cache.
The CCD feature parameter named CCD Maximum Number of Learned Patterns is
configurable at Cisco Unified Communications Manager. This parameter specifies the
number of patterns that this Cisco Unified Communications Manager cluster can learn from
the SAF network. Thc higher the number of allowed learned patterns, the more memory
and CPU processing power is required for your server.
When Cisco Unified Communications Manager attempts to learn more patterns than arc
specified in the parameter configuration CCDLearnedPattemLimttReached, an alarm is
issued. To access the feature parameters that support the CCD feature, choose Call
Routing > Call Control Discovery > Feature Configuration in Cisco Unified
Cisco Unified Communications Manager Express shows how many patterns are allowed to
learn by using the following command:
BR-l#show voice saf dndb summary
Total r.c. ci patterns in ctD/max allowed : 3/6000
Patterns c'. assifieci under" dial plans (pr ivate/global ) : 3./0
Ir.torT.at i oca'- ;Err;r sracs -
Patterns * ' lnva.i;; expr det e;cte(l while add
Par.r.srr.s duplicated under the same instance
Patterns rejected overa:1 due to max capacity
Attempts to delete a pattern which is invalid
0
0
0
20
4-48 Troubleshooting Cisco Unified Communicalions (TVOICE) v8.0 2010 Cisco Systems. Inc
Verifying Advertising and Requesting Services
This figure shows which configuration elements to verify for advertising and requesting
services torun properly atCisco Unified Communications Manager.
Verifying Advertising and Requesting
Services
Check at remote CCD client Check at local CCD client
SA' SIP Ttj-.^ 5A'_S|P M **
felMkt*iWwp CCDJ
Arvl-tê SAFTru-**
. W
|: .î.^. 1
h>u e+i bmi i"f* - .^^-^~-
VA
1
H....J CH 0t~ !<
--.- |,Ui. |
i^^^ |
DOTH ifw5*--:i f>0** 3
r ' Ui* Hi*-** *TH I* J*f
The CCD advertising service, which supports the CCD feature, allows Cisco Unified
Communications Manager toadvertise the hosted directory numbers for the cluster and the
PSTN failover configuration to remote call control entities that use the SAF network. At the
remote Cisco Unified Communications Manager, choose Call Routing>Call Control
Discovery >Advertising Service to display the CCD Advertising Service Information
configuration page.
Ensure that the Activated Feature check box ischecked. If theActivated Feature check box is
not checked, the CCD advertising service does not work. Also verify that you have chosen the
trunk that you want to use with this CCD advertising service (SIP orH.323). For inbound calls
to the Cisco Unified Communications Manager, thecall is routed totheappropriate trunk that
is advertised by theCCD advertising service.
Supported with the CCD feature, hosted directory number groups are acollection ofhosted
directory number patterns that you group in Cisco Unified Communications Manager
Administration. After you assign a hosted directory number group to the CCD advertising
service inCisco Unified Communications Manager Administration, thcCCDadvertising
service advertises all thehosted directory number patterns that area part of thehosted directory
number group. You can assign only one hosted directory number group per the CCD
advertising service.
The Hosted DN Pattern Configuration window supports the CCD feature, which allows Cisco
Unified Communications Manager to advertise itshosted pattern tothe SAF network and allow
other remotecall control eniitiesto learnthis information. Makesure that a correctpatternis
enteredat remoteCiscoUnifiedCommunications Manager. The hosteddirectory number
pattern can contain amaximum of50 characters, and you can enter the international escape
character + followed by pattern ordialable digits (0-9A-Da-d), pattem ([6-9]), wildcard
character (X), or (A) with optional %or ! at thc end of the entry.
The CCD requesting service, which supports the CCD feature, allows Cisco Unified
Communications Manager to listen for advertisements from remote call control entities that use
the SAF network. In addition, thc CCD requesting service ensures that learned patterns are
inserted into the digit analysis.
Ensure that the Activated Feature check box is checked. Ifthe Activated Feature check box is
not checked, thc CCD requesting service docs not work. At least one SAF-enabled trunk must
exist in the Selected SAF Trunks pane; otherwise, the CCD requesting service does not start for
thc local cluster. Cisco Unified Communications Manager does not register with the SAF
forwarder, and patterns are not learned.
4-50 Troubleshooting Cisco Unitied Communications (TVOICE) v8.0
Tracing the CCDAdvertising Patterns Process
This section demonstrates the trace output for advertising hosted patterns to the SAF network.
Tracinq the CCD nq P;
CCDAdve rtis ingS arvi en
haoge-0, pat-1, id-0
CCDAdvertisingServic<
change-1, pat=>l, id-0
CCDAdvertiaingservic*
lane-587. 0.0,0,0.0 "* *'
CCDRdvaitisingService: -puhlishSarviCH () trunk-1.1.1. lOil.t;
veraioD-6, xol-c?xml veraion--1.0" sneoding.-UTF-8"?>
eservica-description xmlna-"http=7/ww". Cisco.com/naneepaces/Baf-uc"
jailosrisi-"http://winf.w3 .org/lOOl/MLSchema-instance"
xsi =schemaLocation- "http://www. Cisco, com/names paces/aaf -uc sa-uc-v0.95 .
scheaVersioD-"1.0" id-"1.0">
thosted-dn><deacriptioo><product>UCM</product><versioa>8.0.1.1000 0-
40</verBions<enterpriaanaa>CIDl0.1.1. le/anterpriBenaie>clocatlon>C.JCMl-
le/Iocs tion>e/deacription><t rim*-rout e>epiotocol><h323/>
</protocols-:/trunk- route xdn-pat tern vti aion-"2"xp
d."3:511555 ">5112XX!.e/dn -pat ternx/hoatsd-dnx/aervi ce
de acription>I J 0,0,0,0.0"*''*
: tcheckAdvertisafcllServicesO advartli
(17269, ver-5|0,0.0.0.0"""*
: , check*dvartiae115ervic eat)
(1721)8, ver-S1 0,0, 0,0.0""*-
::encodaSafKmlOpaqueDatal) au
adv*3
statu b=i,
Dtatus-2,
prot-6.
To see this trace output, mark the following trace configuration fields of Cisco CallManager
service: Enable All Phone Device, Enable All GateWay Trace, Enable Miscellaneous Trace,
and Enable Forward & Miscellaneous Trace.
The traceoutput showsthe advertisement of a singlehostedpattern 5112XXX withits PSTN
failover information(ToDIDis 3:511555). The CCDAdvertisingService::publishService also
showsthe trunkthat is boundto the pattern(trunk=l. 1.1,10:33031) inwhich33031 is the
ephemeral portnumber thatwaschosen forthe H.323 trunk at thisCisco Unified
2010 Cisco Systems. Inc SAF and CCD Issues
Tracing the CCD Patterns Requesting Process
This section demonstrates the trace output for the process oflearning (requesting) hosted
patterns from the SAF network.
inq the
CCBSe quest ingService; j decodeSaf XndOpagueDa
ta, xnilPtr=<service-description
xmlna=-http=//www.clsco.com/nameapaCBa/B8f -uc
xmlnH ;ui. -http://www.w3.org/20 01/KKLSchema-inatance-
xai!schemaLocation="hctp://www.Cisco.com/n
amespacea/Bat-uc sat-uc-vO.35.Ksd"
achenaVersion-"1.0" id="1.0">
<hoHted-dn>tdescriptionxproductjUCMi/product?tveraionj8.0. 1.100 00-
4 0t/versionxenterprisenaiLe>CID10.2. 1 .li/enterpriaenami>xlocation>CUCMl-
2 </locationx/deecriptionx trunk-routeseprotocolxsip/>
i/protocol xtrunk-acceBflxroute>Hip: dOf 2 8d4d- bce3 -d3be-e8ef -
c3165e818cJ9CUCMl-2</route:>
i/trunk-accessx/trunk-routexdn-pattern v ersion.-l" xp
dr."3i 51255 5"?5 122xxx</ps</[]n-pattern></hoHted-dn>t/aervice-
description! 0,0,0.0.0"""*
CCaRequeetingService::storeLearnsdPatternl afo{) with traosType
6 0,0,0,0.0"*"*
CCDReguestingEarvice::ait SAPEatalnd, not fy - ok, addr^ ',',:. .
decode = true 0,0,0,0.0**"*
Digit analysis, add to the localRegiatrati sub 16 6bc87 9-6661-3adl-dl12-
fdlbbS9253el/5122XXx . PID: 1,203.5* patte
trnUsage - [21J 0,0.0,0.0*-'-
Djgit analysis: addPatternToDigitAnalysis patternUsage -[21J
=allableEndPointName. 15122XXX: 1 66bc879-6 661-3adl-dl42 - tdlbbB9253el]
patternHodeld= [512 2XXX :16tibca79-6661-3adl- il42-fdlbb89253el] 0,0,0,0.0*'*
Io see this trace output, mark the following trace configuration fields ofCisco CallManager
service; Enable All Phone Device. Enable All GateWay Trace, Enable Miscellaneous Trace,
and Enable Forward & Miscellaneous Trace.
The trace output shows that a pattern 5122XXX has been learned from the SAF layer and
decoding at theCCD layer. Thepattern wasoriginally advertised bytheCisco Unified
Communications Manager with the IPaddress 10.2.1.1 named CUCM 1-2 being part of the
cluster with identifier CID10.2.1.1. The pattern isreachable atthis call control agent by using
the SIP trunk. Youcanalsoseethe PSTN failover field (ToDID) for thispattern as
d-"3:5 12555".
When the pattern is received, it is entered immediately todigitanalysis andthe cache.
The pattem is learned from Cisco Unified Communications Manager. Forpatterns learned from
Cisco Unified Communications Manager Express, the trace output isslightly different, as
follows. It does not showa cluster identifier (because thereis none) nor the client name:
CCDReqjestir.gService: : decodeSaf XmlOpaq'jeDat a, xml Ptr-< service -descript ier.
xmins-"http: .' /www .Cisco . com/namespaces/saE- uc"
xm',ns:xsi-"htr.p: //www.wj org/2001/XMLSchema instance"
xsi; schcrv.a_.ocat icn=- "nttp: //www . ciscc.. com/namespdees/sat- uc saf -uc -vO . 35 . xsc,"
scheir,aVersion="l .0" id=" 1 . 0" =< hosted -dnx trunk -
routea<prctocc1><sip/></protocol>=/trunk-routexdn-pattern veision-"1">5213XXX</dn pattern?-;/hosted-dnx/servi ce -
description?.0,0,C,0.0*"*'
4-52 Troubleshooting Cisco UnifiedCommunications (TVOICE)v8 0
i 2010 Cisco Systems. Inc
Troubleshooting an Incorrect Learned Pattern Prefix
This section demonstrates how totroubleshoot the situation when a pattern isunreachable
because of an incorrect learned pattern prefix configuration.
Troubleshooting incorrect Learned
Pattern Prefix
Hosted pattern Learned and Reachable in Cisco Unified
RTMT Learned Pattern report.
- But dialing to the pattern fails.
QunlM
1 VMOHf" [ " TbORJ
=mi icbci -sum*
1Q1 250 I j\VI60i ' 5'1^*=
Stationlnit: (0000002) EnblocCall
calledPatty.5122001. !1,100.49,1.6313"10.1.2.12"SBP0024C445AD8
Digit analysis: match(pi-'2'.fqcn.'S115552001-, en-"2001-, plv=-5",
pss.-HQ Emergency Pt:BQ_Intl Pt:HQ LD Pt:HQ_Loeal_PtiIfitBrnal_Pt:CCD_Pf,
T:,,- i,,-f?=, ,-ev E-a-srs-.E--' rz-.Y/v Ii;t.l Fti.'ifj -& f:-:HQ I-o-:-il I-t: J-n*:ni-nal Pi"-
".t n-, dd-5122001',dac.-1").1,100,49,1.6313"10.1.2.12"SEP0024C4454AD8
Digit analysis: potentialHatches-HoPotentialMatchesExist|
1.100,49,1.6313"10.1.2.12"SBP0024C4454AD6
Ifcalling to alearned pattem fails, usually the first troubleshooting step is to verify ifthe
pattern has been properly learned from the SAF network. The upper figure shows that all
patterns learned have theReachable status.
The bottom figure shows the tracing output ofCisco Unified Communications Manager digit
analysis. You can see that the number that matches one ofthe patterns that was learned was
dialed at the IP phone. The number 5122001 was dialed. Despite its match with the learned
pattern, however, it remains unreachable, as indicated in the trace output digit analysis
immediately reporting NoPotentialMatchesExist.
The digit analysis also shows the list ofpartitions that the calling phone can reach:
TodFilteredPss="HQ_Emergency_Pt:HQ_lntl_Pt:HQ_LD_Pt:HQ_Local_Pt:Internal Pt
:CCD_Pt".
The CCDPt ispart ofit, so the calling does not fail due towrong calling privileges.
SAF and CCD Issues
DNA > Service > Control Center
E23-
DNA can be enabled lo analyze, also
based on CCDlearned patterns
,G^3
Before performing analysis byusing DNA, synchronize CCD
learned oallems Celweendigitanalysis lable and DNA
| ONA - Dialed Number Analyser
' *" '" "=' """
Hnlillitf..|t.
Dialed Number Analy/er (DNA) can be enabled to perform analysis that is based on the
patterns that have been learned by CCD. IfDNA service was activated and is running, vou can
enable this type ofanalysis from the menu ofDNA, Choose Service >Control Center and
click Add. IfCisco Unified Communications Manager has been registered to at least one SAE;
forwarder, the status should change to Learned Patterns Added, and Update and Remove
buttons should replace thc Add button.
Ifyou want to perform thc analysis that is based on CCD learned patterns, use the button
Update to update DNA from the digit analysis table. The CCD requesting service installs the
patterns, because DNA docs not do it automatically. Then you can use DNA for the analysis
that is based on CCD learned patterns, thc same as for static patterns that are configured
manually at Cisco Unified Communications Manager,
The figure shows the analysis of5122001 dialing. Based on both static and CCD learned
patterns analysis, the result is loblock this pattern because ofanunallocated number.
Troubleshooting Incorrect Learned
Pattern Prefix (Cont.
CCDRequeBtingService::decodesafXmlOpaqueData, xmlPtr.<ser
m.lns--h "p://*. Cisco, com/naneapacea/saf-uc
ln9:xi-"http!//w.-3.org/JOOl/WLSchelw-instance '
jtsiiachemaLocati on--http://. cisco. com/name spaces/ ua -=.
gchemaVe rBion^'l-O'1 id"*1.0">
ehoated-dn><descriptloQ.<product>UCM</product<varsiDn>8. 0.1.10000-
40/inn>t.tsri1.i.>C.'.-'^ .! 5l/t.prlMn locttlon, WU
-</loction>e/da scriptionxtrunfc-route xprotocolxsip/>
^protocol.* trunk-c!.M<t>.lp.faid*d-bc.3-a3b.--f-
c316 5818c29cnCHl-2':/routa>
s/truafc-BCceasx/trualt-routaxan-pattBrD vtiiioM-l'xp
d--3 ,512655 .5122XXIe/px/dn-patteriix/host =d-dnx/sarvice-
deBcriptioriJ ,0,0.0.0.0"*"*
CCDBequestiagServicei iBtoreLairoedPattemlntol) with tranBType -
6 0,0,0,0.0**"*
W^.Mly.1.. dd to the local Registrations 16Gbc879-6661-*adl-dl42-
fdlbb8-253el/88.5122-C-X , PID: 1,203,Si patterDUsage - [21] | 0,0, 0,0. 0
Digit analysis. Odd local pattem 166bc879-6661-3adl-dl42-
fdlbbB9353Bl/8B.5122X)t3L . PID; 1,203.8,0,0,0,0.0 * *
nioit analvais- addPatternToDlgitRnalyBis - pattarnUsage .[211
ci?;.birEDdP0intNM.-[BB.5122^,166bc879-6661-3.dl-dl42-dlbb89253Bll
pattemNodeld- [SB. 512 2XXX =166bc87 9-6=56 l-3adl-dl42-
fdlbb892 53Eil 0,0,0,0.0**"* ^
ce-daacription
c-vO .95.
Returning to the trace output shows that the CCD requesting service has added to the digit
analysis the pattem that includes the prefix 88, forming anew pattern 885122XXX.
This prefix does not show in the Cisco Unified RTMT learned pattem report, and it is visible
only at the CCD requesting service configuration page of Cisco Unified Communications
Manacer Administration.
SAF and CCD Issues
4-57
Troubleshooting CCD PSTN Failover
This section shows how to troubleshoot an issue when PSTN failover does not work for the
CCD patterns.
4-58
Troubleshooting CCD PSTN Failover
The problem:
1When a pattern ismarked UnReachable, CCD PSTN failover
does not workforthat pattern.
Consider the following causes:
- AAR CSS atthe phone does not allow accesstothe PSTN
failover pattem
' Advertising the SAF client sends incorrect ToDID information.
- Incorrect PSTN prefix applied atCCD requesting setvice.
1 Incorrect CCD FeatureParametersset to control PSTN
failover.
Local PSTN dial plan or PSTN call routing issues exist.
It apattern that ,s learned from the SAF network cannot be reached by using on-net callmg the
pattern is marked in digit analysis as UnReachable. Sec this also in thc Cisco Unitied RTMT
Learned Pattern report. To maintain calling to such apattern, the advertising service can
propagate PSTN failover information to reach the destination by using off-net as backup.
If the pattern ,s marked UnReachable, but CCD PSTN failover docs not work for that patten,
consider these causes:
The PSTN faikner mechanism of CCD uses automated alternate routing (AAR) at Cibco
tinned Commumcations Manager. AAR do not need to be enabled (you can leave r at the
St.SCH!ng) mC,sco CallManager service parameters. The AAR group is nol needed lor CCD
PSTN ta.lover. because the complete PSTN number is constructed based on the ToDID field
that is associated with the learned pattern. However, to gain access to the PSTN service AAR
CSS must be configured at the calling phone. If there is no AAR CSS configured at the phone
or it it does not allow access to thc PSTN route pattem that is used for PSTN failover call
routing, the call is blocked.
The process of CCD PSTN failover relies on ToDID information that is advertised with the
hosted pattern. Uan advertising SAF client (Cisco Unified Communications Manager)
"1C0TTeCt rD'D mronlion-lhe fi^l numbcr that is constructed is not mutable and
PSTN failover fails. The ToDID field comprises the parameters that are configured at the
hosted directory number group or the hosted directory number pattern configurations.
' Svhl' 'C-amcd Patlem Prefix CiirMcr-thc CCD jesting service might apply an incorrect
ISINprefix that makes the PSTN number unreachable. If, for instance, PSTN access code
9must be dialed lo reach the PSTN, you would configure thc PSTN prefix ofthe CCD
requesting senice lhat is equal to 9. In this way, thc PSTN failover process can use thc dial
plan that was configured previously at Cisco Unified Communications Manager
mm
The CCD PSTN failover operation is controlled by thc CCD Feature Parameters that are
configured at Call Routing > Call Control Discovery > Feature Configuration. If they arc
misconfigured, the entire PSTN failover might be unsuccessful or might lead to unexpected
behavior. The parameters are explained in the next section.
Because PSTN failover relies on the local Cisco Unified Communications Manager PSTN
dial plan, its incorrect configuration and operation affects the PSTN call routing during
failover periods. The calling phone also has to have CSSs to call PSTN route patterns that
are used for the CCD PSTN failover. Troubleshoot these particular issues as dial plan
issues.
CCD Feature Parameters for the PSTN Failover
This section explains how to adjust the CCD Feature Parameters to support thc CCD PSTN
failover process.
Number of seconds that learned patterns stay active, until PSTN
failover starts to apply.
h=99*41ir<J 'tfu*
Number of minutes that calls to unreachable learned patterns are
routed through the PSTN If set to 0, PSTN failover never occurs.
The CCD Feature Parameters are configured at the Call Routing > Call Control Discovery >
Feature Configuration page.
The CCD Learned Pattern IP Reachable Duration parameter specifies the number of
seconds that learned patterns stay active (reachable) before Cisco Unified Communications
Manager marks those patterns as unreachable. For example, you configure 20 seconds for
this parameter. When Cisco Unitied Communications Manager cannot communicate with
the SAF forwarder after 20 seconds, all calls to learned patterns fail over to the PS I N until
IP connectivity to the SAF forwarder is restored.
During the PSTN failover. Cisco Unified Communications Manager cannot learn new
pancrns. After the time that you specified for this parameter elapses, Cisco Unitied
Communications Manager marks the learned patterns as unreachable. Use this parameter
with the CCD PSTN Failover Duration parameter, which allows patterns that have been
marked as unreachable to be reached through PSTN failover. You can enter a number
(seconds) from 0 to 300: the default equals 60 seconds.
The CCD PSTN Failover Duration parameter specifies thc number of minutes that calls to
unreachable or inactive learned patterns arc routed through the PSTN gateway and then
purged from the system. Thc configuration for this parameter does not take effect until after
the timer expires for the CCD Learned Pattern IP Reachable Duration parameter.
The expiration of the CCD Learned Pattem IP Reachable Duration parameter indicates that
IP connectivity fails between the SAF forwarder and Cisco Unified Communications
Manager, and all learned patterns are marked as unreachable. Then, when thc duration
expires for the CCD PSTNFailover Duration parameter, all learned patterns are purged
from the system and calls to purged paiterns are rejected (caller hears reorder tone or
"number is unavailable" announcement).
4-60 Troubleshooting Cisco UnifiedCommunications (TVOICEl v8 0
*
Setting this parameter to "0" means that PSTN failover cannot occur. If the SAF forwarder
cannot be reached for the number of seconds that you defined in the CCD Learned Pattern IP
Reachable Duration parameter, no failover option is provided over the PSTN, and calls to
learned patterns immediately fail. Setting this parameter to 525600 means that PSTN failover
never expires and learned patterns never are purged because of IP connectivity issues. You can
enter a number (minutes) from 0 to 525600; the default equals 2880.
>2010 Cisco Systems, Inc. SAF and CCD Issues 4-61
Ensure that Correct PSTN Number Is Made for CCD PSTN
Failover
This figure describes how a complete PSTN number is made up for the CCD PSTN failover.
Ensure the Correct PSTI
"---
Ltvr.ct if.tai Cr,1 .
:::-_
Qitmrntnam
Mt-dtlH* CUCWt 1 -
Pntwo Aflr*fl WtMti
* i
i 1
j p
a 1
ty
ToDID: 512 Stripped, 512555 Prefixed
PSTN Prefix 00
005125552001
PSTN
The instructions on how to manipulate a learned pattern when CCD PSTN failover takes place
is advertised with the hosted pattern as a ToDID field. The field that is shown in this Cisco
Unified RTMT learned pattern report shows that three digits should be stripped (started from
the pattern beginning), and 512555 should be prefixed to create the PSTN routable number.
You can also configure the CCD requesting service that adds another PSTN prefix as the
pattern is being added to digit analysis. In this case, 00 would be added to all learned patterns.
Make sure that the final PSTN pattem is correct; otherwise, call routing during CCD PSTN
failover periods will fail.
4-62 Troubleshooting Cisco Unilied Communications (TVOICE) v8 0 2010 Cisco Systems, Inc
t*
%
*
>*"
Tracing a Successful PSTN Failover Call
This section explains trace output of a successful CCDPSTN failover call.
Tracing a Successful PSTN Failover
ScatloDlDit: (0000 002) EnblocCall
callsdP*rty-5132C01. 1,10 0.49.1.7331"10.1.3 . 12 "SEPO0 24C44 54Anfl
nip -
Digit analyaiB: analysis resultsi1,100,49,1.7331*10.1.2.13"SBP0024C4454AD8
Pi-BtransSormCall ingPartyHunber-SOOl
CallingPaityHuinbec-2 001
DialingPaititiomCCD Pt
DialingPattern-5 122XJUI
FullyOualif iedCalledPartyNuiDbar.5 122001
.PoteQtialHatches.NoPotentislMatchosExist
- mnip -
CCDRequestingServiceChild - buildPstoFailoverDigi tstring - r.riij'Ur^j-j -.
=-. ' , mPSTNFailoverTrlggeted- 0, failoverPattern.
3:512555 '1,100.4 9.1.7 331"10.1.!.12"SEPOC21C44S4AI>8
- > nip -
Clgit Analysis, star DaBeqs Hatching Lsgaqy Numeric,
digit -005125552001 1.100,49,1.7331*10.1.2.12*SP0024C4454AD8
The trace starts with the IP phone calling 5122001 that matches the learned pattern 5122XXX.
The complete digit analysis traceoutput for this call is shownhere. Notealso the CCDpartition
and CSS:
Digit analysis: match (pi="2", fqcn-"5115552001". en-"2001",plv="5",
pss="KQ_Emergency_Pt:HQ_Intl_Pt:HQ LD Pt:HQ__Local Pt:Internal Pt.CCD_Pt",
ToaFUteredPss="HQ_EmergencyPtiH^
D_Pt", dd="5122001",dac="l")|1,100,4 9.1.7331"10.1.2.12*SEP0024C4454AD8
Digit analysis: analysis results|l,100,49,1.7331*10.1.2. 12*SEP0024C4454AD8
Pretrar.sformCallingPartyNumber=2001
|CallingPartyNumber-2001
JDialir.gPartition=CCD_*?t
|Dialir.gPattern=5122XXX
|FullyCualifiedCalledPartyNumber=5122001
|DialingFatternRegularExpression=(5122[0-9] [0-9] [0-9])
|DialingWhere=
|PatternType=Enterprise
|PotentlaiMatches^NoPotentialMatchesExist
|DialingSdlProcessld=(l,203,15)
|PretransformDigitString=5122001
|PretransformTagsList-SUBSCRIBER
|PretransformPositionalMatchList=5122001
|CollectedDigits=5122001
|UnconsumedDigits=
|TagsList=SUBSCRIBER
|PositionaiMatchList=5122001
|VoiceHailbox=
|VoiceMailCallingSearchSpace=
|VoiceMailPilotNumber=299 0
|RouteBlockFlag=RouteThisPattern
jRouteBlockCause=l
|AlertingName=
|UnicodeDisplayName=
|DisplayNameLocale=l
|Intercept Part ition=CCD Pt
4-64
InterceptPattern=E)122XXX
IntercepzWhere-
InterceptSd:ProcessId=(1,203, 15 )
Intercept5s7ype=]6 77 72 72
InteicepLSsKcy=C
Ir.tcrceptSsNot ifyType =I
GverlapSendingFiaqi;nabled=U
WithTags-
w'thvalucs-
CaUingPa:'. yN jiiicc-'"I:i ^Kot -Se 1ectea
Conr.ectedPa: tyN'umoerPi =KotSelected
Cal 11ngPa rtyNamei'i-NotSei ected
Connected PartyNamePi =Nnr. Selected
CallManagerDeviceType-KoDeviceTypc
Part err.Precedent:eLeveL=Roat me
Cal lableEndPcmtName- [5'. 22XXX :166bc8 79-6661 -3adl dl4 2 fdlbb8 92 53elj
PatternKodeId=[5122XXX:166bc879-6661-3adl-dl42-idlbb89253el]
AARNeighborhcod= ; ]
AARDest mat icriMask- [ \
AARKeepCal IHistcry-Lue
AARVo". ceMa i ILr.ahlea-taise
iNetworkLocat icn=OnKet
Because the calling has taken place during thc PSTN failover period (the learned pattem
5122XXX marked UnReachable). the CCD needs to build a complete PSTN number that is
based on the original called number (origString).
CCDRequestingSeiviceChild buildPstnFailoverDigitString - ongSr muj =
51220C1, r.PSTMFai Iove:"Tr:ggered = 0 , f ai love i Pat t ern-
3 :bl2bbb ; ". , ". CO, 49, 1 .7331*10.1.2. 12"SEP0024C44S4ADS
The bottom of the trace output shows that the digit analysis now works with thc complete
PSTN number that also includes the PSTN prefix that is configured at the CCD requesting
Digit Analysis: star DaRcq: Matching Legacy Numeric,
digits=005125552001|1,ICO,49,1.7331*10.1.?.12*SEP0024C4454AD8
Troubleshooting Cisco UnifiedCommunications (TVOICE) v8.0 2010 Cisco Systems. Inc
mm
Tracing a Successful PSTN Fi
Call (Cont.)
Digit analysis: analysis results ' 1,100,49 , 1.7 331* 10 .1. 2 .12 *SEP0024C445=iADS
prBtransformCalliogPartyKuinber-20 01
CallingPartyNumber=5115552001
DialingPartition.HQ LD Pt
DialingPattern.O .0 11-9] 1
FullygualifiedCaliedPartyNunber-005125552001
DialingPatternRegula repression- (0) (0{1-91 [0-91.)
DialingWhere.
Pat t emType-Enterpri se
PotentialKatcheB-NoPotantialHatehesExi at
- mnip -
PretranBformPosltionalHa tchl.ist-0:05125552001
CollectedDigits-0512 5552001
Due onauinedD igi t b -
TagsList.SUBSCRIBER
PositionalKatQhLint-0512 5552001
VoiceKailbox.
VoicBKallCalllngSearchSpacesi
voiceMailPilotNumber.299 0
RouteBlockPlag.RouteThis Pattern
- IDlp -
Sit"".'1 i- :.;-=-_?: :indLocalDevlce - N?rr.-EU i>^-s il Key-f04bae66-G233-cel2-
7f76 -eaB334f 3197d isActvie.l Pid-(1, 7 3 , 2 )
found 1. 100, 49, 1. 7331*10. 1.2. 12"SEP0024C45 4j\DO
Because the called numbcr has been manipulated into a complete PSTN format, thc PSTN route
pattern now applies for call routing. This trace output shows that the route pattem 0.0[l-9j! has
now been used for the call routing via PSTN. The complete trace output is shown here. Note
also the partition and AARCSS(highlighted) of thecallingphone.
Digit Analysis: getDaRes data: daRes.ssType=[0] Intercept DAMR.sstype= [0] ,
TPcount^ [0] , DAMR.NotifyCount=[0] ,
DaRes.NotifyCount=[0]|l,100,49,1.7331*10.1.2.12"SEP0024C4454ADB
Digit analysis: match (pi="l", fqcn="5115552001", cn="2001",plv="5",
pSS="HQ_Emergency_Pt:HQ_Intl_Pt:HQ_LD_Pt:HQ_Local_Pt:Internal_Pt",
ToaFilteredPss"HQ_êrgencyj&iHQ~l8U^
dd="00512555200l",dac="l")|l,100,49,1.73 31*10.1.2.12"SEP0024C4454AD8
Digit analysis: analysis results|l,100,49,1.7331*10.1.2.12*SEP0024C4454AD8
PretransformCaliingPartyNumber=2001
|CallingPartyNumber=5115552001
|DialingPartition^KQ_LD_Pt
|DialingPattern=0.0[l-9] !
|FullyQualifiedCalledPartyNumber=005125552001
|DialingPatternRegularExpression=(0) (0[l-9] [0-9]+]
|DialingWhere=
|PatternType=Enterprise
|PotentialMatches=NoPotentialMatchesExist
|DialingSdlProcessId= (0,0,0)
jPretransformDigitString=005125552001
|PretransformTagsListÂCCESS-CODE:SUBSCRIBER
jPretrar.sformPositionalMatchList =0:05125552001
]CallectedDigi"_s=05125552001
|UnconsumedDigits-
|TagsList-SUBSCRIBER
|PositionalMatchList=05125552001
|voiceMailbox=
|VoiceMailCallingSearchSpace=
|VoiceMailPilotNumber=
|RouteBlockFlag=RouteThisPattern
|RouteBlockCause-0
|Alert ingtJame-
|UnicodeDisplayNan-.e =
|DisplayNameLocale=l
|OverlapSendingFlagEnabled=0
,^6>:interrial_Pt",
WithTags=
W:t'nValues=-
CaliingPartyNumbe:Pi =NotSelected
Ccnnec t ed?a r t yNi: mber Pi-Kgt Selected
Callir.gPartyKamep-. =Not Se". ected
ConnectedPart y Maine Pi -No'_ Select cd
CallManagerDeviceType NcDevicc-Typc
Pat ternPrccedenceLeve1 =Rout ine
Cal . aolcL:r.dPoir.tKaTr.e= [fC4bee66- 6233 eel2 7f 7 6 -eaS33 <1 tl 1 97d!
Patterr.Ncdeld- [797bl267-d799-b653-f179 79e8f c84178e]
AARNeighborhood= C
AAROest mat ionXask- \]
AARKeepCallHistory-true
AARVoiceMailEnaDled=fa!se
Net.wor.KLocatio:i=Of f Net
Ihe bottom of the trace output on this figure shows that the route list was found (EU-PSTN rl).
From this moment on, ihe call proceeds as a normal PSTN call.
mm,
Troubleshooting an Incorrect PSTN Prefix
This traceoutputshowshowan incorrect PSTN prefixthat is configured at the CCDrequesting
service shows up in the CCD PSTN failover trace.
Troubleshooting an Incorrect PST!
Prefix
Stationlnit: (0000002J t. ; : "si
y-,- <.-- ;: -. . i,ioo,49,i.s63i"io.i.a . i2*sepoo24C44 5*adb
- Snip -
StstionD: (00000021 DialedNumber dialedNumbsr-5122001 linelnntance.l
callReference-209 4934 9. 1,100,49,1.6631 *10.1.2.1!"SEP0024C4454ADB
- snl p -
r i : * .Vis'isif! atarDaRaq: daReq.partitionSearchSpace (07ccc3e-da16-aa49-
1235 -ec6a0574Cl) ,
!-:].-ij-. -if, ;- '.i;;i;j- 1 3;; =- ? J r iii-J iHg F.i[.--rgri! :; I't.is-; J.;,l I F-r. :Hj I:!; V=::K5 !.''
partitionSearchSpacsStrlng I80_Eioergancy_Pt: HQ_Intl_Pt:HQ_LD PttHQ LocalPt: I
nternalPt) 1,100, 49, l.S631'10.1.2 .12"SEP0024C44 54ADB
Digit Analysis: atarPaPeq: Hatching Legacy Numeric,
digita=B05125552001,1,100,49,1.6631"10.1.2.12'SEPC02 4C445 4AI>B
Digit Analysis: getDaRee data: daRes.SBType-[01 Intercept Dbmr.sstype-(0),
TPcouot-101, DAMR.NotifyCouat.101 ,
DaRes.No tlfyCount* [01 .1,100,49,1.6>S31"10.1. 2 .12' SEP0024C4 454ADa
Digit analysis: matcMpi-'l-,fqcn-"5115552001", CQ--2001-, plv-"B",
psu-'HQ EmargeDcy_Pt:HQ_Inti_PtiBQ_LD PtiEQ Local Pt ilnternalpf,
"".: ils.'irs-iiS ss- ::,' Er. >;-.'n-.-^ i~:. :ti? inti Vt ;l!=j Li? Vt-bQ i,r,-iil fi. jlnlirustl st "
. dd=-60512555200 1".dac."I")|1.100,49,1.6631"10.1.2.12"SEP002 4C445 4ADB
Digit analysis: potentialMatchea.NoPotantlalMatchesExiBt|
1.10O.19.1.6 631_10.1.2.12-SEPO024C4454ADB
When dialing 5122001 at the IP phone, the learned pattern 5122XXX is marked UnReachable
in the Cisco Unified RTMT Learned Pattem report.
The traceoutputshowsdigit analysisthat highlights the CSSthat is used at the phone. Then
digit analysis showsthe Matching LegacyNumericevent. Notethat thedigits havechanged
from5122001 to 805125552001. This means that digit analysis is aware of the PSTN failover
requirement and has appliedboththeToDIDrules and the PSTNprefix. As ToDIDrules have
not changed fromthe previous case, the PSTN prefix of 80 must have been configured at the
CCD requesting service.
This PSTNprefix 80 is added to 5125552001, making up the wrong called number
805125552001 for the CCDPSTN failover. The call fails because no route pattern is found that
can route this call further.
The PSTN prefixdoes not showin the CiscoUnifiedRTMTLearnedPattern report. In addition
to visiting the Cisco UnifiedCommunications Manager AdministrationCCD requesting service
configuration page, tracing is the only available way to identify this issue.
CCD PSTN Failover Unconditionally Applies
This figure describes how to troubleshoot the issue when CCD PSTN failover applies also
during periods when thc hosted pattern would be reachable on-net.
CCD PS lover Un
The problem:
* Calling always takes routing through CCD PSTN failover
regardless of pattern status.
The cause:
* Trunk type (H.323 or SIP) to reach the pattern is not
configured or not selected
l.- - ------v .-.:--.:.
riimrnn.
MKtlNMI
EaUsai,
This issue is common when thc CCD requesting service is misconfigured. particularly when the
trunk type that is being advertised to reach the learned pattern is not selected in the CCD
requesting service.
Note that in the Cisco Unified RTMT Learned Pattem report, the patterns 5122XXX and
52I3XXX are reachable by using the SIP protocol. If the SIP trunk, enabled for CCD. is not
selected for the CCD requesting service as shown in the figure, the calling to 5122XXX and
5213XXX will always fail over to PSTN regardless of the status of the learned pattern
(Reachable).
Simply, if a pattern is advertised as 11.323-reachable, an 11.323 trunk has to be selected for the
CCD requesting service. If a pattern is advertised as SIP-reachable, a SIP trunk has to be
selected for the CCD requesting service. Configure both trunk types for the CCD requesting
service at the same time to avoid this issue.
Trouble shooting Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems, Inc
Cisco Unified SRST CCD Considerations
This figure describes the CCD considerations for Cisco Unified Survivable Remote Site
Telephony (SRST).
Cisco Unified SRST CCD
Considerations
Cisco Unified SRST subscribes to the CCD service but does
not publish any patterns.
During WAN failures, Cisco Unified SRST uses learned
patterns to transparently reroute calls over the PSTN.
Cisco Unified SRST can also participate in the SAF network and interact with the SAF
forwarder as a SAFinternal client. Cisco Unified SRSTsubscribes to the CCDservice likeany
other client, but because of the natureof howCiscoUnifiedSRSToperates, SRSTdoes not
publishany patterns. For a typical consumer of theCCDserviceduringWANfailures, Cisco
Unified SRST routes calls that are based on the patterns that were learned fromthe SAF
network.
Summary
The CCD feature leverages the SAF network service to
facilitate dynamic provisioning of intercall agent information.
The most common issues of CCD include patterns that are
not being learned, patterns that have been learned but are
unreachable, or PSTN failover that does not work for learned
patterns or that does not stop when it is no longer needed.
In this lesson, you have learned to explain the common issues that relate to CCD as an
application of SAF and identify the most likely causes of these issues.
References
For additional infonnation, refer to these resources:
Cisco Unified Communications Manager Features and Services Guide. Call Control
Discovery at
http: \\\\\\.ei>co.eom en LS parmer docs.'\'oiee_ip cumm/cucnVadmin 8 0 2 cemfeui
fêlilk,oniiold]êo\er\ .html
Cisco Unified Communications System 8.x SRND - Call Routing and Dial Plan
Distribution Using Call Control Discovery for the Service Advertisement Framework at
http: viwweisco.com en US docs voicc_ip eominxucnv'srnd:''Kv'niodeKhiml"\\p! lOliS-l'
Module Summary
Module Summary
The most common issues with the SAF network are that SAF
forwarders cannot establish their adjacency the SAF client
doesnot interact with theSAF forwarder, and SAF forwarders
do not exchange call control information.
Major issues of CCD include patterns that are not being
learned, patterns that are learned but are unreachable PSTN
failover not working for learned patterns or not stopping when
it is no longer needed.
In this module, you learned how to solve the common issues of aService Advertisement
framework (SAF)-enabled network and Call Control Discovery (CCD).
References
For additional information, refer tothese resources:
Cisco IOS Service Advertisement Framework Configuration Guide at
httpr w.cisco.com en/US'does/ios/saf/configumtion/guide/
sal eg pslu591 TSD Products Configuration Guide Chapter.html
Cisco Unified Communications Manager Features and Services Guide, Call Control
Discovery at
http: uwu.cisco.com en/US partner/docs voiceJP comm'cucm/admin.'8 0 2'ccmfcat
tscallcontroldiseovery.html "" ~
Cisco Unified Communications System 8.x SRNDCall Routing and Dial Plan
Distribution Using Call Control Discovery for thc Service Advertisement Framework at
http: www csco.com enUS docs-voicc.ip.conmTcucm/snid/iix/modcls.htmlfrwpl 101849
" Z '. i-T-wf-iir-ci c n 2010 Cisco Systems. Inc
4-72 Troubleshooting Cisco Unif,ed Communications (TVOICE) vB 0
Module Self-Check
Use thc questions here to review what you learned in this module. The correct answers and
00 Which of the following arc three typical internal SAF clients? (Choose three.) (Source:
Troubleshooting the SAF)
A) Cisco IOS voice-enabled gateway
B) Cisco UnifiedCommunications Manager Express on a Cisco Integrated
Services Router
C) Cisco Unified Communications Manager
D) Cisco Unified Border Element on Cisco Integrated Services Router
E) Cisco Unified Border Element on any Cisco gateway platform
F) Cisco Unified SRST on Cisco Integrated Services Router
G) Cisco gatekeeper
Q2) SAF is a SAP that uses EIGRP for service message transport enablement and EIGRP
IP routing should be run in the network for most efficient SAF operation. (Source:
Troubleshooting the SAF)
A) true
B) false
Q3) SAFprevents advertisement loops by usingthe and
__. (Source: Troubleshooting the SAF)
Q4) Which two statements about SAF are correct? (Choose two.) (Source: Troubleshooting
the SAF)
A) It is mandatory to associate a SAF client with at least two SAF forwarders for
redundancy.
B) To exchange call-routing infonnation properly with the SAF network, a SAF
client has to be properly registered to the SAF forwarder.
C) Once a SAF client is properly registered to the SAF forwarder, the client can
advertise service, but it cannot request service until the SAF forwarder
registers with at least one additional SAF client.
D) To verify the registrationof the SAF forwarder, Cisco Unified Serviceability
must be used.
E) To verify the registration of the SAF forwarder, thc Cisco Unified Real-Time
Monitoring Tool (RTMT) can be used.
Q5) What are three critical parameters that are used when establishing a SAF external client
to SAF forwarder relationship? (Choose three.) (Source: Troubleshooting the SAF)
A) EIGRP process ID
B) client label
C) SAF client IP address
D) SAF forwarder IP address
E) SAF forwarder name
F) TCP port number
G) UDP port number
2010 Cisco Systems, Inc SAF and CCD Issues
Q6) Match each Cisco Unified Communications Manager CCD configuration element to its
characteristics. (Source: Troubleshooting the CCD)
A) SAF Trunks
B) Hosted DN Group
C) Hosted DN Pattern
D) CCD Advertising Service
E) CCD Requesting Service
1. DN or DN range to be advertised. Configured with PSTN failover strip
digits and PSTN failover prepend digits. If not configured, Hosted DN
Group configuration is used. Refers to Hosted DN Group.
2. Configured with PSTN failover strip digits and PSTN failover prepend
digits. Applied to Hosted DN Patterns.
3. Configured with route partition, learned pattern prefix, and PSTN prefix.
Refers to SAF Trunks.
4. One SAF SIP trunk and one SAF H.323 trunk can be configured. They are
not configured with a destination IP address. Thc rest of thc configuration
is like normal SIP and H.323 trunks.
5. Refers to Hosted DN Group, SAF SIP trunk, and SAF 11.323 trunk.
Q7) Which command is used to verify whether patterns are learned at the Cisco Lnified
Communications Manager Express client? (Source: Troubleshooting the CCD)
A) show eigrp saf pattern
B) show voice ccd pattern
C) show voice saf dn-block
D) show voice saf dndb
Troubleshooting Cisco UnifiedCommunications (TVOICE) v8 0 2010 Cisco Systems, Inc
Ql) B.D.I'
02) B
Q3) Diffusing Update Algorithm (DUAL) and splil horizon
Q4) B. t-
05) 1). I). F
Q6) 4-A.2-B, I-C.5-D. 3-E
Q7) D
4-76
, _, , ,Ti/nir-c\ ,=bn 2010 Cisco Systems. Inc
Module 5
Features and Application
Issues
Overview
This module discusses how to troubleshoot the key feature and applications of Cisco Unified
Communications Manager. For each application discussed, it reviews the architecture,
functionality, and configurationelements. Then it lists the most common issues, outlines the
probable causes, and names options to find the remedies.
Module Objectives
Upon completing this module, you will be able to troubleshoot issues that are related to Cisco
Unified Communications Manager features and applications. This ability includes being able to
meet these objectives:
Explain the common issuesthat are relatedto CiscoDeviceMobility and identifythe most
likely causes of these issues
Explain the common issues that are related to Cisco Extension Mobility and identify the
most likely causes of these issues
Explain the common issues that are related to Cisco Unified Mobility and identify the most
likely causes of these issues
Explain the common issues that are related to native Cisco Unified Communications
Manager presence and identify the most likely causes of these issues
5-2 Troubleshooting Cisco Unified Communications (TVOICE) v8.0 2010 Cisco Systems Inc
Lesson 1
Troubleshooting Device
Mobility Issues
Overview
The Cisco Unified Communications Manager Device Mobility feature dynamically changes
important location settings- - suchas callingsearchspace(CSS), region, date and timegroup,
and Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST) referencefor
roaming devices. CiscoUnifiedCommunications Manager uses devicepool settingsto
implement newparameters whenthe phoneroamsawayfromits homelocation. Administrators
no longer need to reconfigure location settings when a phone changes location. This lesson
discusses potential issues that can be experienced with Device Mobility.
Objectives
Upon completing this lesson, you will be able to explain the common issues related to Device
Mobility and identify the most likely causes of these issues. This ability includes being able to
Briefly review Device Mobility operation, configuration, and considerations when using
single or multiple Device Mobility groups and using local route groups and name general
issues that users experience when using Device Mobility
Describe how to troubleshoot most typical IP infrastructureproblems in Device Mobility
Describe howto troubleshoot most typical Device Mobility configuration mismatches that
lead to selecting incorrect call codecs, incorrect gateway, incorrect CAC operation,
incorrect Cisco Unified SRST operation, and selection of incorrect media resources
Describe how to troubleshoot typical Device Mobility call-routing problems that lead to
calls blocking, inefficient call routing, and incorrect AAR operation
Describe how to troubleshoot typical Device Mobility call privilege problems that lead to
incorrect calling permissions
Device Mobility and Its General Issues
This topic briefly reviews Device Mobility operation, configuration, andconsiderations when
you use single or multiple Device Mobility groups and local route groups and names general
issues that users experience when they use Device Mobility.
,&
jevn.;e M
Main Sile Cisco Unified
IP Subnet l
10 128 0 0/16 x
Device poof mam:
Region
Locaton
SRST teference
Deuce MoMilyCSS
AAR CSS
AARgrouo
PSTN
Device Mobility
Miwt^dk$M&JSM*& U^m^^xi
&>
Remote Site
Remote
Gateway
7*?>K
f IP Subnet
192 168 10.0/24
Device pool remote:
' Region
* location
SRST reference
Device MobilityCSS
* AAR CSS
* MS groue
Device mobility canbeused in multisite environments with centralized call processing. The
feature supports any Skinny Client Control Protocol (SCCP) or Session Initiation Protocol
(SIP) device thatcanbeconfigured inCisco Unified Communications Manager.
De\icc Mobility allows users and their IPphonestypically CiscoUnified Wireless IPPhones
or Cisco IP Communicator to roam between sites.
When the device isadded tothe network of a roaming site, the device is first assigned an IP
address. Because IP networks differ on a per-site basis, Cisco Unified Communications
Manager can determine the physical location of the IPphoneaccording to its IPaddress.
Cisco Unified Communications Manager reconfigures iheIPphone with site-specific settings
that are based on the physical locationof the phone.
As the figure shows, the location-dependent parameters, such as roaming-sensitive settings and
Device Mobility-related settings, are configured at devicepools. The phone, whichis
associated with a dc\iee pool, uses an IPsubnet. Cisco Unified Communications Manager uses
this information to select theappropriate device pool andapply thelocation-dependent
parameters. With the introduction of Device Mobility. Cisco Unified Communications Manager
isaware of the physical location ofa device and applies the appropriate location-specific
configuration byselecting the corresponding device pool.
Troubleshooting Cisco Unrfied Communications (TVOICE) v8 0
Dynamic Phone-Configuration Parameters
There are two types of phone-configuration parameters that DeviceMobility canassign
dynamically.
Dynamic Phone-Configuration
Parameters
Device Mobility can apply two types of phone settings:
Roaming-sensitive settings
- Date and time group
Region
Cisco Unified SRST reference
Physical locations
Device Mobility group
Device Mobility-related settings
Device Mobility CSS
AAR CSS and AAR group
- Calling-party transformation CSS
- Called-party transformation CSS
DeviceMobility can reconfigure site-specific phone-configuration parameters according to the
physical location of the phone. DeviceMobility does not modifyany user-specific phone
parameters or any IP phone button settings, such as directory numbers.
Thc phone-configuration parameters that can be dynamically applied to the device
configuration are grouped in two categories:
Roaming-sensitive settings
Date and time group
Region
Location
Connection monitor duration
Network locale
SRST reference
Media Resource Group List (MRGL)
Physical location
Device Mobility group
2010 Cisco Systems. Inc Troubleshooting Cisco Unified Communications Manager Features and Application Issues 5-5
Device Mobility-related settings
Device Mobility CSS
Automatedalternate routing (AAR) CSS and AARgroup
Calling-party transformation CSS
Called-party transformation CSS
Roaming-sensitive settings do not affect call routing. However, Device Mobility-related
settings directly affect call routingby modifying the device CSS, AARgroup, and AAR CSS.
Depending on thc implementation of Device Mobility, onlyroaming-sensitive settings or both
roaming-sensitive settings andDevice Mobility-related settings canbeapplied to a roaming
phone.
5-6 Troubleshooting Cisco Uoified Communications (TVOICE) v8 0 2010 Cisco Systems. Inc
Device Mobility Configuration Elements
This section reviews the configuration elements of device mobility.
Device Mobility Configuration
f -.-*
c *.^.f.'.ra
!. - * r .E*?
Device pool
Device mobility info
Physical locaion
Device mobility giuip
5*3fl:,.,Jv\7 ' :
'in*1 > *
t* - - _:-_--
Oeines a set d common characteristics lor
devices. Ill a device pool contains only device-
and locatlan-relaleii information. One device
pool must be assigned lo each device.
Spacifles an IP subnet and associates il with
one or more device pools. Multiple device
mobility info instances can be associated with
one device pod.
The physics) location is a lag assigned to one or
more device pools. This tag is used to determine
whelher a device is roaming wilhin a physical
local ion or between physical locations.
The device mobility group is stag assigned to
one or more device pools This lag is used to
determine whether a device is roaming within a
device mobility group or between device mobility
groups.
This table lists the Device Mobility-related configuration elements and reviews their function.
Thc newly introduced elements are device mobility info, physical location, and device mobility
group.
Thc device mobility info is configured with a name and an IP subnet and is associated with one
or more device pools. Multiple device mobility info elements can be associated with the same
device pool.
The physical location and the device mobility group arc tags; they are only configured with a
name and do not include any other configuration settings. Both are nonmandatory device pool
configuration parameters: at the device pool, one or no physical location and one or no device
mobility group can be selected. These parameters are used to determine whether two device
pools are at the same physical location or in the same device mobility group.
2010 Cisco Systems, Inc. Troubleshooting Cisco Unitied Communications Manager Features and Application Issues 5-7
Relationship of Device Mobility Configuration Elements
This example reviews howthe Device Mobility configurationelements relate to one another.
Hk j in* is'.ip or btviL-r ivUtbil \\
uonnourauon tie merits
i :
"l .
SJ1 dmi
(10 1 1 0/24)
SJ A dp
(Building A)
SJ2 dmi
(10 1 2 0/24)
s
SJ B1 dp
(Building B)
v %
..! SJ pi ;: \ ^\
'. | (SJ-campus) m\ us dmq H
A ;;:'; ^---"---"
z.
SJ3 dmi
(10 1 3 0/24)
SJ B2 dp
(Building B)
NY dmi
(10 3.1.0/24)
NY_dp
**.l NYpI j! i
i (NY-campus) -I i
- - - , 1
J ;
LON dmi
(10 10 1 0/24)
LON_ dp
'.. +' LON pi M ' GB dmg m
j (LON-campus) ! * J 1
1 '-
----- - -- i .
The figure shows five device mobility info elements:
SJldmi: The IP subnet of this device mobility info is 10.1,1.0 with 24-bit subnet mask.
This de\ice mobility info is used at Building A of the San Jose campus and is associated
with SJ_A_dp.
SJ2_dmi: The IP subnet of this device mobility info is 10.1.2.0 with 24-bit subnet mask.
This device mobility info is used at Building B! of the San Jose campus and is associated
ttiihSJ Bl dp.
SJ3_dmi: The IP subnet of this device mobility info is 10.1.3.0 with 24-bit subnet mask.
Like SJ2 dmi. this device mobility info is used at Building BI and is associated with
device pool SJB1 dp. The device mobility info is also used at Building B2 and is also
associated with device pool SJ_B2_dp.
NY_dmi: The IP subnet of this device mobility info is 10.3.1.0 with 24-bit subnet mask.
This device mobility info is used at the New York campus and is associated with device
poo! NY dp.
LONdmi: The IP subnet of this device mobility info is 10,10.1.0 with 24-bit subnet mask.
This device mobility info is used at thc London campus and is associated with device pool
LON_dp,
Device pools SJAdp, SJ_ B1 dp. and SJ_B2_dparc all configured with the same physical
location the SJ pi because they are all used for devices that are located at the San Jose
campus.
Device pool \Y_dp. which serves the New York campus, is configured with physical location
NY_pl; device pool LON dp. which serves the London campus, is configured with physical
location LON_pl.
Trjubleshootmg Cisco Unified Communications (TVOICE) vB.O ) 2010 Cisco Systems. Inc
All device pools that are assigned with a U.S. physical location SJ_A_dp, SJ_Bl_dp,
SJ_B2_dp, and NY_dpare configuredwith device mobility group US_dmg. Therefore, all
U.S. device pools are in the same device mobility group. The London campus is in a different
device mobility group: the GBdmg.
In summary, the U.S. device mobility group consists of two physical locations: San Jose and
New York. At San Jose, IP subnets 10.1.1.0, 10.1.2.0, and 10.1.3.0all with 24-bit subnet
maskare used; New York uses IP subnet 10.3.1.0 with 24-bit subnet mask; and London is
configured with IP subnet 10.10.1.0 with 24-bit subnet mask.
Based on the IP address of an IP phone, Cisco Unified Communications Manager can
determine one or more associated device pools and the physical location and device mobility
group of the device pools. If an IP phone uses an IP address of IP subnet 10.1.3.0 with 24-bit
subnet mask, two candidates for the device pool exist. However, in this example, the physical
location and device mobility group are the same for these two device pools.
>2010 Cisco Systems. Inc. Troubleshooting Cisco Unified Communications Manager Features and Application Issues
When and How Phone Configuration Is Modified
This section reviews howDevice Mobility works and how it modifies phone configuration.
Each phone is configured with a device pool (the home
device pool).
IP subnets are associated with device pools.
Ifthe IP address of the phone matches a configured IP
subnet, one of the associated device pools is selected (load
shared).
If the selected device pool is different from the home device
pool the following settings of the two device pools are
checked:
If the physical locations are different, the roaming-
sensitive settings of the roaming device pool are applied.
If the device mobility groups are the same, the device
mobility-related settings of the roaming device pool are
also applied.
In all other cases, the home device pool configuration is
applied.
As discussed earlier, each phone is configured with a device pool. This device pool is the home
device pool of the phone.
You can associate IP subnets with device pools by configuring device mobility info elements.
If a phone for which Device Mobility is enabled registers with Cisco UnifiedCommunications
Manager by usingan IPaddress that matches an IP subnet that is configured in device mobility
info, the following happens:
The current dev ice pool is selected as follows:
If the device mobility info is associated with the home device pool of the phone,
then the phone is considered to be in its home location. Device Mobility will not
reconfigure thc phone.
It the device mobility info is associated with one or more device pools other than the
home device pool of the phone, then one of the associated device pools is selected,
based on a load-sharing algorithm round robin.
If the current device pool is different fromthc home device pool, the followingchecks are
performed:
It the physical locations are thc same, then the phone configuration is not modified.
It the physical locations are different, then the roaming-sensitive parameters of thc
current -in other words, the roaming - -device pool are applied.
if, in addition to having different physical locations, the device mobilitygroups are
the same, then Dev ice Mobility-related settings are applied in addition to the
roaming-sensitive parameters.
Insummary, the roaming-sensitive parameters are applied when the physical location of the
current device pool isdifTerent from thephysical location of the home device pool, as is the
casewhen roaming between physical locations. TheDevice Mobility-related settings are
applied inaddition to theroaming-sensitive parameters when thephysical locations are
different and the devicemobilitygroups are the samethat is, when roamingbetweenphysical
locations within the same device mobility group.
Consequently, you shoulduse physical locations anddevicemobilityas follows:
Configure physical locations insuch a waythat coder-decoder (codec), selection, andCall
Admission Control(CAC) trulyreflect the current location of the deviceand that local
Cisco Unified SRST references and local media resources at the roaming site are used
instead of those that are located at thc currently remote home network. Depending upon the
network structure and allocation of services, you might define physical locations that are
based on a city, enterprise campus, or building.
Configure a devicemobilitygroupto definea groupof sites that have similardialing
patterns or dialing behavior. Device mobility groups represent thehighest-level geographic
entitiesin your network. Depending uponthe network sizeand scope, your devicemobility
groupscouldrepresentcountries, regions, statesor provinces, cities, or other entities,
Because Device Mobility-relatedsettings, which are applied only when roaming within thc
same device mobility group, affect call routing, you should set up different device mobility
groups whenever roaming users should notbeforced toadapt their dialing behavior. Inthis
case, when roaming between different device mobility groups, thephone configuration
parameters that affect call routing the DeviceMobility-related settings arc not modified.
2010 Cisco Systems, Inc. Troubleshooting Cisco Unified Communications Manager Features and Application Issues 5-11
Device Mobility Considerations
5-12
This section reviews primary facts that you should consider when implementing Device
Mobilitv.
Vir.fi ui,
Rojming-sensilive settings
Ensure the use of local media resources and Cisco Unified SRST
references
Ensure correct use of codecs and CAC between sites
Always apply to roaming devices
'-:. !'.-;. affectcal routng
Which gatewayto use for PSTNaccess and AAR PSTNcalls(device
CSSand AAR CSS], howto compose ihe AAR number(AAR group).
Changes .. ' r'.." k,<- =,: .-. ,=v .-. (forexample, different
PSTNaccess codes, differentPSTNnumbering plans).
Which number format to display on the phone.
Users mighlbe confusedbyhaving their homeextensionbut being
required lo followdial rules of roaming site.
this behavioris. . - - it-.=. suppress application of device mobility-
related settings by assigning ,ir..-. >t '...<- '-,-!.. -.; ..-ii;i=..
Notapplyingdevice mobility-related settings mighl lead to v,l>;>m= -..si
Roaming-sensitive settings ensure that the roaming device uses local media resources and
Cisco Unified SRST references. In addiiion, these settings ensure the correct use of codecs and
CAC between sites, Typically, this behavioris alwaysdesiredwhena deviceroams between
different sites. However, the behavior is not required when the device moves only between IP
subnets within the same site. Therefore, the recommendation istoassign all device pools that
are associated with IP subnets device mobility infoelements -that are used at the same site to
the same physical location, fhis setup results inphone configuration changes only when the
phone roams between sites physical locations and when the phone is moved only between
different networks of the same site.
Device mobility-related settings affect call routing. The application of thcdeviceCSScauses
AAR group andAAR CSS calls toberouted differently. Thesettings at theroaming device
pool determine which gateway will beused for public switched telephone network (PSTN)
access and AAR PSTNcalls depending on the device CSS and AARCSS and how the
number that will beused for AAR calls iscomposed depending onthe AAR group.
Such changes can result indifferent dialing behavior. For instance, during roaming between
countries, the PSTN access codemight bedifferent andPSTN numbering plans, such as how to
dial international, might apply. As an example, to dial the Austrian destination -+43 699 I89(1
0009 from Germany, users dial 000 43 699 1890 0009 but dial 9 01 I 43 699 1890 0009 from
the United States.
A German user who is roaming, with a softphonc, to thc United States might be confused by
the need to use United States dialing rules, such as using the access code 9 instead of 0 and 011
instead of 00 for international numbers, to reach a home extension. If this dialing behavior is
not desired, then the application of device mobility-related settings should be suppressed. To do
so, assign device pools that are to be used at sites with different dialing rules into different
Device Mobility groups in different physical locations. Then, when the user roams from
Germany to the United States, all the roaming-sensitive settings to use local media resources
and Cisco Unified SRST gateways and to apply codecs and CACwill be applied correctly,
but the device mobility-related settings will not apply. Although the user moves to another site,
thc phone will use the PSTN gateway and dial rules of its home location. The user does not
have to adapt to the dial rules of the currently local site.
2010 Cisco Systems, Inc. Troubleshooting Cisco UnifiedCommunications Manager Features and Application Issues 5-13
Device Mobility and CSSs
This section discusses how CSSs are managed when device mobility is being used.
5-14
Device Mobility never modifies line CSS.
Device CSS is modified only when roaming between physical
locations and within the same device mobility group:
Operation when using the line or device CSS approach;
* Line CSS is not modified; CoS settings are kept.
' Device CSS is modified, allowing local gateway
selection by applying CSS of roaming device pool.
When using the traditional CSS approach (only one CSS
at phone), use the device CSS instead of the line CSS to
allow Device Mobility to modify the CSS.
-'-' .w is configurable only at the device and, therefore, is
always correctly >. ;..:.=.,>.. when roaming between physical
locations and v.;t''.t. ?,-i=, i:-."n<- .V'-:cc rroh :i!v ;ir,;>ii,>
An IP phone can be configured with a line CSS and a device CSS. If both exist, the partitions of
the line CSS are considered before the partitions of thc device CSS when routing a call.
These two CSSs allow the use of the line or device approach for implementing calling
privileges and the selection of a local gateway for PSTN calls. With the line or device
approach, all possible PSTN route patterns exist once per location, configured with a site-
specific partition. This partition is included in the device CSS of the phones and, therefore,
enables the use of a local gateway for PSTN calls. To implement class of service (CoS), PSTN
route patterns that should not be available to all users for example, international calls, long
distance calls, or all toll calls are configured as blocked route patterns and are assigned to
separate partitions. The line CSS of a phone then will include thc partitions of those route
patterns that should be blocked for that phone. Because the line CSS has priority over the
device CSS, thc blocked pattern takes precedence over the routed pattern, which is found in a
partition list at the device CSS.
Device Mobility never modifies the line CSS of a phone.
Device Mobility docs change the device CSS and thc AAR CSS of a phone when the phone is
roaming between physical locations within the same device mobility group.
If you use the line or device approach which is recommended for implementing CoS in a
multisite environment this is the operation:
The line CSS implements CoS configuration by permitting internal destinations other
phone directory numbers, access to features such as Call Park, and Meet-Me conferences
and blocked PSTN destinations. Because Device Mobility docs not change the line CSS,
the CoS settings of the device are kept when the device is roaming.
Troubleshooting Cisco Unified Communications (TVOICE) v8 0 )2010 Cisco Systems Inc
The device CSS is modified when roaming within the same device mobility group. In this
case, the device CSS that is used at thc home location is replaced by a device CSS that
applies to the roaming location. This device CSS will refer to the local gateway of the
roaming site instead of to the gateway of the home location.
If you use thc traditional approachonly one CSS that contains CoS and gateway selection
then you must use the device CSS. Device Mobility cannot modify the line CSS, which has
priority over the device CSS. However, Device Mobility can modify the device CSS.
The AAR CSS is configurable only at the device level and, therefore, is always correctly
replaced when roaming between physical locations within the same device mobility group.
2010 Cisco Systems, Inc Troubleshooting Cisco Unified Communications Manager Features and Application Issues 5-15
Device Mobility Example with Different Device Mobility Groups
This tigure demonstrates anexample of Device Mobility with different device mobility groups.
'ice Mobility " . mile with
BR Phone i
Device CSS BR
BR Phone [Hoir.e)
Device CSS BR,
uup HQ-GW
Route Pattern
9
Partition. Branch
Route List
Route Group; BR-GW
Device pool HQ
Physical location HQ
Device mobility group. HO
(Device Mobility CSS. HQ)
Device pool. BR
Physical location BR
Davicp mobility group BR
(Device Mobility CSS BR)
*HQ - headquarters
"BR = branr.h
HQ EU numbering plan, BR'NANP;
BR user roaming to HQ uses NANP dial rules and BR gateway.
The example showstwo sites. Theheadquarters site, IIQ, is in Europe; the branch site, BR, is
in the United States, Separate route patterns, representingthc different dial rules, are configured
indifferent partitions. Thc CSSof headquarters phones provides accessto the IIQgateway; the
CSS of branch phones provides access to the BR gateway.
Device mobility is configured with different device mobility groups. This approach allows a
branch user who is roaming to headquarters to use the home dial rules. Device Mobility does
not update the device CSS so the CSS still provides access to the branch route pattern 9.w;. As
a consequence, the BR gateway is used for all PSTN calls.
mm Device Mobility Example with Same Device Mobility Group
The figuredemonstrates an exampleof DeviceMobility withthe samedevicemobilitygroup.
Device Mobility Example with Samt
Device Mobility Group
Device pool: HQ
Physical location: HQ
Device mobility group:
Worid
Device Mobility CSS: I
Device pool BR
Physical location: BR
Device mobility group1
World
(Device Moblity CSS
BR)
HQ EU numbenng plan. BR NANP;
BR user roaming to HQ must use EU dial rules and uses HQ gateway.
"HQ - headquarters
"BR = branch
This example is identical to the previous example, with one exception: This time, the device
mobility group of the home and the roaming device pool are the same.
When a branch user roams to headquarters, the device CSS of the phone is updated with the
CSS of thc roaming device pool. In the example, the BR CSS changes to the HQ CSS.
Therefore, the phone has access to the HQ partition, which includes PSTN route patterns in
European dialing format 0.!.
The roaming user must followEuropeandial rules. Calls to 9.@ are no longer possible. This
configuration allows the branch user to use the HQ gateway when roaming to headquarters.
>2010 Cisco Systems, Inc Troubleshooting Cisco Unified Communications Manager Features and Application Issues 5-17
Interaction of Local Route Groups and Device Mobility
Cisco Unified Communications Manager Version 7.0 introduced the Local Route Group
feature. When using local route groups, gateway selection is totally independent of the matched
routepatternand referenced route list and routegroup. Device Mobility benefits fromthis new
dial plan approach.
Device Mobility benefits from the globalized dial plan
approach:
" Updates of roaming-sensitive settings always apply (no
changes).
1 Local route groups and globalized call routing allowcombination
of both advantages (instead of choosing one ofthem}.
Roaming gateway can be usee
Home dial rules can be used.
No updates of device mobility-relatedsettings (device CSS, AAR
CSS, and AARgroup) is required
Local route group setting in device pool, instead of site-
specific device CSS, selects gateways
Globalized call routing eliminates the need of AAR prefixes
(controlled by AARgroups and AARCSS).
1he use of the Local Route Group feature makes nochanges regarding roaming-sensitive
settings. Theapplication of these settings always makes sense when roaming between sites. The
settmgs have no influence to the gateway selection and the dial rules that a user must follow:
However, the dial plan-related part of Device Mobility changes substantially withthe new dial
plan concept, This concept allows a roaming user to follow the home dial rules for external
calls but use the local gateway of the roaming site.
This scenario is possible because thenew dial planimplements localized call ingress at thc
phone by applying a CSS at the line level. The line CSS of thc phone provides access tophone-
specific translation patterns that normalize toa global format the localized input of theuser.
The device CSS that was used for gateway selection isobsolete because the Local Route Group
feature performs gateway selection.
The AAR CSS and AAR group that are configured at the device level can be the same for all
phones, as long as the AARnumber is always in global format. You can achieve that
requirement by either configuring the external phone number mask or the AAR transformation
mask to E.164 format. Different AARgroupsare no: required becausedifferent prefixes
basedon the location of the twophonesare not necessary, furthermore, different AARCSSs
are not necessary because the gateway selection is not based on different route lists-
referenced from different route patterns indifferent partitions but on the local route group that
is configured at the device pool of the calling phone.
In summary, when you use thc new dial plan features, Device Mobility allows users to use their
home dial rules to access local gateways. The need to apply different device CSSs, AAR CSSs,
AAR groups, and Device Mobility groups no longer exists.
5-18 Troubleshooting Cisco Unified Communications (TVOICE) v8 0 2010CiscoSystems Inc
mmi
Device Mobility Example with Local Route Groups
This section illustrates how Device Mobilitybenefits fromthe newdial plan features that were
introduced with Cisco Unified Communications Manager v7.0.
Device Mobility Example witl
Route Groups
Headquarters
BR Phone I
Line CSS BR
BR Phone (Home]
Line CSS BR .
+ 1 521 555- -
3XXX 4
*s| RG'.BR
MO Translation Paltems
(Partition HO. CSS. System;
000 ' -> DDI PreDot. Prefix *
00! - DDI PreDot, Prefix +49
0! -> DDI PreDot. Prefix +4940
Route Pattern: \+!
Partition. System
ffi
Single Roule List
Default Local Route Group
BR Translation Patterns
(Partition BR. CSS. System
91.[2-9]XX(2-9]XXXXXX -> DDI
PreDot. Prefix +1
9[2-9]XXXXXX -> DDI PreDot,
Prefi x +1408
9011.! -> DDI PreDot, Prefix +
Device Pool:
HO
Physical
Location: HO
Local Route
Group: HO
Device Pool BR
Physical
Location; BR
CSS: BR]
Local Route
Group BR
RG = Route Group
HQ EU numbenng plan, BR: NANP; normalization of localized calf ingress in place;
BR user roaming to HQ uses NANPdial rules and HQ gateway.
This example is based on the previous scenario: Headquarters is in Europe, the branch is in the
United States. A branch user is roaming to Europe.
However, in this example, globalized call routing has been implemented. The line CSS of
branch phones provides access to translation patterns that convert localized call ingress at the
phone: North American Numbering Plan (NANP) to global E. 164 format. European phones
have access to translation patterns that convert European input to global E.164 format.
A single PSTN route pattem \+! -is configured and is in a partition that is accessible by all
translation patterns.
When a branch user roams to headquarters, the line CSS is not modified; no device CSS is
configured at the phone or at the device pool. Also, no device mobility groups are set or are set
differently.
Effectively, no change in matching the translation patterns occurs: The branch user uses NANP
dial rules, just like at home. The number is converted to international format by translation
patterns and matches the only PSTN route pattern. The route pattem refers to a route list that is
configured to use the Default Local Route Group. Thc Default Local Route Group is taken
from the roaming device pool. If the phone is physically located in the branch office, the local
route group is BR; if the phone is roaming to the headquarters site, the local route group is HQ.
As a result, the local gateway is always used for a PSTN call.
>2010Cisco Systems, Inc. Troubleshooting Cisco UnifiedCommunications Manager Features and Application Issues 5-19
Device Mobility General Problems
This section lists the most common Deviee Mobility problcms.
ICO /ior>ti
General problems include:
' Device Mobility disabled (set to Off) on a phone or in service
parameters
Problems with IP subnets and DHCP and how a physical
location is identified
* Problems with CAC and codecs
* Cisco Unified SRST reference problems
" Media resource problems
* Call routing and AAR problems
s CoS problems
When a phone uses Device Mobility and roams away from the home site, the user might
experience the following general problems:
Device Mobility Mode set to Off on a phone or in service parameters
Problems with IP subnets and DHCP, problems with how Deviee Mobility identifies a
physical location, and differences between softphone and hardware IP phone
Problems with CAC and codecs
Cisco Unified SRST reference problems
Media resource problems
Call routing and AAR problems
CoS problems
Troubleshooting Cisco Unified Communications (TVOICE) v8.0 12010 Cisco Systems, Inc
T
few
Troubleshooting IP Infrastructure Problems
This topic describes howto troubleshoot most typical IP infrastructure problems in Device
Mobility.
Troubleshooting Typical IP
infrastructure Problems
Voice Subnet
10.10.0 0/16
DHCP problems:
DHCP server unreachable
Wrong DHCP scope
No gaieway propagated
Softphone problems:
Wrong static IP address
(IP addressing handled at OS)
Data instead of Voice subnet
Wrong pj-one {eUwgf Phone cannot reach
Unified CM'
WAN
192.168.10.0/24
WAN problems:
NAT or PAT address translation
Network connectivity issues
Data Subnet
Hardware phone problems:
Wrong manual network settings
* Unified CM = Cisco Unified Communicalions Manager
The Device Mobility feature depends on thc IP address of the device that registers with Cisco
UnifiedCommunieations Manager. The phone must have a dynamic IP address to use Device
Mobility.
The IP subnet associates with device mobility info elements that connect the phone to thc
appropriate configuration. If device mobility info misconfiguration or mapping exists, the
phone receives the incorrect settings. If a phone with the incorrect static settings roams in a new
site, the settings cannot be dynamically updated to reflect the newenvironment. The phone is
unable to reach Cisco Unified Communications Manager.
In the new site, the phone must receive its IP address froma DHCP server. These are typical
DHCP problems:
The DHCP server cannot be reached. DHCP requests are broadcasts that need to be
propagated across the network to reach the DHCP server.
The incorrect DHCP scope is configured and provides addresses to a limited number of
phones.
DHCP does not propagate the default gateway IP address or, in the case of hardware
phones, TFTP server address.
If the phonewithstatic network settings is transferred to a newsite, the phonecannot adapt to
the new networkingenvironment. Sometimes, hardware phones are configured with static
network settings and DHCP is disabled.
Cisco IP Communicator runs on a laptop. The operating systemmanages the network settings.
Also, the laptop might have static IP addressing or incorrect gateway information, and Cisco IP
Communicator running on that laptop will be unable to reach Cisco Unified Communications
Manager.
>2010 Cisco Systems. Inc. Troubleshooting Cisco Unified Communicalions Manager Features and Application Issues 5-21
In most network designs, computers attach to data VLANs instead of to voice VLANs for
security and QoS reasons. If a computer that is running Cisco IP Communicator is connected to
a data VLAN and its IP address is not included in device mobility info, then Cisco IP
Communicator will not update its Device Mobility settings.
If a phone deviee is assigned an IP address by using Network Address Translation (NAT) or
Port Address Translation (PAT), then thc IP address that is provided during registration might
not match the actual IP address of the device.
If a phone does not work in the new site, also consider WAN network-connectivity issues.
Troubleshooting Typical IP
Infrastructure Problems (Cont,)
How to troubleshoot most typical IP problems in
Device Mobility:
Verify that the phone obtained IP address, gateway, and
TFTP address.
Verify IP connectivity between Cisco Unified Communications
Manager and phone, by using ping.
* Verify DHCP server and scope settings.
* Verify that the network settings on the IP phone are correct.
Consider NAT or PAT IP address translations.
* Make sure that IP address matches device mobility info.
Make sure device mobility info matches correct device pool,
then physical location and device mobility group.
Consider WAN network-connectivity issues.
The figure shows the list of options to troubleshoot typical Device Mobility problems that
relate to IP addressing and new roaming-site recognition.
First, verify that the phone has obtained an IP address, gateway address, andif the phone is a
hardware phoneTFTP address. Depending on whether the phone is a softphone or hardware
phone, this step is performed in the computer operating system or the hardware phone display.
If you suspect that IP connectivity does not exist between Cisco Unified Communications
Manager and the phone, you can use ping from Cisco Unified Platform Administration; if Cisco
IP Communicator is used, use ping from the host computer.
Verify that the DHCP server is configured properly and is operational, and that the scope
settings are sufficient.
Verify that the network settings on the IP phone are correctespecially that the DHCP client is
enabled on the phone and that the client can reach the DHCP server. A phone that uses Device
Mobility should not use static IP addressing.
If NAT or PAT is used, also consider how the IP address is translated.
Make sure that thc IP address that the phone has obtained matches the device mobility info that
is configured for the new site in which the phone roams. Device mobility info is the most
fundamental configuration element to recognize that the phone has been roaming in a new site.
Make sure that the device mobility info matches the correct device pool and that the device
pool matches the correct physical location and device mobility group. These mappings are
important to find and apply the correct contiguration to the roaming phone device.
At that moment when Device Mobility problcms are experienced, the WAN network might be
having connectivity issues. Use IP networking troubleshooting procedures to solve WAN
connectivity problems.
) 2010 Cisco Systems, Inc Troubleshooting Cisco Unified Communicalions Manager Features and Application Issues 5-23
Troubleshooting Device Mobility Configuration
Mismatches
This topic describes how to troubleshoot most typical Device Mobility configuration
mismatches that lead to the selection of incorrect call codecs, incorrect gaieway, incorrect CAC
operation, incorrect Cisco Unified SRST operation, and the selection of incorrect media
resources.
U^hootlnfj li'pncil Ir*- "i< e f bfhi_/ i *i(
Typical configuration problems:
Incorrect codec or CAC parameters cause call-setup issues.
Incorrect media resources exhaust WAN bandwidth.
* Incorrect SRST reference blocks phone registrations.
The device mobility info is configured with a name and an IP subnet. Based on the IP address
of an IP phone. Cisco Unified Communications Manager can determine thc physical location of
the phone and can assign correct roaming-sensitive settings to the phone.
The roaming-sensitive settings are configured under the deviee pool that Device Mobility
assigns to the phone to replace its home configuration settings when a new physical location is
identified. Vou might experience various issues lhat relate to the roaming-sensitive settings
configuration and application.
When a phone roams in a new site, the phone might have call-setup issues that are caused by
incorrect codec settings and location parameters.
When a phone mo\es across the network to the new site, the phone might still use media
resources, such as a conference bridge or transcoder. from the old site. This behavior can cause
unnecessary WAN bandwidth exhaustion and suboptimal use of resources or even
unavailability of the services and features that require media resources.
When a phone moves from another branch site, the phone can have registration problems and
can be isolated when Cisco Unified Communications Manager is unreachable, even if Cisco
Unified SRST is implemented in thc new site.
Troubleshooting Typical Device
Confiauration ft
Verify configuration:
Device mobility info and physical location
to device pool mappings
Device pool mapping
Location or region and MRGL
SRST reference
Device Mobility maps various configuration elements to apply the correct site-specific
parameters to the phone when it roams in another site.
The physical location and device mobility info map to one or more device pools. When Cisco
Unified Communications Manager identifies a new physical location of the phone, roaming-
sensitive parameters should replace the device configuration of the phone. The roaming-
sensitive parameters arc configured under the device pool to which device mobility info and
physical location map. Verify that this mapping is configured as expected.
The device pool references new site-specific locations and regions. If call-setup issues related
to CAC and codecs are experienced, then verify that the device pool matches the location and
region that apply to the new phone site.
The device pool references new site-specific media resources by using an MRGL reference. If
the features that rely on media resources have issues, or if the roaming phone does not use site-
specific media resources, then the device pool probably does not reference the correct site-
specific media resources. Verify the MRGL mapping under the device pool.
If the roaming phone does not register to the site-specific Cisco Unified SRST gateway during
Cisco Unified Communications Manager failure or isolation, then verity that the correct Cisco
Unified SRST reference is configured under the site-specific device pool.
12010 Cisco Systems, Inc. Troubleshooting Cisco Unified Communications Manager Features and Ap plica lion Issues 5-25
Troubleshooting Device Mobility Call-Routing
Problems
This topic describes howto troubleshoot typical Device Mobility call-routing problems that
lead to calls blocking, inefficient call-routing, and incorrect AARoperation.
1 i.'it i th* rt"ij it. vie) Sri im\ I
Typical call-routing problems:
* Incorrect gateway selected
Incorrect number globalization or localization: blocked calls
- AAR does not work
* Incorrect globalized or localized numbers shown on phone
K J
When a phone changes its site, you can experience call-routing problems at the newsite.
Phone device configuration might still refer to old site-specificsettings, such as CSS., dial plan.
AAR. and PSTN gateway.
When the phone configuration is improperlyupdated as the phone transfers to its newsite, thc
PS1Ncalls that originate fromthe phone might be routed via the old site PSTNgateway.
Or, if you use incorrect CSS and partitions, then the called number globalization and
localization processes can refer to the settings of the old site and completely block calls at the
new site.
AAR operation is dependent on howPSTN calls are managed at the site, so AARmight not
work when the phone roams in thc newsite. Applying incorrect location settings can also cause
AAR failure,
New dial plan features have been implemented since the introduction of Cisco Unified
Communications Manager v7.x. Phones can originate calls to E.164 numbers, and selected IP
phones also display calling numbers in correct localized format. The globalization and
localization procedures are site-dependent because they perform digit manipulations that are
based on number formats that are typically used in the PSTN that is close to the site. Therefore,
when a phone roams at another site, number globalization and localization processes might not
relate to the site in which the phone resides at that moment and might produce incorrect
numbers.
Troubleshooting Cisco Unified Communicalions (TVOICE) v8 0 12010 Cisco Systems. Inc;
Troubleshooting Device Mobility
CaM-Routing
SJC phone roams in MUC
Device CSS
Lines Phone
DP SJC
CSS. SJC
AAR CSS SJC
AAR Grp: SJC
CgPTP SJC
COP TP-SJC
CSS: SJC and MUC access to:
Route Patterns
(> Route List > Route Group > Gateway)
Translation Patterns(globalization)
Called Number Transformation Patterns
SJC = San Jose
MUC = Mumcn
DM CSS = Oev.ce Mobility CSS
CgP TP = Callig-Party Transforms Iion CSS
CdPTP = Called-Party Transfo'rnaliori CSS
DMG = Device Mobility Oiouc
Phone
DP. WK.
CSS.fv'i"".
AARCSS: '..Vii
AAR Grp: f.'UC
CgPTP. ivnjf,
CdP TP. MUC
Lines Phone
DP:F.*i'C
CSS: SJC
AAR CSS: SJC
AAR Grp: SJC
CgP TP. SJC
CdP TP. SJC
Munich
Via MUC ojtt".*-'(.ij'
AAR rw/
Via SJC gateway
SJC dial plan
AAR issues
Incorrect CSSimplementation causesmost call-routing problems. The figure illustrates howthe
application of a new, site-specific CSS influences call routing.
The figure shows that the SanJose phoneroams in Munichand the situationin whichCSS is
applied to the device.
The phone is configured with its San Jose-specific configuration. Among other parameters, the
phone device configuration contains the San Jose device pool, San Jose CSS, San Jose AAR
CSS, San Jose AAR group, San Jose calling-party transformation pattern, and San Jose called-
party transformation pattem.
For the Munich site, a specific device pool is configured. This device pool contains Device
Mobility-related settings that affect howcalls are routed at the Munich site.
San Jose-specific and Munich-specific CSSs realize an access to route partitions that are
assigned to the following:
Route patterns for the site-specific dial plan that selects the gateway
Translation patterns for site-specific called number globalization
Called number transformation patterns for site-specific called number localization
Depending on the Device Mobility design, especially on how device mobility groups are
configured, the following things happen to the roaming phone at the Munich site.
When the device mobility group is the same for the San Jose and Munich sites, the device
mobility-related settings from the Munich device pool replace the phone settings. The device
pool. CSS. AAR CSS, and AAR group are set to Munich. Also the calling-party transformation
pattern and called-party transformation pattem are set to Munich.
Consequently, all settings that are now specific to the Munich site will ensure that the Munich
PSTN gateway and dial plan will be used for calls that originate at the roaming phone. AAR
has thc correct, Munich-specific configuration and AAR will work for this roaming phone.
Also, the numbers that are displayed on the phone are in the correct format for the Munich site.
>2010 Cisco Systems. Inc Troubleshooting Cisco Unified Communications Manager Features and Application Issues 5-27
When the device mobility group isnot the same for San Jose and Munich, thc Device Mobility-
related settings are not applied. Thcphone device keeps its San Jose-specific configuration.
Consequently, the PS'fNcalls that originate fromthe roaming phone are routedvia the San
JosePSTN gateway (unless youuselocal route groups with globalized call routing). TheSan
Jose-specific dial plan is used. AAR remains configured with theSanJose-specific
configuration, suchas the PSTN accesscode and prefix, and will not workfor this phonein the
new site. The numbers that are displayed on the phone are in an incorrect format for the
Munich site.
5-28 Troubleshooting Cisco Unified Communications{TVOICE) v8 0 2010 Cisco Systems, Inc
Troubleshooting Device Mobi
Routing Problems (Gont.
Lines Ptione
CSS SJC
DP SJC
CSS <None>
AAR CSS SJC
AAR Grp SJC
CgPTP SJC
CdPTP SJC
Munich
Munich DP
DM CSS M'.u
AAR CSS iV.
AAR Grp. '!'
CgPTP
CdPTP
SJC = San Jose
MUC = Munich
DM CSS = Device Mobility CSS
CgP TP = Calling-Party Transformalion CSS
CdP TP = Called-Party Transformation CSS
DMG = Device Mobility Group
Line CSS
CSS: SJC and MUC access to:
Route Patterns
(> Route Lisl > Route Group > Gateway)
Translation Patterns (globalization)
Line CSS has higher priority than device CSS
Munich
Lines Phone
CSS: SJC
DP: W. !:
CSS: f.HJ ;
AAR CSS: f-U1"
AAR Grp: NVJC
CgPTP'MUC
CdPTP MUC
Lines Phone
CSS: SJC
DP: (','! "..
CSS <None>
AAR CSS: SJC
AAR Grp: SJC
CgP TP. SJC
CdP TP: SJC
Via SJC gateway
SJC dial plan
AAR issues
Via SJC gateway
SJC dial plan
AAR issues
This figure shows the situation when CSS is applied to the line instead of to the device.
The phone brings its San Jose-specific line and device configuration to Munich.
For the Munich site, the specific device pool is configured with the Device Mobility-related
settings.
The San Jose-specific and Munich-specificCSSs that realize access to route partitions are the
same as in the previous example:
Route patterns for a site-specific dial plan that selects the gateway
Translation patterns for site-specific called number globalization
Called number transformation patterns for site-specific called number localization
Depending on the Device Mobility design, especially on how device mobility groups are
configured, the following things happen to the roaming phone at the Munich site.
When the device mobility group is the same for the San Jose and Munich sites, the Device
Mobility-related settings from the Munich device pool replace the phone device settings, such
as device pool. CSS. AAR CSS, and AAR group.
However. Device Mobility never replaces the line CSS, which has precedence over the device
CSS. If the line CSS gives access to the call-routing partitions and as well as to the CoS
partitions, the consequences are as follows.
PSTN calls that originate from the roaming phone are routed via the San Jose PSTN gateway,
and the San Jose-specific dial plan is used. AARwill not work despite its updated configuration
from the Munich device pool because PSTN calls are routed via the San Jose gateway. The
numbers that are displayed on the phone are in the correct format for the Munich site.
When the device mobility group is not the same for San Jose and Munich, the Device Mobility-
related settings are not applied. The phone device keeps its San Jose-specific configuration.
)2010 Cisco Systems, Inc. Troubleshooting Cisco UnifiedCommunicalions Manager Features and Application Issues 5-29
Consequently, the PSTN calls that originate from the roaming phone are routed via the San
Jose PSTN gateway. The San Jose-specific dial plan is used. AAR remains configured with the
San Jose-specific configuration, such as the PSTN access code and prefix, and will not work for
this phone in the new site. The numbers that are displayed on thc phone are in an incorrect
fomiat for the Munich site.
Gail-Routing Problems (Cont.)
CSS SJC and MUC access tn
Route Pallerns
;> RL Local Route Group)
Translation Patterns (globalization)
SJC = San Jose
MUC =lv un.cn
DM CSS Device Mobility ;ss
CgPTP = Calling-Party TransfornBtio CSS
CdPTP = Called-Party Tra 5/rjrmal io css
DWG = Device Moiil.ty Gnpup
Munich
VMMUCg3i'*vv^
MUCsta'p'm
Lines Phone
DP-.--AVC
CSS: SJC
AAR CSS. SJC
AAR Grp SJC
CgPTP SJC
CdPTP SJC
s.
ViaMUO^.y.-.y
SJC dial plan
AARm.gh! w,! k
This figure shows how Local Route Group configuration influences Device Mobility call
routing.
The Local Route Group feature removes gateway selection from CSS and adds a local route
group to the route list and device pool.
The phone brings its San Jose-specific line and deviee configuration to Munich.
Depending on the Device Mobility design, the following things happen to the roaming phone at
the Munich site.
When the device mobility group is the same for the San Jose and Munich sites, the Device
Mobility-related settings from the Munich device pool replace the phone device settings, such
as device pool, CSS, AAR CSS, and AAR group.
PSTN calls that originate from the roaming phone are routed via the local gateway at Munich
and are based on the route list and device pool local route group settings. Also, the Munich-
specific dial plan is used because the CSS was updated on the phone. AAR also works because
its parameters were set to the Munich-specific AAR CSS and AAR group. The separate AAR
group is not even needed because the gateway localization and the use of the local gateway
allow AAR to work properly. The numbers that are displayed on the phone are in the correct
format for the Munich site.
related settings are not applied. The phone device keeps its San Jose-specific configuration.
Despite the San Jose-specific configuration on the phone, the PSTN calls that originate from
the roaming phone are routed via the local PSTN gateway and are based on the route list and
device pool local route group settings. The San Jose-specific dial plan is used. Also, AAR
remains configured with the San Jose-specific configuration, but if the San Jose dial plan and
San Jose AAR CSS permit and if the AAR group contains the prefix that can be applied in
Munich, then AAR can work. Alternatively, if San Jose and Munich use the same AAR group,
AAR can work properly. The numbers that are displayed on the phone are in an incorrect
format for the Munich site.
Troubleshooting Device Mobility Call-Routing Problems
Summary
This section summarizes the Device Mobility call-routing problcms.
Troubleshootinq Device Mobilil
Most Device Mobility call-routing issues are caused by the
following:
Incorrect CSS model implementation
CSS misconfiguration
Incorrect Device Mobility design (device mobility group)
- Incorrect device pool configuration
To troubleshoot Device Mobility call-routing issues, do the
following:
* View Current Device Mobility Settings atthe phone
- Use Cisco Unified Communications Manager Dialed Number
Analyzer and trace to check dial plan results
- Venfy CSS and partition configuration
Venfy device pool configuration
- Confirm trie device mobility group design
The following cause most Device Mobility call-routing issues:
Incorrect CSS model implementation
CSS misconfiguration
Incorrect Device Mobility design, especially if device mobility groups are configured to be
separate or thc same
Misconfigurationof device pool parameters that apply to the phone at the new roaming site
To troubleshoot Device Mobility call-Toutingissues, do the following:
Using thc link i'iew Current Device MobilitySettingson the phone configuration page,
verify the current phone configuration.
Use Cisco Unified Communications Manager Dialed Number Analyzer and trace to check
the dial plan results.
Verify CSS and partition configuration.
Verify the configuration of the device pool that is applied in the roaming site.
Confirm that the device mobility group design is correct.
Troubleshooting Cisco Unitied Communications (TVOICE) v8.0 )2010 Cisco Systems Inc
Troubleshooting Device Mobility Call Privilege
Problems
This topic describes how to troubleshoot typical Device Mobility call privilege problcms that
lead to incorrect calling permissions.
Troubleshooting Device Mobility Caf
Privilege Problems
Typical call privilege problems:
- Denied access to destinations that should be permitted
* Permitted access to destinations that should be denied
San Jose
San Jose policy
UK destination calling denied
Gateway
Munich
Munich policy:
All jihonos wiih f-iimo
frii'QJnyô-'. CoS no irilcrnntsssnal
Another set of problems that Device Mobility users can experience is the problem of CoS.
When a phone roams in a new site, it can carry old CoS permissions that Device Mobility does
not update.
The figure shows two different sites with different CoS policies. San Jose includes two groups
of phone users: employees, who do not have access to international PSTN destinations, and
managers, who are allowed to reach all destinations.
In Munich, all phone users are identical: no one is allowed to reach international PSTN
numbers.
What happens when the manager phone is transferred from the San Jose site to the Munich
site?
Depending on how CSS implements the CoS policies, a user who is calling from this roaming
phone in Munich might not be able to reach formerly permitted destinations, such as the U.K.
PSTN phone in this example.
These CoS problems typically relate to Device Mobility:
Denied access to destinations that should be permitted
Permitted access to destinations that should be denied
i 2010 Cisco Syslems, Inc. Troubleshooting Cisco Unified Communications Manager Features and Applicalion Issues
5-34
Troubleshooting Device Mobility n i
Privilege Problems (Cont.)
SJCjjhone_roair s_in MIIC
'*^Ss
Ceu. eCSS
Munich
SJC = Rm .ns
MUC Wi urt
DM CSS = IP ceMoDihyCSi
DMG -Ue ice MobililyGroup
Lines Phone
DP SJC
CSS
| Munich |
Lines Phone
DP .
CSS
MUC-Lmployees
Lines Pfione
dp r ;.
CSS
1 .i!" \1S5T3CK-!
Phone provides
worse CoS than
expected
Phone provide-:
ocnv-ctC*
The figure shows a situation in which CoS is implemented by using the traditional CSS model.
The CSS that implements manager permissions in San Jose is assigned to the phone device.
The new roaming site is associated with a device pool that contains the Device Mobility CSS
that implements employee permissions in Munich.
Dependingon the device mobility group, the device CSS will either keep its original settings or
will be updated.
If San Jose and Munich use the same deviee mobility group, then the Device Mobility CSS
from the Munich device pool replaces the device CSS.
In this situation, thc manager CSS is gone and the user that uses this roaming phone will not
have access to international PSTN numbers.
When the device mobilitygroup is not the same for San Jose and Munich, the Device Mobility-
related settings are not applied. The phone device keeps its San Jose-specific configuration and
the user can still call international PSTN numbers.
Troubleshooting Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems. Inc.
Privilege Problems (Cont.)
SJCpnonemams_in MUC
Line CSS
Munich
Lines Phone
CSS
DP. SJC
CSS snone>
Line CSS not changed
by Device Mobility
SJC = San Jose
MUC - Munich
DM CSS = Denes Modihty CSS
DMG = Ue tfice Mobil'ly G="oud
Line CSS has higher priority
than device CSS
/
| Munich |
Lines Phone
YES
CSS:
SJC-Manager
DP. mj;;
CSS
MUC-r"!lp:'/yt!ti,>
JÔ
^*i^
Lines Phone
CSS.
SJC-Manager
DP: F-"UC
CSS: <none>
Ptiofie jwov
correct 0o<
oifr-oci CoS
This figure describes the situation in which CoS is implemented by using the line or device
CSS model.
The CSS that implements manager permissions in San Jose is assigned to the phone line.
As before, the new roaming site is associated with a device pool that contains the Device
Mobility CSS that implements employee permissions under Device Mobility-related senings in
Munich.
Examine how the device mobility group affects the final privileges. If San Jose and Munich use
the same device mobility group, then the Device Mobility CSS from the Munich device pool
replaces the device CSS.
But the line CSS is not replaced, so the phone line keeps its original, manager-specific CSS.
In the line or device CSS model, the line CSS always overrides the device CSS, and settings
that are under the line CSS take precedence over settings that are under the device CSS.
Because the manager CSS is still assigned to thc line, the user receives the same CoS as before
and is able to call international PSTN numbers.
related settings are not applied. The phone device keeps its San Jose-specific configuration for
device CSS and line CSS. In this case, the user can still call international PSTN numbers.
As when troubleshooting Device Mobility call-routing issues, do the following when
troubleshooting CoS issues:
Use trace to check the final CoS.
Verify CSS and partition configuration.
Verify configuration of the device pool that is applied in the roaming site.
Confirm that your device mobility group design is correct.
)2010Cisco Systems, Inc Troubleshooting Cisco Unified Communications Manager Features and Application Issues 5-35
Summary
mimai
e General Device Mobility issues include IP infrastructure, CAC
and codec, Cisco Unified SRST reference, media resource,
call routing, and call privilege problems.
IP infrastructure problems include IP addressing, gateway,
TFTP. and NAT or PATproblems.
Device Mobility configuration mismatches cause incorrect
codec selection, call setup and media resources issues, and
incorrect Cisco Unified SRST reference blocking phone
registrations.
' Device Mobility call routing problems include incorrect
gateway selection, incorrect digit manipulation, calls blocking,
and nonoperational AAR.
Device Mobility call privilege problems cause either denial of
access to destinations that should be permitted or permitting
access to destinations that should be denied.
This lesson has explained the common issues that are related to Device Mobilityand identified
the most likely causes of these issues.
References
5-36
Cisco Systems. Inc. Cisco Unified Communications Manager Features and Services iiiiide.
Release tiM(l). April 20(19 and updated April 2010.
http. www cisco.com en l.'S does voice ip uoiiimcucmadmin 8 0 1 comical'
Niid-SOI-cni htm!
Troubleshooting Cisco UnifiedCommunications (TVOICE)v8 0 2010 Cisco Systems. Inc
Lesson 2
Troubleshooting Cisco
Extension Mobility Issues
Overview
In multisite environments, it is common that some users roam between sites on a regular basis.
When such users use phones that are provided at a site that they visit, they miss their personal
settings, such as directory number, speed dials, calling privileges, and Message Waiting
Indicator (MWl). A professional Cisco Unified Communications solution needs to provide a
solution to this problem.
Cisco Extension Mobility allows Cisco Unified Communications Manager user to log into an
IP phone and apply a personal profile, regardless of the device and physical location.
This lesson describes how to troubleshoot Cisco Extension Mobility most common issues.
Objectives
Upon completing this lesson, you will be able to explain the common issues that are related to
Cisco Extension Mobility and identify the most likely causes of these issues. This ability
includes being able to meet these objectives:
Briefly review the Cisco Extension Mobility operation, configuration, and considerations of
managing various phone types and interactions of Cisco Extension Mobility with the two
CSS implementation models and Local Route Group and name the general issues that users
experience when using Cisco Extension Mobility
Explain the Cisco Extension Mobility error messages that can be experienced, describe
their most probable causes, and explain how to troubleshoot Cisco Extension Mobility
login and logout problems
Describe how to troubleshoot typical Cisco Extension Mobility call privilege problems that
lead to incorrect calling permissions
Describe how to troubleshoot typical Cisco Extension Mobility call-routing problems that
lead to calls blocking, inefficient call routing, and incorrect Caller ID
Cisco Extension Mobility and Its General Issues
This topic reviews the Cisco Kxtension Mobility operation, configuration, andconsiderations of
managing various phone types and interactions of Cisco Extension Mobility. Thc topic also
covers the twocalling search space (CSS) implementation models andLocal Route Group and
names general issuesthat usersexperience when they use CiscoExtension Mobility.
Cisco
Unified
Communications!
Manager
Logout
Mam Site
Gaieway
Device profile ot
the user
=-T&ejkxale
\jio MOH auta.
Pry Dintm
Templffle
Sofftey tefntjiala
Lmel 1001
mm CSS
WAN
Remote
Gateway
Remote Site
Login
Roaming User
t
i
/
Device profile of
Bie user:
H
login User
and PIN
- tlHflf locale
Uiei MOH auAo
Source
TwnplatB
* Softkey Lsmâle
Linei 1DG1
LmeCSS
\
Cisco Extension Mobility allows users to log into any phone and get their individual user-
specific phone configuration applied tothephone. Thus, users canbereached at theirpersonal
directory number, regardless of the location or physical phone. CiscoExtension Mobility is
implemented as a phone service andworks within a CiscoUnified Communications Manager
cluster.
1he user-specific configuration is stored in device profiles. Based on thc user ID that is entered
dunng login, Cisco Unified Communications Manager can apply the personal device profile of
the user and reconfigure the phone to use the configurationprofile of the individual user who
logged in.
Two types of configuration parameters are dynamically configured when you use Cisco
Extension Mobility:
User-specific device-level parameters: User-specificphone configuration parameters,
suchas user music on hold(MOH), audiosource, phonebutton templates, softkey
templates, user locales. Do Not Disturb (DND), privacy settings, and phone service-
subscriptions are configured at the device level of an IP phone.
Configuration of phone buttons, including lines: Cisco Extension Mobilityupdates all
phone buttons not only the button types as specified in the phone button template but also
the complete configuration of the phone buttons. This includes all configured lines with all
the line configuration settings, speed dials, service URLs, ('all Park buttons, and any other
buttons that are configured in the device profile that is to be applied.
5-38 Troubleshooting Cisco Unified Communications (TVOICE] v8.0 2010 Cisco Systems, Inc
After successful login, the phone is reconfiguredwith user-specific parameters; other device-
specific parameters remainthe same. If a user is associated with multiple deviceprofiles, the
user must select which device profile to use.
If a user logs in with a user ID that is still logged in at another device, one of thc following
options can be configured:
Allow multiple logins: When this method is configured, the user profile is applied to the
phone on which the user is logging in, and the same configuration remains active at the
device where the user has already logged in. The line numbers become shared lines because
they are active on multiple devices.
Deny login: When this method is configured, the users get an error message. Login is
successful only after the user logs out at thc other device on which the user logged in
before.
Auto logout: Like the deny login option, the auto-logout method also ensures that a user
can log in at only one device at a time. However, this method allows the new login by
automatically logging out the user at the other device.
On a phone that is configured for Cisco Extension Mobility, either another device profilea
logout deviee profile -can be applied, or the parameters, as configured at the phone, are
applied. The logout can be triggered by the user or enforced by the system after expirationof a
maximum login time.
2010CiscoSystems, Inc. Troubleshooting CiscoUnified Communications Manager Features and Application Issues 5-39
Cisco Extension Mobility Configuration Elements Review
This section reviews thc configuration elements that Cisco Extension Mobility uses.
leinei
Device profile
Default device profile
Phona service
litv Lontu
q&i
Stores configuration of physical phones. Configuration parameters
include device-spedfie phone parameters (such as device CSS,
location, or MRGL). user-specific phone parameters |such as user
MOH audio source. DND, or sotlkey template), and (user-specific)
button configuration (such as lines or speed dials).
The end user is associaled with one or more device profiles. The user
ID and Ihe PIN are used to lag into a phone thai usea Cisco Extension
Mobility.
Stores user-specific phona configuration in logical profiles
Configuration parameters include user-specific phone and button
(such as lines and speed dials) parameters The parameters of the
device profit* are applied to a physical phone after a user logs into tie
phone that uses Cisco Extension Mobility.
Slaes ihe default device-conlguration parameters mat should be
applied when the phone model of the device profile of a user is
different from the phone model of the phone on which (tie user logs in
Cisco Extension Mobility is implemented as a phone service.
Hantwara phones and device profiles musl subscribe lo the service
Ihe table lists the extension mobility-related configuration elements and describes their
function. The most-fundamental elements are a phone and an end user.
The phone stores the configuration of physical phones. Configuration parameters include the
following:
Device-specific phone parameters, such as device CSS, location, or Media Resource Group
List (MRGL)
User-specific phone parameters, such as user MOH audio source, DND, or softkey
template
User-specific button configuration, such as lines or speed dials
The end user is associated with one or more device profiles. Use thc user ID and thc PIN to log
into a phone that uses Cisco Extension Mobility.
The configuration elements that are introduced with Cisco Extension Mobility are the device
profile and the default device profile.
The device profile is configured with all the user-specific settings that are found at the device
level of an II* phone -user MOII audio source, phone button templates, softkey templates, user
locales, DND and privacy settings, and phone service subscriptions plus all phone buttons
lines, speed dials, and others. One or more device profiles are applied to an end user at the end-
user configuration page.
The default device profile stores default device configuration parameters that Cisco Extension
Mobility will apply when the actual phone model on which the user logs in does nol match the
phone model that is configured in the device profile for that user. Thc default device profile
exists once per phone model type and per protocol: Session Initiation Protocol (SIP) and
Skinny Client Control Protocol (SCCP). All the parameters that cannot be applied from the
di i- profile of the user are taken from thc default device profile.
5-40 Troubleshooting Cisco Unified Communicalions (TVOICE) v8 0 2010 Cisco Systems. Inc
For example, a user is associated with a device profile for a Cisco Unified IP Phone 7940
SCCP. If this user logs into a Cisco Unified IP Phone 7961 SIP, some feaUircsconfiguration
parameters exist on the target phone but are not configurable at thc Cisco Unified IP Phone
7940 device profile. In this case, the configuration parameters that are unavailable on the
device profile of the user are taken from the default device profile of the Cisco Unified IP
Phone 7940 SCCP.
If a device profile includes more parameters than the target phone supports, the additional
settings are ignored when reconfiguring the target phone with the user-specific settings.
Cisco Extension Mobility is implemented as a phone service. Hardware phones and device
profiles must subscribe to thc Cisco Extension Mobility service.
i 2010 Cisco Systems. Inc. Troubleshooting Cisco UnifiedCommunications Manager Features and Application Issues 5-41
Relationship of Cisco Extension Mobility Configuration
Elements
The figure shows how the Cisco Extension Mobility configuration elements relate to one
another.
5-42
Jh itiorwhip tit Tuco Es-vinîoi.
lofoilitv Confiquration Ei
End User
Userl
End User
User2
Device
Profile A
Device
Profile B
Cisco Extension
Mobility Phone
Service
Device
Profile C
7940 Phones 7961 Phones
7940 Phone
SCCP Default
Device Profile
7940 Phone
SIP Default
Device Profile
7961 Phone
SCCP Default
Device Profile
7961 Phone
SIP Default
Device Profile
An end user is associated with one or more device profiles.
For each possible IP phone model and protocol -SCCP and SIP a default device profile can
be configured.
Because Cisco Extension Mobility is implemented as an IP phone service, all phones that
should suppon Cisco Extension Mobility must subscribe to the Cisco Extension Mobility phoiu
service to allow a user to log into the phone.
In addition, each device profile must be subscribed to the Cisco Extension Mobility phone
service to allow a user to log out of a phone.
How Does Cisco Extension Mobility Manage Phone Model
Differences?
When using different IPphone models ina CiscoUnified Communications Manager cluster in
whichCiscoExtension Mobility is enabled,an enduser can log intoan IPphonethat is of a
different model thanthe one that is configured inthe deviceprofileof thc user.
How Does Cisco Extension Mob
Handie Phone Model Differences
Auser can log into a different phone model than
configured at the user device profile:
Default device profiles can be configured.
When a phone model mismatch is identified, Cisco Extension
Mobility works as follows:
-- DeviceProfile: Copyall device-independent parameters from
the device profile of the user.
Default Device Profile: Apply device-dependent parameters,
such as phone button and softkey template, from the default
device profile.
- Device Profile: Copy device-dependent parameters that can
be applied, and apply phone service subscriptions from the
device profile of the user.
- Cisco Extension Mobility Feature Safe: Phones can use any
phone button template that has the same number of line buttons
that the phone model supports.
When a user logs into a phone that supports more features than the model that is associated
withthe user, thc defaultdeviceprofileis used to applyparameters that the target phone
supports but thatare not included in thedevice profile of theuser. Thedefault device profile
includes phone configuration parameters such asphone button templates, softkey templates,
phoneservices, and other phoneconfiguration settings but does not includebutton
configuration and line buttons.
Following is a review of theprocess of how thephone is configured when using Cisco
Extension Mobility with difTerent phone models.
After successful authentication, if the phonemodel of the deviceprofiledoes not matchthe
phone model of the used phone, the following happens:
Device-dependent parameters, suchas phonebuttonand softkeytemplates from the default
deviceprofile, are appliedto the phone.
Then, the system copiesall device-independent configuration settings, suchas user hold
audiosource, user locale, speeddials, and line configuration -except for the parameters
thatarespecified under thelinesetting for thisdevice from thedevice profile to thelogin
device.
Next, the applicable device-dependent parameters of the deviceprofileof the user are
applied. These parameters include buttons such as line and feature buttons that are based on
thephone button template that hasbeen applied from thedefault device profile.
If supported on thelogin device, phone service subscriptions from thedevice profile of the
user are applied to thc phone.
2010Cisco Systems, Inc Troubleshooting CiscoUnified Communications Manager FeaturesandApplication Issues 5-43
If the device profile of the user does not have configured phone services, then the system
uses the phone services that are configured in thc default device profile of the login device.
Cisco Extension Mobility equivalency eliminates the phone-model dependency of phone button
templates.
Cisco Extension Mobility equivalency includes the Feature Safe on Phone Button Template
support feature. This feature was introduced in Cisco Unified Communications Manager
Version 7.x.
Phones can use any phone button template that has the same number of line buttons that the
phone model supports.
Cisco Unified Communications Manager v7.0 enhanced the existing Cisco Extension Mobility
equivalency mechanism. The equivalency enhancement works across these Cisco Unified IP
phone types:
IP Phone 794x models are equivalent and can share a Cisco Extension Mobility profile.
IP Phone 796x models are equivalent and can share a Cisco Extension Mobility profile.
IP Phone 797\ models are equivalent and can share a Cisco Extension Mobility profile.
The enhancement works for all equivalent phone models, requires no administration tasks to
activate, and is protocol independent.
Cisco Extension Mobility General Issues
This section describes how to troubleshoot and fix the most common Cisco Extension Mobility
problems.
Cisco Extension Mobility General
Issues
Users might experience these general issues while
using Cisco Extension Mobility:
Various login and logout procedure problems
Phone button problems
Phone service problems
Call routing problems
CoS problems
Additional problems specific to Cisco Extension Mobility
Cross Clusters
Depending on how Cisco Extension Mobility is set up in a Cisco Unified Communications
Manager cluster, users might experience the following general problems when using the
service:
Problems (typically authentication issues) related to login and logout procedures
Problems that are related to how functions such as lines and speed dials are assigned to
phone buttons
Problems accessing subscribed-to phone services when the user logs in
Call-routing problems when the user logs in
Previous class of service (CoS) unavailable when the user logs in
Various specific issues of Cisco Extension Mobility Cross Cluster when a roaming user
logs in across a cluster boundary
2010 Cisco Systems, Inc. Troubleshooting Cisco UnifiedCommunications Manager Features and Application Issues 5-45
Troubleshooting Cisco Extension Mobility Error
Messages and Login and Logout Issues
"I his topic explains the Cisco Extension Mobility error messages that can be experienced,
describes their most probable causes, and describes how to troubleshoot Cisco Extension
Mobility login and logout problcms.
[201f-Autri entication error
[22]-Devlogwi disabled
[205|-User Profile Absent
(208]-EM Service Conn, error
[25]-User togged in elsewhere
Host not found
Http Error [503]
Check that the correct user ID and PIN are
entered.
Ensure thai Cisco Enable Extension Mobility is
checked on ihe phone.
Ensure ttiai you have associated a device
profile io the user.
Verify that the Cisco Extension Mobility service
is running.
The user logged in to another phone Ensure
that Multiple Login Behavior is enabled.
Ensure thai the Cisco Tomcat service is
runnhg.
Ensure [hat the Cisco IP Phone Services
service is running.
Use thc information in this table to troubleshoot the error codes and error messages that display
on the phone when Cisco Extension Mobility is used. The following list explains error codes
with recommended action to resolve the issue:
201-Authentication error: The user should confirm that the correct user ID and PIN" arc
entered and should check with the system administrator to confirm that the same user ID
and PIN are configured correctly in Cisco Unified C'ommunications Manager.
22-Dev.logon disabled: Make sure that yoti have cheeked the Cisco Enable Extension
Mobility check box in the Phone Configuration window.
205-User Profile Absent: Make sure that you have associated a device profile to the user.
208-EM Service Connectivity Error: Verity that the Cisco Extension Mobility service is
ninning. Choose Cisco Unified Serviceability in Tools > Control CenterFeature
Services.
25-User lugged in elsewhere: Determine whether the user is logged into another phone. If
multiple logins need to he allowed, ensure that the Multiple Login Behavior service
parameter is set to Multiple Logins Allowed.
Host not found: Checkthat the CiscoTomcatservice is running. ChooseCisco Unified
Serviceability in Tools > Control CenterNetwork Services.
Http Error 503: If you get this error when the Services button is pressed, check that the
Cisco Communications Manager Cisco IP Phone Services service is running. Choose Cisco
Unified Serviceability in Tools > Control CenterFeature Services. If you get this
error when you select the Cisco Extension Mobility service, check that thc Cisco Extension
Mobility Applicationservice is running. Choose Cisco Unified Serviceability in Tools >
Control CenterNetwork Services.
2010CiscoSystems, Inc Troubleshooting CiscoUnified Communications ManagerFeatures and Application Issues 5-47
lobility Error (Vlessays
[202}-Blank userid or pin
(26[- Busy, please try again
[63-Database Error
[207}-Device Name Empty
Enter a vaid user ID and PIN
Verify thai Ihe number of concurrent login and
iogoul requests is not greater than permitted.
Determine whether a large number of database
requests exists.
Verify that the correct URL is configured for
Cisco Extension Mobility.
202-Blank user id or pin: Enter a valid user ID and PIN,
26-Busv, please try again: Check whether the number of concurrent login and logout
requests is greater than the Maximum Concurrent Requests service parameter. If so. then
lower the number of concurrent requests. To verify the number of concurrent login and
logout requests, use Cisco Unitied Communications Manager Real-lime Monitoring Tool
(RTMT). to \iew the Requests In Progress counter in the Cisco Extension Mobility object.
6-Database Error: Check whether many requests exist. If so, the Requests In Progress
counter in the Cisco Extension Mobility object counter specifies a high value. If the
requests are rejected because of many concurrent requests, thc Requests Throttled counter
also specifies a high value. Coiled detailed database logs.
207-Device Name Empty: Check that the correct URL is configured for Cisco Extension
Mobiliu.
Troubleshooting Cisco Extension Mobility Login Problems
This section describes thc most-common problems that exist when logging into Cisco
Extension Mobility and how to solve them.
Troubleshooting Cisco bxtension
in
Problem:
* Service is
unavailable.
Possible cause:
Phone device or logout
device profile does not
subscribe to Cisco
Extension Mobility.
1want to log in
yavioes
?
'
User cannot log in.
Services
Extension
Mobility
Phone restarts
instead of
resetting.
Restarting
Wrong username or PIN
or missing permissions.
Username or PIN is
invalid; logon is
disabled.
User locale of user or
profile is not the same
as the locale of device.
When a user attempts to log into Cisco Extension Mobility and presses the Services button, the
Cisco Extension Mobility service is unavailable and does not appear on the phone display.
This issue occurs when either a phone-device or a logout-device profile that the phone is
using depending on the current state of the phone -does not subscribe to the Cisco Extension
Mobility service. In Cisco Unified Communications Manager Administration, choose Device >
Device Settings > Device Profile and verify that the device profile that is used as a logout
profile for the phone subscribes to the Cisco Extension Mobility service.
When the user selects Cisco Extension Mobility service and supplies the requested username
and PIN. the user gets the authentication error. The most common reason for a username and
PIN not working is that the user mistyped them when entering them or the Cisco Unified
Communications Manager Administrator mistyped them when configuring them. Verify that
they match.
[f the username and PIN are correct, then other causes that relate to permissions might exist.
Make sure that the phone is enabled for Cisco Extension Mobility and that thc Enable
Extension Mobility check box on the Phone Configuration page is checked.
Make sure that the end user who is attempting to log in has been associated with a Device
Profile for Cisco Extension Mobility.
Also determine if the user is logged into another phone. If multiple logins need to be allowed,
ensure that the Multiple Login Behavior service parameter is set to Multiple Logins Allowed.
After performing a successful login, the user finds that the phone restarts instead of resetting.
J2010 Cisco Systems, Inc Troubleshooting Cisco Unified Communications Manager Features and Application Issues
If the user locale that is associated with the login user or profile is not the same as the locale for
thcdevice, thenafter a successful login, thc phonewill perform a restart that is followed bya
reset. This issue occurs because the phone configuration file is being rebuilt.
Make sure that the user locales match across the phone, the device profile, and the end-user
configurationin Cisco Unified Communications Manager Administration.
5-50 Troubleshooting Cisco Unified Communications (TVOICE) v6 0 J2010 Cisco Systems, Inc
q^^
Troubleshooting Cisco Extension Mobility Logout Problems
Some issues might occur during the logout procedure.
Troubleshooting Cisco Extension
Mobility Logout Problems
Problem:
Service is
unavailable.
I want to log out.
Phone restarts
instead of resetting.
Services
Restarting
Possible cause:
User device profile not
subscribed to Cisco
Extension Mobility.
User locale of user or profile
is not the same as the locale
of the device.
When a user attempts to log out from Cisco Extension Mobility and presses the Services
button, the Cisco Extension Mobility service is unavailable and the user is not offered the
option to log out.
This issue occurs when the device profile to which thc user is associated does not subscribe to
the Cisco Extension Mobility service. In Cisco Unified Communications Manager
Administration, choose Device >Device Settings >Device Profile, verify the user device
profile, and subscribe it to the Cisco Extension Mobility service.
After performing a successful logout, the user finds that the phonerestartsinsteadof resetting.
Thc most probable cause is the same as thc problem that occurs when logging in: a mismatch
between user locales.
)2010Cisco Systems. Inc Troubleshooting Cisco UnifiedCommunications Manager Features and Application Issues 5-51
Troubleshooting Cisco Extension Mobility Phone Button
Problems
When Cisco Extension Mobility is enabled on different Cisco Unified IP Phone models, end
users might log into an IP phone that is of a different model than the one that is configured in
their device profile.
5-52
Tim,M-'Miu.mmi Cisi .> ! *tef>^
Problem;
After login button functions are unavailable
Possible causes, recommended actions:
> Check for rjhone model mismatch in multimodel environment
- Combine phone mod el-dependenI and model-independeni parameters and Feature
Safe rrodel equivalency
1 Create user device profile per expected phone model, supporting al least he/ us.er-
se.ected features
- Use Cisco Unified Commun'Cations Manager trace to identify button function
assignments
Main Site
Cisco Unified
IPPhcr.e 7&71
User Device Profile
for Cisco Unified IP
Phone 7971
y Where are my lines
and speed dials''
Remote Site
Cisco Urafed
IP Phone 7940
The problem is a phone-model mismatch in a multimodel environment. If thc user device
profile uses more buttons than the target phone model can physically support, then Cisco
Unified Communicalions Manager must choose which button functions and other features the
target phone model will offer to the user.
In addition to thc user device profile, the default deviee profile exists and includes phone
configuration parameters such as phone button templates, softkey templates, phone services,
and other phone configuration settings. However, thc default device profile docs not include
button configuration and line buttons. The phone modcl-dcpcndcnt parameters and model-
independent parameters are combined when a mismatch is identified.
Cisco Unified Communications Manager v7.0 further enhances the existing Cisco Extension
Mobility equivalency mechanism through Feature Safe model equivalency: Phones can use any
phone button template that has the same number of line buttons as the target phone model
supports. All individual phone models within a phone model family are equivalent and can
share a Cisco Extension Mobility profile.
As an alternative, you can create a user device profile per expected phone model to support at
least the primary user-selected features on phone models for which all requested features will
not be possible.
Thc Cisco Unified Communications Manager trace for the Cisco Extension Mobility service
can be used to identify how button functions are assigned to a target phone.
Troubleshooling Cisco Unified Communicalions (TVOICE) v8.0 12010 Cisco Systems, Inc
Troubleshooting Cisco Extension Mobility Call
Privilege Problems
This topic describes howto troubleshoot typical Cisco Extension Mobility call privilege
problems that lead to incorrect calling permissions.
Mobility Call Privilege Issues
Problem of broken CoS policies when roaming:
* Earlier denied destinations are reachable after login.
Eariier permitted destinations are not reachable after login.
Consider CSS misconfiguration, follow potentials of
selected CSS model.
+ 1 900 555-1000 +49 6 9455 62 00
San Jose i\ /* . Frankfurt
A user who logs into Cisco Extension Mobility might experience problems that relate to CoS,
or the company policies can be broken.
The figure shows that a user who logs in at the remote site, can reach the public switched
telephone network (PSTN) numbers that were denied earlier, such as expensive 900 services.
Or. the user can no longer reach thc destinations that were accessible at thc main site.
These problems are caused by misconfigurationor incorrect design of the CSS. The two CSS
models differ in their potential. Cisco Extension Mobility never updates the phone deviceCSS,
but updates the line CSS after a successful user login.
When the traditional CSS model is used with Cisco Extension Mobility, the device CSS keeps
its original setting, which might not address all CoS permission levels across all sites. Instead,
thc CSS relates to the users in the local site only.
The line or device CSS model is recommended for CoS implementationwith Cisco Extension
Mobility. In this model, the line CSS should include the CoS that relates to thc particular user
and that is assigned to the user device profile.
>2010Cisco Systems, Inc. Troubleshooting Cisco UnifiedCommunications Manager Features and Application Issues
Cisco Extension Mobility and CSSs
This section reviews how Cisco Extension Mobility interacts with CSSs,
<..*t~A>St
u 11 h mm
Consider the following with Cisco Extension Mobility:
- Device CSS is never modified by Cisco Extension Mobility.
Line CSS is modified by Cisco Extension Mobility.
Operation when using local route groups or the line or device
CSS approach:
Line CSS of user device profile is applied and CoS settings
of the user are enforced.
Device CSS is not modified, which allows local gateway
selection, depending on the device used (by different device
CSS or local route group).
When using the traditional CSS approach (only one CSS at phone),
the same CSS (either device or line) is used all the time, causing
problems in multisite environments with different CoS for users.
AAR CSS is configurable only at the device and is never updated by
Cisco Extension Mobility, which allows the local gateway to b used
for AAR calls.
Cisco Extension Mobility docs not modify the device CSS or the automated alternate routing
(AAR) CSS; both are configured at the device level. It docs replace the line CSS that is
configured at thc phone with thc line CSS that is configured in the device profile of thc logged
in user.
This means that in an implementation that uses the line or device approach, the following
applies:
> The line CSS of the login deviee is updated with the line CSS of the user. This is done to
enforce the same CoS settings for the user, independent of the physical deviee that the user
uses.
Because the device CSS of the login device is not updated, the same gateways are used for
external route patterns, which were initially configured at the phone before the user logged
in. Because the phone did not physically move, the same local gateways are used for PSTN
calls even uhen a different user is currently logged into the device.
Note that uhen using the Local Route Group feature, the situation is similar. The line CSS is
updated and the phone device CSS is maintained. However, in this ease, it is irrelevant if the
device CSS is modified because all phones arc configured with the same device CSS when you
use local route gumps. The device CSS of all phones provides access to the same system
partition that includes all PSTN route patterns.
If you use the traditional approach to implement partitions and CSS. the following applies:
If only device CSSs are used, thc CSS is not updated and no user-specific privileges can be
applied. The user will have the privileges configured at the device that is used for login.
If only line CSSs are used, the line CSS that is configured in the device profile of the user
replaces the line CSS of the login device. In a multisite environment, this replacement
might cause problems in terms of gateway selection as a wrong gateway can be selected
(unless local route group is used in this scenario).
m
CSSs Example: Device CSS Only
This section inspects the problemthat arises when you use Cisco Extension Mobility in an
environment where a device-only CSS is used.
CSSs Example: Device C!
HQ Phone (: i,c.;>.d-0-.nSÛ.)
HQ Phone (After Login)
Line
No Line
CSS
Device Level
Device CSS:
Line
No Line
CSS
Device Level
Device CSS:
Default CoS and
BR User Profile -^
* Line-level CSS is not used.
No updates are made.
Line
No Line
CSS
Device Level
No CSS Configurable
No user-specific CoS possible
In this case, the device CSS of the phone provides CoS and gateway selection. Assuming that
no line CSS is configured in thc user profile, it is evident that CoS that is specific to thc user
who has been logging in cannot be applied to the phone.
2010 Cisco Systems, Inc Troubleshooting Cisco Unified Communications Manager Features and Application Issues 5-55
CSSs Example: Line CSS Only
In this example, you see the problem that arises when using Cisco Extension Mobility in an
environment where onlv line CSSs are used.
HQ Phone
Line
Line CSS.
Device Level
No CSS Configured
BR User Profile
Line Device Level
Line CSS No CSS Configurable
User CoS
and
BR_gw
Line Device Level
Line CSS: No CSS Configured
User CoS
and
BRgw
Device level CSS is not used.
Line CSS determines CoS and
gateway to be used.
After login, CoS is updated as
desired.
Gateway selection is
suboptimal when logging in at
multiple sites (HQ phone uses
BR gateway).
If a user logs into a phone at the user site, there arc no problcms. The line CSS of the phone that
provides both functionalities the selection of the gateway and the application of calling
privileges during the logged-out state is simply updated with thc line CSS of the user profile.
The phone to log into is located at the headquarters site, and in the logged-out state, only line
CSSs are used at the phone. The default phone CSS implements default calling privileges and
the selection of a gateway IIQ gw.
The device profile for the user from thc branch site, that is going to roam at headquarters,
contains Line CSS that implements the user-specific calling privileges. As the user did not
roam in other than the branch site before, the deviee profile Line CSS also implements gateway
selection that chooses BR gw.
If this user logs into the phone at headquarters, during the login process thc Line CSS of the
user device profile overwrites Line CSS of the headquarters phone. Because the gateway is
selected based on thc profile of the user, the same gateway, in this case the branch gateway
BRgw, will be used all the time. Please note that this scenario docs not have thc local route
group implemented. If it would, it would provide the correct local gateway also in case of using
Line CSS only as shown on the next figure.
Using a line-only CSS implementation model is, therefore, only practical in a single-site
environment or when users do not roam to other sites.
Troubleshooting Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems, Inc.
tf^^^
^
CSSs Example: Line CSS with Local Route Group
In this example, line CSS is combined with the local route group to decouple calling privileges
from the gateway selection.
CSSs Example:
jroup
HQ Phone i Stafc)
Line
Line CSS:
Device Level
No CSS Configured
Device Pool HQ
\
Device Pool HQ
LRG: HQ_gw
Device Pool ' >-
LRG' . . S
BR Phone (L:>c:jf Sidte)
Line
Line CSS:
Device Level |
No CSS Configured
Device Pool
Line
Line CSS:
User CoS
Device Level
No CSS Configured
Device Pool HQ,
/
BR User Profile
Line
Line CSS:
User CoS
Device Level
No CSS Configurable
\
BR Phone (After Login)
Line
Line CSS:
User CoS
Device Level
No CSS Configured
Device Pool :>:.;
The focal route group is always pointing to the gateway that is specific to a site. Phones at the
headquarters site are associatedwith the device pool HQ, which points to the route group that
has the local gateway HQ_gw set as its local route group.
Phones at the branch site are associated with the device pool BR, which points to the route
group that has the local gateway BRgw set as its local route group.
When a number is dialed, match against route pattern is found and the route list that is
referenced has the standard local route group configured. A local route group from a calling
phone device pool is always used for the call.
The Cisco Extension Mobility login process does not affect the device pool parameter, and it
remains the same whether it is in the logged-out or logged-in state.
If a branch user logs into Cisco Extension Mobility at thc home site BR, the user-specific
calling privileges are inherited from the user device profile as a line CSS. The BR phone uses
the local route group that is pointing to the local gateway BR_gw after the login.
When a branch user logs into Cisco Extension Mobility at the headquarters site, the user-
specific calling privileges are taken from the user device profile as a line CSS, but the HQ
phone continues to use the local gateway HQgw, which is taken fromthe HQ phone device
pool local route group.
Using a line-only CSS implementation model that is combined with the local route group
solves the problemof using a line-only CSS and decouples the gateway selection fromcalling
privileges that are transferred fromthe user device profile line CSS. This model is very
practical in multisite environments when users with different calling privileges roam to other
sites and call routing needs to be kept optimal at all times.
CSSs Example: Line or Device CSS
The figure illustrates the operation in the line or deviee CSS implementation mode).
p/i
HQ Phone { . . \ )
Line
Line CSS
Device Level
Device CSS
User Profile
Line
Line CSS
User CoS
Device Level
No CSS Configurable
BR Phone ( .;, - :....- )
\
Line
Line CSS'
Device Level
Device CSS'
Line
Line CSS.
User CoS
Device Level
Device CSS.
BR Phone (After Login)
Device-level CSS remains
the same
HQ phone still uses HQ
gateway.
Line CSS is updated.
User-independent default
CoS of phone(used
during logged-out state)
is updated with individual
user CoS. Line
Line CSS'
User CoS
Device Level
Device CSS:
There is a headquarters phone that is configured with a line CSS and a device CSS. The line
CSS is used to implement CoS, while the device CSS is used to select a site-specific gateway.
The user profile cannot be configured with a device-level CSS; it only supports a CSS at the
line level.
When a user logs into the phone, the device-level CSS of the phone remains the same.
Therefore, the gateway that is used for PSTN calls does not change.
The line CSS is updated and, therefore, the CoS that was applied to the phone in the logged-out
state is updated with thc user-specific CoS settings that are configured in the user profile.
In a multisite deployment that uses the line or device CSS implementation model. Cisco
Extension Mobility works in an optimal way: the phone still uses its local gateway but thc CoS
of the user is applied.
This model can also be combined with the local route group that would replace the gateway
selection of the device CSS. Combining line or device CSS with the local route group would
lead to a simpler configuration than when user calling privileges that are implemented by using
line CSS and optimal call routing by using thc local route group, as in thc previous figure.
5-58 Troubleshooting Cisco Unified Communications (TVOICE) v8.0 ) 2010 Cisco Systems, Inc.
Troubleshooting Cisco Extension Mobility Call
Routing Problems
This topicdescribes howto troubleshoot typical CiscoExtension Mobility call-routing
problems that lead to calls blocking, inefficient call-routing, and incorrect caller ID.
Mobility Call-Routing Issues
Possible causes:
CSS problem, route group not set.
local route group is misconfigured
Call-routing problems:
No PSTN calls is possible after login.
Wrong gateway is used for PSTN.
angulation
Local route group missing or
misconfigured, wrong route group is
set in device pool or route list.
Phone CCS does not include
globalization or localization route
partitions
4^
-49 6 9455 62 00
San Jose ^_i__-. Frankfurt
^-r/"^
^<s- -> ^
Misconfiguration often results in call-routing problems after a user logs into the Cisco
Extension Mobility service.
After thc successful login, the user attempts to call a PSTN number that the user could reach
earlier (when not logged into Cisco Extension Mobility at that phone), but the call is blocked
and the user hears the reorder tone.
If lack of calling privileges is not an issue, the most common reason for this problem is that the
CSS that the logged-in phone is using selects no gateway or the wrong gateway. As explained
earlier, the Cisco Extension Mobility user device profile does not update the phone device CSS,
but it updates line CSS that could have no gateway selection implemented.
If the phone failed even before logging in, thc proper route group might be missing in the route
list, if the Local Route Group feature is implemented, the local route group might be missing in
the route list. Or, an incorrect local route group might be selected in the device pool.
When a user moves between the main and remote sites, the calls that are placed from the
remote site might still be using the main site gateway for PSTN calling. As the figure shows,
the user has temporarily moved to Frankfurt, but calls to local German PSTN phones are still
using the San Jose PSTN gateway.
In this case, the route list probably does not include the local route group, or the device pool
refers to the wrong route group, or the route pattern that the user is dialing does not have a
route list that contains the local route group.
2010 Cisco Syslems, Inc. Troubleshooting Cisco Unified Communications Manager Features and Application Issues 5-59
Summary
Ihis topic summarizes thc key points that were discussed in this lesson.
inmary
Cisco Extension Mobility tssues include login and logout
problems and problems with call routing and call privileges.
Cisco Unified Communications Manager provides error
message diagnostics for some Cisco Extension Mobility
issues.
When troubleshooting Cisco Extension Mobility call privilege
problems, rememberthat device CSS is never modified by
Cisco Extension Mobility; only line CSS is modified.
When troubleshooting Cisco Extension Mobility call routing
problems, consider phone CSS or local route group
misconfiguration.
In this lesson, you have learned to explain thc common issues that are related to Cisco
Kxtension Mobility and identify the most likely causes of these issues.
References
Cisco Systems. Inc. Cisco Unified Communications Manager Features andServices Guide
Release S.Otll. April 2009 and updated April 2010.
http: wu A.i'ivi.'u.i.-om en l;S docs'wice ip comm euan adiiiui'iS 0 ] cemfeitt
fiL'd-Hul-cm.himl
Lesson 3
Mobility Issues
Overview
The growing use of mobile devices allows users who are on the move-whether on aretail
floor, at an airport, or at aWi-Fi hotspot in alocal coffee shop-to enjoy the efficiencies and
speed of Cisco Unified Communications. However, as more people own multiple devices -
ranging from office phones to home office phones, laptop computers to mobile phonesthey
must spend more time managing communications across different phone numbers and voice
mailboxes. This necessity can limit the ability ofusers to accomplish work efficiently. Cisco
Unified Mobility allows users to be reached at asingle number, regardless of which device they
use.
This lesson will describe some typical Cisco Unified Mobility problems and how to
troubleshoot them.
Objectives
Cisco Unified Mobility and identify the most likely causes ofthese issues. This ability includes
being abletomeet theseobjectives:
Describe Mobile Connect and Mobile Voice Access operation and configuration, describe
the ToD access and service parameters that are related toCisco Unified Mobility, the
Enterprise Feature Access and dusting features, and describe general issues that users
experience when using Mobile Connect and Mobile Voice Access
Describe bow torecognize and troubleshoot most common Mobile Connect issues
Describe how to identify and troubleshoot Mobile Voice Access incoming and outgoing
call problems
Describe how totroubleshoot typical Enterprise Feature Access and dusting feature
problems when auser cannot park, transfer, or resume the call on aremote phone
Cisco Unified Mobility and Its General Issues
This topic makes abrief review of Mobile Connect and Mobile Voice Access operation and
configuration, ,t briefly reviews the time-of-day (ToD) access and service parameters that are
related to Cisco Umtied Mobility and reviews the Enterprise Feature Access and dusting
features. 1he topsc concludes with outlining general issues that arc experienced when usinu.
Mobile Connect and Mobile Voice Access.
t.'icjJjilc Cof,=
Cisco Unified
Communications Manage
Mobile Connect
tniii
Outside Caller
408 555-1001
Outside caller calls office phone 2001 (dials 1511 555-2001).
Mobile Connect rings office phone and remote phone.
Call is picked upby at remote phone, caller ID ofoutside caller is
preserved at remote phone.
Cisco Unified Mobility consists oftwo main components: Mobile Connect and Mobile Voice
Access:
Mobile Connect allows an incoming call to auser enterprise phone number to be offered to
thc user office phone and to as many as 10 configurable remote destinations. Such remote
destinations typically are mobile or cellular telephones or home office phones.
Mobile Voice Access is built in addition to the Mobile Connect application and provides
similar features for outgoing calls. When Mobile Voice Access isenabled, auser who is
outside the enterprise can make calls as ifthe user was directly connected to Cisco Unified
Communications Manager. This functionality iscommonly referred to as Direct Inward
System Access (DJSA) intraditional telephony environments.
Both features allow active calls to be switched between the IP phone and the remote phone For
example, ifauser initiates acall from amobile phone while on the way to thc office, the user
canswitch the call totheoffice phone upon arrival.
Mobile Connect enables users toreceive business callsby using a single phone number,
regardless ofthe device that is used for acall. Mobile Connect allows users to answer incoming
calls on the office phone oratthe remote destination and to pick up in-progrcss calls on the
office phone or remote destination, without losing the connection. When the call is offered to
both the desktop and remote destination phones, the user can answer atany ofthose phones.
After answering the call on one ofthe remote destination phones or the IP desk phone, the user
can hand offthe call tothe office phone. Active calls on the office phone can be handed offtoa
remote phone.
The figure illustrates the call flow when using Mobile Connect. The figure shows an IP phone
withextension 2001 anda mobile phone thatbelongs to thcuserof thcIPphone.
Apublic switched telephone network (PSTN) user calls thc office number ofthe IP phone user.
Because Mobile Connect is enabled, both thedesktop phone 2001 andtheconfigured remote
destinationmobile phone 408 555-1001 ring simultaneously. The call ispresented to the
remote phone with the original caller ID^t79555 1555. As soon as the call is accepted on one
ofthe phones, the other phone stops ringing. During the call, the user can switch between the
office phone and thc mobile phone without losing the connection.
2010 Cisco Systems. Inc. Troubleshooting Cisco Unified Communicalions Manager Features and Application Issues 5-63
Mobile ConnectInternal Calls Placed from Remote Phone
This section rc\ lews thc call fiow of acall that is placed from aremote phone to an internal
destination.
5-64
.UbiU Cjiuttc;- inwrn
rom Remote Phone
Cisco Unified
Communications Manage
Mobile Connect
Gateway
X^ 2^-^s. 511555"
.2XXX
Remote phone calls internal phone 2002 (dials 1511 555-2002).
Mobile ConnectreplacescallerID (408555-1001) of remote
phone with directory number ofassociated office phone (2001).
Mobile Connect influences the calling-number presentation. When a call is received from a
recognized remote destination, thc corresponding internal directory number, rather than the
E.164 number of thc remote device, is used for the calling number.
In this example, extension 2001 has a Mobile Connect remote destination 40H 555 1001 --the
cell phone ofthe user of2001. The user places acall from the mobile phone to an enterprise
PSTN number ofa colleague by dialing I 511 555-2002. The called colleague sees the call as
coming from the internal directory number 2001 rather than thc external mobile phone number.
The same applies tocalls that areplaced toother internal destinations, such as voice mail. If the
user ofextension 2001 places acall from the cell phone to Cisco Unity, Cisco Unity will see
the source ofthe call as directory number 2001 rather than the PSTN number ofthe cell phone.
Cisco Unity can then identify thc user by this directory number and provide access to the
appropriate mailbox instead of playing a generic welcome greeting.
Torecognize Mobile Connect remote destinations, theMobile Connect remote destination
number must match the Automatic Number Identification (ANI) ofthe incoming call. Mobile
Connect remote destinations, such as 9 1408 555-1001, typically include an access code.
Therefore, the access code "9" and the long-distance code "1" must be prefixed to the incoming
ANI 408 555-1001 to recognize the sourceas a Mobile Connect remotedestination.
Alternatively, the Cisco CallManager service Matching Caller ID with Remote Destination
parameter can beset toPartial Match, and theNumber of Digits for Caller ID Partial Match can
be specified. This number specifies how many digits ofthe incoming ANI, starting with the
least significant digit, must match a configured remote destination number.
If the source of the call is not recognized asa Mobile Connect remote destination, then the
PSTN number ofthe remote destination isused for the catling number and is not changed to the
internal directorv number.
*" Mobile Voice Access
This section reviews Mobile Voice Access.
Mobile Voice Access
Cisco Unified
Outside Destination
Remote Phosif
2"0'=
Remote phonedials Mobile Voice Access number (1 511 555-2999).
Caller is authenticated at Cisco Unified Communications Manager
and requests call to outsidedestination9 1 479 555-1555.
Caller is connected to Mobile Voice Access media resource from
whichthe outgoing call is placed on behalf of the officephone
(2001).
By using Mobile Voice Access, users can place calls tothe outside from aremote destination,
asif they were dialing from adesktop phone. Inthe figure, the user of the IPphone with
directory number 2001 uses the cell phone 408 555-1001 todial the PSTN number of the
headquarters extension 2999. The gateway isconfigured tostart aninteractive voice response
(IVR) call application for calls that are placed tothis number. The Voice Extensible Markup
Language (VoiceXML, also known as VXML)-based call application offers a prompt and asks
for the remote destination number and the PIN of the user. After logging in, the user can
activate or deactivate Mobile Voice Access as well as initiate a call fromthe enterprise
network. The call issetupwith the E.164 PSTN number ofdirectory number 2001 instead of
with408 555-1001. Thisbehaviorallows thecalledpartyto identitythe caller bya single office
number. That the call was actually placed from a mobile phone instead of theoffice IPphone is
irrelevant; the call appears to comefromthe officephone.
When the user has used Mobile Voice Access to initiate a call from a remote destination, the
usercanswitch thecall totheoffice phone without losing theconnection andcanswitch back
again as needed.
Troubleshooting CiscoUnified Communications Manager FeaturesandApplication issues 5-65
Cisco Unified Mobility Configuration Elements
Cisco Unified Mobility requires Cisco Unified Communications Manager Version 6.0 orlater.
To start, at least one Cisco Unified Communications Manager needs the Mobile Voice Access
service, uhich interacts with thc call application that runs on aCisco IOS gateway. Mobile
Voice Access requires an H.323 or Session Initiation Protocol (SIP) gateway toprovide a
VXML call application toremote callers who dial a certain numbcr. Media Gateway Control
Protocol (MGCP) isnot supported because it does not support call applications.
End user
Phone
Remote desisiaSon profile
Remote destine a on
Mobile Voice Access media resou
.'>nl
The end user is referenced by the office phone and remote
deslinalioii profile Mobile Conned or Mobile Voice Access must be
enabled. Amammiin number of remota destinations can be
configured
Theoffice phone must beconfigured with an owner (i.e., enduser)
Awituaf phone device. Ashared Ime is configured peroffice phone
nunber End user, (device) CSSs, and MOHaudio sources are
specified. One or more remole destinations are added
Associated with shared lines of remote destination profile
Configured wiih deslinalion number Across lists can be applied
Mobile Phone and Mobile Conned functionsare selectively
enabled
Fillers used to permit or deny incoming calls placed io the olice
phone to nng a remote destination Permitted or denied caller Ds
are specified
Media resource used to interact with the VXML call application
running on a Cisco IOS router Required only for MobileVoice
Access
Thetable liststhe CiscoUnified Mobility configuration elementsand reviews their function:
End user: Each enduser must havea configured PIN, which is used for authentication
when using Mobile Voice Access. Three important Cisco Unified Mobility-related settings
can be configured at the end user:
Enable Mobility: This cheek box must be checked to allow thc user to use the
Mobile Connect feature to receive enterprise calls at oneor more remote
destinations and toplace calls from a remote phone tothe enterprise.
Enable Mobile Voice Access: This check box must be activated for the user to
place outgoing enterprise Mobile Voice Access calls from a remote phone onbehalf
of the office phone.
Remote Destination Limit: This settinglimitsthe amount of remotedestinations
that can be configured. The maximum is 10.
IP phone: The office phone of a Cisco Unified Mobility usermustrefertothcend-user
name. This requirement is met bysetting the owner at the phone configuration page tothe
userIDof theenduser. At theend-user configuration page, theendusercanbeassociated
with one or more devices, such as IP phones. Such an association allows thc end user to
configure the device from thc Cisco Unified Communications Manager user web pages but
is irrelevant toCisco Unitied Mobility. The mapping of an IPphone toanenduser must be
done bysetting the owner at thcphone configuration page.
Remote destination profile: This element is a virtual phone that is linked to the end user
and that represents all remote destinations. The element includes phone device-level
configuration settings such as calling search space (CSS) and user and network music on
hold (MOH) audio sources. For each office phone that an end user should be able to use for
Cisco Unified Mobility, a line that is shared with the office phones must be added to the
remote destination profile. In addition, the remote destination profile is configured with
remote destinations.
Remote destination: A remote destination is associated with one or more shared lines of a
remote destination profile. For each remote destination, the remote destination number
must be specified as dialed from within the enterprise. The CSS of the associated line and
the CSS of the specified remote destination profile are combined and used like a line CSS
and device CSS of an IP phone. The partitions of the line CSS are combined with the
partitions of the device CSS with higher priority being given to the partitions of the line
CSS. Use this combined CSS to look up the configured remote destination number in the
call-routing table.
Access list: Access lists can be configured to permit or deny calls to be placed to a remote
destination when the shared line is called; the filter is based on the calling number. An
access list is configured with one or more members who specify the calling number that
should be permitted or denied. Access lists are also configured with an owneran end-user
IDand are applied to remote destinations. An allowed, blocked, or no access list can be
applied. If an allowed access list is applied, then all calling numbers that are not listed in
the access list are blocked. If a blocked access list is applied, then all unlisted numbers are
allowed. If no access list is applied, then all calling numbers are allowed to ring the remote
destination.
Mobile Voice Access media resource: This media resource interacts with the VXML call
application that is running on the Cisco IOS gateway. The resource is required only for
Mobile Voice Access. The number at which the Cisco IOS router can reach the media
resource must be specified, a partition can be applied, and one or more locales must be
selected.
2010 Cisco Systems, Inc. Troubleshooting Cisco UnifiedCommunicalions Manager Features and Application Issues 5-67
Shared Line Between Phone and Remote Destination Profile
The figure illustrates how a remote destination profile shares its lines with associated Cisco IP
phones.
v*h.ji LU L ss k.
U:\r\Oixi
Lmel 2001
Partition
CSS
Etc
Call 10 sha.-ed
line rings office
phone line and
remote
dest.nat.ons
associated with
corresponding
lines of remote
destination
profile
!5tin;
Office Phone 1
MAC Address
Owner
CSS
Etc.
Line1:2002 Office Phone 2
Partition MAC Address
CSS Owner
Etc. CSS
i
Etc.
Remote
Destination1!:
9 1 408 555-
1001
Line 1 2001
Partition
CSS
Etc.
Remote
Destination
Profile
User ID
CSS
Rerouting CSS
Etc.
\
Remote
Destination2:
9 1 479 555-
1555
Line2: 2002
Partition
CSS
Elc
A remote destination profile is associated with one or more IP phones. Hach phone line that an
end user should be able to use with Cisco Unified Mobility must be added to thc remote
destination profile that is associated with the end user. In this way, the directory number is
associated with two devices: the IP phone and the remote destination profile. Such a directory
number is also called a shared line. The end user who is associated with the remote destination
profile must own the IP phones that share their line with the remote destination profile.
Remote destinations are associated with one or more shared lines that are configured at remote
destinations
As described earlier, thc settings of thc shared directory number including partition and
CSS apply to all associated devices. The remote destination profile is configured with a
standard CSS. That CSS is used for calls that a remote phone places when using Mobile Voice
Access. Thc profile is also configured with a rerouting CSS, which applies to calls that are
placed to a remote destination.
For example, when a call is placed lo directory number 2002, Line I at Office Phone 2, all
remote destinations that are associated with Line2 of the remote destination will ring. For the
call to thc remote destination number, the rerouting CSS of the remote destination profile is
used.
If the remote phone with number 9 I 479 555-1555 calls into the mobile voice application and
requests an outgoing call to be placed, then the CSS of Line 2 and the remote destination
profile CSS are combined and used for the outgoing enterprise call that the remote destination
has initiated.
*" Relationship of Cisco Unified Mobility Configuration Elements
This section discusses how Cisco Unified Mobility configuration elements interact with one
another.
Relationship of Cisco Unified
Service Activalion
Service Parameter
Media Resources
IVR Application
VXML
Call MVA
EiilerROanaPIN -
H 323 Gaieway
AL Access Lisl MC. Mobile Connect, MVA: Mobile Voice Access, RD: Remote Destination,
RDP. Remote Destination Profile
The basis for Cisco Unified Mobility is to activate the Cisco Unified Mobile Voice Access
service, if Mobile Voice Access is desired in addition to Mobile Connect functionality.
When thc Cisco Unified Mobile Voice Access service is activated, a corresponding media
resource is automatically added. The media resource must be configured with the Mobile Voice
Access number, a partition, and locales.
The configured number must be reachable from the Cisco IOS router that provides remote
phones access to a VXML IVR call application.
Aremote destination number and a PIN authenticate incomingMobile Voice Access callers.
The PIN is configured for the user who is associated with the remote destination profile that is
referenced from the corresponding remote destination number.
When you use Mobile Connect and send incoming calls that were placed to a line that is shared
by an IP phone and a remote destination profileboth of which refer to the same end-user
IDyou can apply access control lists (ACLs) to remote destinations to control which callers
can ring the remote destination. The ACL must refer to the end user who is configured at the
remote destination profile that has the remote destination assigned.
For an active call to be handed over from an IP phone to a remote destination, thc IP phone
must have the mobility softkey that is configured for the Connected call state. If the mobility
softkey is also added to the On-Hook call state, then the softkey can be used to check the status
of Cisco Unified Mobility Mobile Connect on or off.
In summary, the central end-user element is associated with IP phonesconfigured as owner
access lists, and remote destination profiles. Remote destinations are associated with shared
lines of remote destination profiles and access lists. For Mobile Voice Access, the appropriate
service must be activated and the automatically generated media resource is made available to a
router that runs the VXML call application.
) 2010 Cisco Systems, Inc. Troubleshooting Cisco UnifiedCommunications Manager Features and Application Issues 5-69
CSS Handling Since Cisco Unified Communications Manager
7.x
CSS handling has changed since Cisco Unified Communications Manager 7.0.
5-70
Crîn
Review of CSS handling in Cisco Unified Communications
Manager 6.0:
' CSS used for Mobile Voice Access calls: line CSS of shared line
combined with CSS of remote destination profile (partitions of line
CSS considered first)
CSS used for Mobile Connect, rerouting CSS of remote destination
profile only
Changes to CSS handling in Cisco Unified
Communications Manager 7.0:
- New Cisco CallManager service parameter determines CSS for
Mobile Voice Access calls: inbound CSS for remote destination
Trunk or gateway inbound CSS (default value in Cisco Unified
Communications Manager 7.0)
Remote destination profile + line CSS (equivalent to Cisco
Unified Communications Manager 6.0 handling)
> No changes for Mobile Connect
In Cisco Unified Communications Manager 6.0, the CSS that is used for Mobile Voice Access
calls is composed of the line CSS, which is configured at the line that the office phone and the
remote destination profile share, and the device CSS, which is configured at the remote
destination profile. Just as with normal phones, thc line CSS and thc device CSS are combined,
and higher priority is given to the line CSS.
For Mobile Connect. Cisco Unified Communications Manager 6.0 used the rerouting CSS.
which is configured at thc device-level remote destination profile. The CSS of the shared line i*>
not considered.
Cisco Unified Communications Manager 7.0 introduced a new Cisco CallManager service
parameter for Mobile Voice Access calls. The service parameter is called Inbound CSS for
Remote Destination. This sen ice parameter can be set to one thc following values:
Trunk or gateway inbound CSS: This is the default value in Cisco Unified
Communications Manager 7.0. If this option is selected, then Cisco Unified
Communications Manager uses thc CSS of the trunk or gateway from which the Mobile
Voice Access call arrived. If you use this option, then an incoming call from thc remote
destination can access partitions that are specified in the trunk or gateway inbound CSS.
This service parameter does not affect calls from nonrcmole destinations.
Remote destination profile + line CSS: If this option is selected, then Cisco Unified
Communications Manager manages Mobile Voice Access calls in the way in which they
were treated in Cisco Unified Communications Manager 6.x. If you specify this option,
then it will use the combined CSS of the shared line and thc CSS that is configured at the
remote destination profile to determine what partitions the incoming call can access.
Troubleshooting Cisco Unified Communications (TVOICE) v8.0 12010 Cisco Systems. Inc
Access Lists Since Cisco Unified Communications Manager
7.0
In Cisco Unified Communications Manager 7.0 and later, the end user and the administrator
can control access to remote destinations according to the time of day and the day of the week.
Access Lists Since Cisco Uniffr
Communications
Remote destination configuration page is extended by Rtng
Schedule pane.
Ring schedule can be generally enabled (all the time) or explicit time
ranges can be configured.
Time ranges are configurable per day of week:
Enabling the whole day (24 hours)
- Specifying a time range (from, to)
Caller IDs can still be limited by access lists at remote destination
configuration page (at administrator and user web page).
Allowed Access List <Access List> renamed to "Ring this
destination only if caller is in <Access Lisl>."
Blocked Access List <Access List> renamed to "Do not ring (his
destination if caller is in <AccessList>"
To support ToD-based access to remote destinations, the remote destination configuration page
allows the configuration of a ring schedule. This ability applies to the remote destination
configuration page at both the administrator and user web pages.
You can enable the remote destinationgenerallyall of the timeor you can configure
explicit time ranges. The default is that the remote destination is enabled all the time. This
default equals the functionality that Cisco Unified Communications Manager 6.0, which did not
support ToD access control, provided.
When an explicit time range is configured, each day of the week can be disabled, enabled for
the whole day--24 hoursor configured with a from-to time range.
Access lists still can limit caller IDs. Just as in Cisco Unified Communications Manager 6.0,
access lists are applied at the remote destination configuration page. However, the parameters
have been renamed as follows:
The Allowed Access List <Access List> setting is now called "Ring this destination only if
caller is in <Access List>."
Blocked Access List <Access List> is now called "Do not ring this destination if caller is in
<Access List>."
>2010 Cisco Systems. Inc Troubleshooting Cisco Unified Communicalions Manager Features and Application Issues
Operation of ToD Access Control
The figure shows the processingof calls that are received at a shared line that is configured at a
remote destination profile.
Remote destination will ring only during specified ring
schedule (regardless of access list configuration).
If no access list is configured, all callers are permitted during
specified ring schedule.
Call is received at
shared line of remote
destination profile.
Consider these two factors when using ToD access control to remote destinations:
1 he remote destination rings only when thc call is received during the specified ring
schedule. This requirement is independent of access lisl configuration.
If no access list is configured, then all callers are permitted. However, this statement
applies only when thc call is received during the specified ring schedule. If a caller is
permitted by an access list configuration but calls outside the configured ring schedule,
then thc call will not be extended to the remote destination.
["or each remote destination that is associated with the called line, the ring schedule that is
configured at the remote destination is checked in thc following way:
If the call is received outside the configured ring schedule, then thc remote destination will
not ring.
If the call is received within the configured ring schedule, the access list configuration of
the remote destination is checked. If the caller ID is permitted, then the remote destination
will ring. If the caller ID is not permitted, then the remote destination will not ring.
The caller ID is permitted in the following scenarios:
The "Always ring the destination" parameter is selected.
An access list is applied by using the "Ring this destination only if caller is in = Access
List>" parameter, and thc caller ID is found in the specified access list.
An access list is applied by using the "Do not ring this destination if caller is in <Access
List-'" parameter, and the caller ID is not found in the specific! a- cess list.
5-72 Troubleshooting Cisco Unified Communications (TVOICE) v8 i 2010 Cisco Systems, Inc.
Enterprise Feature AccessDTMF Directed Call Park
Cisco Unified Communications Manager 7.0 introduced support for Enterprise Feature Access
for Cisco Unified Mobility users where directed Call Park and retrieval are the options of it.
Enterprise Feature Access
Directed Cal
Since Cisco Unified Communications Manager 7.0, Cisco
Unified Mobility users can access the directed Call Park
feature via DTMF tones:
Cisco Unified Mobility calls can be parked and retrieved from
remote phones.
This feature is implemented by combining the DTMF transfer
feature of Cisco Unified Mobility and the standard directed Call
Park feature.
At the remote phone, the user can use DTMF transfer to transfer
the caller (parkee) to a specific park code (directed Call Park).
The call can be retrieved from any remote or enterprise phone.
No configuration is required, but the following must be enabled:
Directed Call Park number (to enable directed Call Park)
Mobile Voice Access and Enterprise Feature Access (to
enable the use of DTMF transfer).
Using directed Call Park from a remote phone, a user parks a call and inputs a user park code.
The user can then use the code to retrieve the call or have someone else do so.
This feature is a combination of the existing dual tone multifrequency (DTMF) transfer and
directed Call Park features.
When the mobile phone user is on an active call, the user can use the DTMF transfer feature to
park the call by transferring the parked party to the park code that the user entered.
The call can later be retrieved from any phone: enterprise or remote.
This feature does not require any special configuration. However, several existing features must
be enabled for DTMF directed Call Park to work:
One or more directed Call Park numbers must be configured. The CSS that is configured at
the remote destination profile must include the partition of the desired directed Call Park
numbers.
Mobile Voice Access must be enabled for the remote user.
Enterprise Feature Access must be enabled.
Cisco Unified Mobility Enterprise Feature Access is disabled by default and is enabled by the
Cisco CallManager service Enable Enterprise Feature Access parameter under "Clusterwide
Parameters System Mobility." The following Enterprise Feature Access codes exist and can
be configured within the same service parameter section:
Hold: default value *81
Exclusive hold: default value *82
Resume: default value *83
) 2010 Cisco Systems, Inc Troubleshooting Cisco Unified Communications Manager Features and Application Issues
5-74
Transfer: default value *H4
Conference: default value *85
DTMF-Directed Call Park Considerations
If a Cisco Unified Mobility remote destination is the parker of the call, then the call reverts to
the shared line of thc enterprise phone that is associated with the remote destination profile.
DTft/IF-Directed Call Park
Considerations
If parked calls are not retrieved on time, they revert to the
parkerphone:
- Ifparker is a Cisco Unified Mobility remote destination,
call reverts to shared line of enterprise phone.
- Such reversion causes all remote destination and
enterprise phones to ring.
If this behavior is not wanted, enter a specific reversion
number (e.g., attendant number) and reversion CSS at the
directed Call Park number.
Cisco Unified Communications Manager plays reorder tone if
parker attempts to park a call by using an invalid park code.
Cisco Unified Communications Manager plays a busy tone if
parker attempts to park a call and the park code is busy
(already used to park another call).
In both situations, the user can try another park code.
This behavior causes all configured remote destinations to ring when the call is reverted.
If this behavior is not wanted, then enter a specific reversion number at the directed Call Park
numbers that arc used by the remote phones. Make sure to configure thc reversion CSS also and
include the partition that is assigned to the entered reversion number. The reversion number is
usually the number of an attendant or any individual or group that is related to the Cisco
Unified Mobility user.
These issues could be encountered during an attempt to park a call:
If the entered call park number is already in use, then Cisco Unified Communications
Manager plays a busy tone to the parker.
If the entered call park number is invalid, then Cisco Unified Communications Manager
plays a reorder tone to the parker. The number is considered invalid when it is inaccessible
to the caller. Inaccessibility occurs when the number does not exist or when the caller does
not have the right, based on partitions and CSS, to access the number. Because the CSS of
the remote device profilenot the rerouting CSSis used in this case, make sure that the
configured CSS includes the partition of the directed call park number.
In either situation, the user can try entering another park code when hearing the reorder or busy
tone.
2010Cisco Systems, Inc. Troubleshooting Cisco UnifiedCommunications Manager Features and Application Issues 5-75
Enterprise Feature AccessDusting Feature
This section describes the dusting feature of Enterprise Feature Access, which was introduced
in Cisco UnifiedCommunications Manager 8.0.
5-76
JU2&&L3m\hist ifc *f* *-,,
Dusting feature can move a single call, conference, and session
collaboration among mobile phone, PC, and desk phone.
After call is dusted to terminating device, it can be either auto
answered or flash, depending on the terminating device capability.
Original conversation can be continued until the dusted call is
answered.
Cisco Unified
User Mobile
User Desk Phone
The complete dusting feature can move a single call, conference, or session collaboration
among mobile phone. PC. and desk phone by dialing a determined DTMF combination.
After the call is dusted to a terminating device, it can be either auto answered or it shows as
flashing, depending on the terminating device capability. To support auto answer, the phone or
PC needs to detect proximity.
One of thc major benefits of the dusting feature over desk pickup is that original conversation
can be continued until the dusted call is answered at the new terminating device.
The dusting feature is implemented by using two service parameters:
Session Handoff Alerting Timer: Determines the time when the dusting call Hashes.
Default value is 10 seconds,
Enterprise Feature Access Code for Session Handoff: Determines thc DTMF feature
code to trigger dusting. Default value is *74,
Thc dusting feature works as in thc following scenario:
User A calls the user B desk phone. Using the single number reach feature, user B answers the
call on the mobile phone and the desk phone goes into rcmotc-in-use state. User B dials *74
(Dusting DTMF code). The user B desk phone (supported SIP and SCCP phone) stans flashing.
User B still talks with user A by using the mobile phone. To move the conversation to the desk
phone, user B has to answer the call from the desk phone before the Session Handoff Alerting
Timer (default 10s) expires. After the timer expires, the desk phone stops flashing. User B can
continue the conversation by using the mobile phone.
Another scenario would trigger the move by using a softkey event (softkey event that is
embedded inside the SIP Refer message). When Cisco Unified Communications Manager
receives the SIP Refer, it parses the XML content that is associated with the SIP Refer message
that identifies the "move" softkey event and triggers dusting.
When a user dusts a call, a new call wilt be presented on a desk phone. While thc desk phone is
flashing, the following features on that desk phone, for the call being dusted, will not be
triggered:
i Divert
Call Forward AH (CFA)
Do Not Disturb (DND)
Call Forwarding
Other midcall features that are triggered by DTMF (Hold, Transfer, and Conference)
If a user dusts a call and does not answer from the desk phone within the alerting timer, the
existing remote-in-usc state on the desk phone is lost. Thus, the desk phone loses shared line
functionality following dusting. The user cannot perform midcall features for that call like Hold
from Mobile (using *81) and Resume from desk, or desk pickup. But thc user can dust the call
again to resume it from the desk phone.
2010 Cisco Systems, Inc. Troubleshooting Cisco Unified Communicalions Manager Features and Application Issues 5-77
Cisco Unified Mobility General Issues
5-78
The section introduces the general issues that users experience when using Cisco Unified
Mobility.
Common issues:
User cannot switch a call between mobile and desk phones.
Associated desk or mobile phone does not ring with inbound call.
Full PSTN number presented instead of associated desk phone
number.
Internal called party cannot be reached.
User can not redirect a call received at desk phone to the remote
phone.
User cannot reach Mobile Voice Access or use it for outbound
calling.
1 User cannot park, transfer, resume or hand off a call on remote
phone
The common issues that are experienced with Cisco Unified Mobility and discussed in this
sson are the following:
User cannot switch a call between thc mobile and desk phones.
Associated desk or mobile phone does not ring with inbound call.
Complete PSTN number is presented instead of associated desk phone number.
User cannot redirect a call that is received at thc desk phone to the remote phone.
User cannot reach Mobile Voice Access or use it for outbound calling.
User cannot park, transfer, resume, or hand off a call on the remote phone.
Troubleshooting Cisco Unified Communicalions (TVOICE) vB.O 2010 Cisco Systems. Inc
Troubleshooting Cisco Unified Mobility Mobile
Connect
This topic describes how to recognize and troubleshoot most common Mobile Connect issues
and how to troubleshoot them.
User Cannot Operate
I want to work witti
Mobile Connect.
Problem.
Error message
received wtien
Mobility button used.
Verify that phone
device is associated
with Cisco Unified
Mobility end user.
Problem:
Mobility button is not
available.
Verify that correct
softkey template is
used for phone device.
The initial problemcan be experiencedwhen a user wants to operate Mobile Connect to
disable or enable Mobile Connect or to redirect a call to the remote destination.
The user might find that the Mobility button does not exist on the office phone.
If the button is unavailable, the most probable cause is that the softkey template with the
Mobility button is not attached to the office phone device. The softkey template must define the
Mobility button for two call states: on hook and connected.
Whenthe softkeytemplate issuehas beensolvedand the Mobilitybuttonappearson the office
phone display, theusercanworkwithMobile Connect. Theuser presses the Mobility softkey
button.
But the user might get an error message on the phone display, saying that thc user is not a valid
Mobile Phone user. That might be another potential problem.
The most probable reason for this message is that the user, who is enabled for Mobile Connect,
might not be properly associated with the office phone device. The user must become an owner
of the phone device in the phone device configuration page.
2010 Cisco Systems, Inc Troubleshooting CiscoUnified Communications ManagerFeatures and Application Issues 5-79
Troubleshooting Inbound Internal and PSTN Calling Problems
Mobile Connect allows the receipt of a call on an office phone and on several remote phones.
IroubU," nootii'tj l*ihom
PCTfJ C isihrs/s C-CAhltitiii..
Typical problem scenario:
Inbound call comes to office phone 2001 from PSTN or internal phone.
Office phone 2001 rings.
The remote phone thai is associated with 2001 does not ring.
Cisco Unified
Mobile Connect
Gateway
Remote phone (.
does nol ring. .'
The end user is associated with the office phone and the defined remote destinations that
represent the remote phones.
When a call is received at the office phone 2001, the office phone starts ringing.
When the call is received at the office phone, all the remote destinations that are associated
with the office phone 2001 should start ringing as well.
But in this problem scenario, the remote phones do not ring. Several reasons could cause this
experience.
mm
Troubleshooting Inbound Inten
PSTN Calling Problems (Cont,)
Cisco Unified
Communications Manager Is Mobile Connect enabled?
Mobile Connect _ .
Gateway
/ User disabled Mobile
/ Connect on office phone.
PSTN.
___^ Admin disabled inend user,
or remote destination
configuration.
Automatically disabled based on *"
remote destination ring schedule.
D Enable Mobile Connect
Ring Schedule
No ring on Sunday
One possible cause is that Mobile Connect is not enabled.
If the user has disabledthe MobileConnect functions on the officephone, then the user has
blocked the remote phone from ringing.
The end user is associated with one or several remote destinations. These remote destinations
represent the remotephones that the user owns. If a CiscoUnifiedCommunications Manager
Administrator disables Mobile Connect in a remotedestination, then the remotephonethat the
remote destination represents will not ring.
In Cisco Unified Communications Manager 7.x and later, the remote destination can be
configured with a ring schedule. The ring schedule lets you define separate times of the day and
week when the particular remote destination is reachable. If the call falls outside of the
configured ring schedule, then the remote phone will not ring.
Troubleshooling CiscoUnified Communications ManagerFeatures and Application Issues 5-81
5-82
Troubleshootinq Inbound Internal ..n
Cisco Unified
Mobile Connect
Remote Destination anc
Issues
Gateway
Remote Destination Profile
2001 in
Phones jf,
Partition.- W&&.
RDP not associated
; with end user
ft-'*'"
RD = Remote Debti nation
ROP = Remote DesMaR
Line 2001 in Gateways
Partition
Rerouting CSS
RD delay before ringing timer set
too high, nnging delayed
Line 2001
Delay Before Ringing
'Timer: 30000
Other Cisco Unified Communications Manager configuration issues can prevent the remote
phone from ringing. Many issues are caused by misconfiguration of the remote destination and
If the office phone line and the line of the remote destination profile are not in the same route
partition, then the remote phone will not ring when a call comes to the office phone. Because
these two lines represent a single shared line, they must be assigned to thc same partition.
If thc remote destination profile is not associated with the end user who holds the office phone,
then the remote destination will not ring. Fix the association at the remote destination profile
configure page.
If the remote destination line is not marked and associated with the remote destination profile
line, then the remote phone will not ring. Check thc Line check box at the remote destination
configuration page.
The remote destination might not ring due to no or incorrectly set Rerouting CSS at the remote
destination profile. The rerouting CSS must have an access to PSTN calling to reach thc remote
phone.
If thc previously mentionedconfiguration is correct, then verify that the Delay Before Ringing
Timer in the remote destination is not set to too high a value. For example, in the figure, the
timer is set for 30 seconds. In this situation, the remote phone will ring, but the ringing will be
delaved bv 30 seconds.
Troubleshooting Cisco Unified Communications (TVOICE) u8 0 2010 Cisco Systems, Inc
Troubleshooting Inbound Intern;
PSTN Calling Problems (Cont)
Cisco Unified
Mobile Connect
Access List Issues
:.Rerftote Destination.-.
Do not ring if in; Blocked-
ACL
CLI R active?
312 555-1000
Outside Caller
(Chicago)
Blocked
Directory Number: 312XXXXXXX
Privata
Not Available
An administrator or user canalso preventthe remotephone fromringingfor particularcalling
numbers by defining an access list and placing the calling number in this list.
Auser is associated with the remote destination that represents the user remote phone. This
remote destination can select two types of access lists. Thc access lists can be defined as
blocked or allowed access lists.
The blocked access list blocks all listed entries. Any entry not on the list is allowed.
The allowed access list works in the opposite way. The entries that are on the list are allowed,
and entries that are not on the list are blocked.
The access list configures three types of entries:
A specific directory number, which can include common wildcard characters
Private
Not available
The lasttwotypes areused insituations inwhich thecalling IDis suppressedwhen the
Calling Line Identification Restriction (CLIR) is in effect or whenthecalling IDis not in the
message.
In the figure, the outside caller calls from the Chicago PSTN area, but the blocked access list
blocks all calls that start with 312 fromringing the remote phone.
Troubleshooting Cisco Unified Communications Manager Features and Application Issues 5-83
totli l. I lhn:.ill !ht"li1.t1 usn'
Cisco Unified
Communications Manager CSS and OutboundCall-Routing Issues
Mobile Connect . ,
Gateway
Office Phone .
CSS1 Remote
t
Rerouting CSS: Remote
Other Call-Routing Issues:
Dial Plan, Globalization, or
Localization
Cisco IOS Gateway Issues
Destination Number
When a call leg to the remote phone is being set up, Cisco Unified Communications Manager
performs all the common outbound call-routing tasks. It processes the remote phone number as
any called number thai undertakes call-routing and permissions verifications.
The route pattern that routes the call to the remote phone via the PSTN is in the PSTN partition.
The remote CSS contains only the Phones partition: the partition to which the office phone line
is attached. Thc user is associated with the remote destination profile that sets the permissions
for outbound remote phone calls by using the rerouting CSS. In the example, the rerouting CSS
is set to the remote CSS described earlier.
The remote CSS does not have access to the PS'fN route pattern because its route partition is
not a member of the CSS. Therefore, the call-setup procedure will fail and the remote phone
will not ring with the office phone.
In addition to the CSS issue, incorrect configuration of the dial plan and called number
globalizationor localization on Cisco UnifiedCommunications Manager can also cause the
remote-phone call setup procedure to fail.
The PSTNgateway might be misconfigured, too. Typically, 11.323 or SIP gateways need dial
peers to a route call to the PSTN, or they might perform incorrect digit manipulation.
Troubleshooting Cisco Unified Communicalions (TVOICE) v8 0 )2010 Cisco Systems, Inc
Tracing Inbound Call to Mobile Connect
This section shows the trace of an inbound call to Cisco Unified Mobile Connect.
Tracing Inbound Call to Mobile
SPKOC gatCtrlPid - ':; ll i r.ijM -a = \ '. '-I4.41 , inputCtrlPid-(1.100,195,1)
DbHobility: getMa tChetJRemOeSt Starts: ::nôi-.i- i " ZTAi i 4 I " *"
DbMobility; getHatcheuRenDest!
DbHobi 1 i ty: getKa tche dRenDeat:
DbKobility: getxatchenReaDeat:
DbHobility: getHatchadRamDest sas1
... truncated ...
Digit Analysis: starDaReq; Matching Legacy Kumaric,
digitn=2001,1.100,150.1.63"10.1.250.101""
Digit analysis: matchtpi.'2", fqcn.", cn."5554444" ,plv
pas."Internal Pt" , TodFilteredPae- "InternalPt- ,
dd-"2001",dac-"l") 1,100,150,1.6 3*10. 1.250.101""
Digit analysis: analysis results! 1,100, 150,1.63" 10 .1.25
PretransfornCallingPactyNumbar-5554444
! CallingPar tyNumber-5E5 4*4
'DialiogPartitioa-Interaal Pt
! 3ialingpattern.=200 1
:FullyQualltiedCalledFartyNunfcer.2001
' Dial ingpat tern Regulars*pies sion* 1200 1!
DialingMhere-
' PatternType-Enterpriae
PotentialHatchea-NoPotentlalNatchesExist
partial match '
This trace demonstrates the stages of the call where the PSTN caller 5554444 is calling the
office phone 2001 that is associated with the remote destination 6065554444.
The trace begins with an indication of an inbound call from 5554444. The calling number is
verified if it would be a known remote destination. The trace shows the result of this check
the 5554444 is not a known remote destination.
Thc trace continues with the digit analysis. Thc inbound call came from an MGCP gateway that
has applied Significant digits=4 (not seen in the trace), thus the called number is 2001. The
inboundgateway was configured with Inbound CSS that lists Internal_Pt route partition. The
destination 2001 is in this route partition. The digit analysis ends with the summary results and
reporting of no more potential matches.
)2010Osco Systems, Inc. Troubleshooting Cisco Unitied Communications Manager Features and Application Issues 5-85
iek| Int
SMDMSharedData::findLoealDevice - Nan=e=2 0Ql:1541598d-0bld-c94e-48a1-
a0 07 6Balcfl2 Key= 6c4ddcBf -00 3 -d31 3-a2 63-6S77e8af52 26 isActvie=l
Pid.(1,154,5) found 1,100,150,1.63"10.1.250.10 1"*
LineCdpc (20) i diapatchToAllDevicefl-, sigName-CeSetupReq,
device-00*065554444 :;doe -Idp 1, 100, 150, 1. 63 "10. 1.250. 101 "*
LineCdpc(2Q) : -dispatchToAHDevices-, aigName^CcSetupReo.,
device-SEP0 024C4454 ACS 1,100,150,1.63*10. 1.250.101"'
SNRD: :jit CcSetupReq (0 0000 03) SendingDchanlSl . ci 1319 50291),
featureCallType [01 1,100,150,1.63"10.1.250.101"*
TODAcceseHelper - isTCDAcc essAUoved: found todPkid - d0c2bl5f-e6bd-43Se-
6d0d-976i5eiSia39. ... . .' . r- .:-!,:. : -,,- ;*=.,; .-**
TODAcceSBHe iper - iSTODAccessAllowed: : ./ ..'.' Ji'i-.i! .-li '. ;
TODAcceBsHelper - iaTODAccessAllcwed: no active access list is found, call
is allowed Ier given cpgn ""=<
SNRD : !iit CeSe tupR eq 10CD00 031 ;-.-.? I ' -i. . . >j .... - . = ', .. i
. - 1,100.150,1.63"10.1.250.101"*
16:19:51.144 SNRD::wait CcSetupReq (0000003) Low/Normal Priority
teaturelype 101 Call, In PKidLiat, featurePriority[1J ,
Normal? eaturePioiity II] , nonTargetPolicy [Extend] . cepn [ce4 844 9e-5af 7-d3 77-
5b7l-361f 8e5bb9e7l , extending Call. ; 1, 100, 150,1. 63"1 0.1. 250 .101"*
After the digit analysis, the internal phone with 2001 was found as active (registered). This is
the office phone that has been associated with thc remote destination 6065554444 based on the
shared line of remote destination profile jdoe-rdp. Cisco Unified Mobile Connect attempts to
dispatch the call to all devices on a shared line 2001; in this ease, the office phone and the
remote destination 6065554444. Cisco Unitied Mobile Connect continues with the verification
if this remote destination has a ToD access list applied. Based on a time zone, the time and the
day of week (1-Monday) uhen the inbound call came, the call is allowed to extend to the
remote destination (in this case, no ToD access list exists).
The trace shows the call setup request, with normal priority, for the call to the remote
destination 006065554444 (includes PSTN access code 0 and long-distance prefix 0) from the
original caller 5554444 at the bottom.
5-86 Troubleshooting Cisco Unified Communications (TVOICE) vB.O 2010 Cisco Systems, Inc
Tracing inbound Call to Mobile Connect {Cont.
CellProxyl
*8 3,XferCode
DbMobility: ii
10) - Hold feature code B1, ExcBoldCode *82, Sesum*Coda
84, ConfCode *BE, DubtingCode -74|1,100,193,10.1"***
itRamDest: device pkid [ce4 B44"e-5af 7-d377-St>71-364ffle5bb9e7]
profile pkid [5a79cad5-365d-0bbl-4c65-3457ael
iflSmartPhone [0] **"
DbHobil ityRemDestTable:initRemDeBt: initial!i
1006065554444] *-*'
' :<>-,==- dump: cnumber . 006065554444, devicaPkid - ce48449a-
5af7-d377-5b71-364Be5bb9e7, renDeBtProiilePkidStr 579cad5-365d-Obbl -
4cS5-d457a78a6a0. isHobilaPbona 1. isDualHods 0, isSmartPhona 0,
isSNREnabled- 1, anawerTooSoonTim*r - 15 00, anawarTooLateTimer 190 00,
delayBeforeRingingCellTimer - 4000= uaerld * jdoe, tlmaZonelndax - 22,
description ]doe-rd, uil |*"**
DbMobilltyRemneatTableînitMobilityUBer: -- created mobility user|*"-*
<\ '..::, - ;-=- dump: userld - jdoa, isIVREnabled - 0,
rMxEeshPickupWaitTiiw. 10000, userLocale - 1|**'*
CellProxyl 10) - StartTranaition - cellNum-006065554444, dn-2001.
smart'0, oobile=l, dualnode-0, anrEaabled-1, anBwer2Soon1500,
answer2Lata.19000, eallSnraenTimar-4000. dalayBPRing-4000,
rerouteCSS-07ccc3e6-dBIS-ae49-12 35-ec6aOS74aOf1, userLocale.1,
deskPickup-10000, mDialingPid- <0, 0, 0), is3G0, aasociatedUaerLocale. 1,
default User Loca le==l 1,100,193,10.1***"
0] , sDualnvode
a remde
(01 ,
This section of the trace output shows the summary of the call to the remote destination.
It begins with the summary of Enterprise Features (remote destination is the mobile phone).
Then thc call to the remote destination is initialized, and the Cisco Unified Mobile Connect
summarizes the remote destination 6065554444, including all timers and the user jdoe.
) 2010 Cisco Systems, inc. Troubleshooting Cisco UnifiedCommunications Manager Features and Application Issues 5-87
Tracing inbound C;
CellProxyl 10) - null CcSetupReq, etSetup-acdpn. nd=D06C65554M 4 ,
ce HPickup-0, CcSetupReq- ncdpn. ccmUrl. mCcmUrlType=0 , ccSetupKeq-
>cdpn.ccmUr1.mS ipUrl.nug er.c str ()., ccEetupReq-
>cdpn.ccmUr1,aSipUrl.mHoat.c str () = , ceSetupHeq-
>cdpn.ccmIlil.mSipUrl.murl,c str(}= lrg-b77c6d54-0 972 - 22f2-1617-
86 48b0feE22b 1, 100, 150,1.63*10.1.250.101"*
Bijit Analysis: star DaReq: Matching Legacy Numeric,
digits=006065554444 1,100.150,1.63'10.1,250.101"*
Digit analysis: natch[pi = 2-. qcn="", cn=*5554444",plv=5",
pss="HQ Emergency PtiHQ Intl Pt:HQ LD Pt:HQ Local Pt:Internal Pt" ,
TndFilteredPas="HC Exergenjy Pt:HC Intl Pt:HQ LD Pt:HQ Local Pt:Interna 1 Pt"
, dd="006C65554444".dac-"1") 1,100,150,1.63*10.1.250.101**
Digit analysis: analysis results 1,100,150,1.63"10.1.250.101"*
Pre transforms 11 ingPartyNumber-55544 44
DialingPartition-Hfl LD Pt
Dial ingPat tern =0.0 11-91 I
FullyQualifled CalledPaityNumbei = 006065554444
DialingpatternRagularExpreaeion. (0) 10 [i- 9j 10-91*)
Dialmgwhere-
PatternTypeaEn terpriae
PotentialHatches-NoPotentialMatchesExist
The call setup to the remote destination has been requested. Since this moment, the call has
been treated as any other outbound PSTN call. It undertakes the digit analysis as shown in the
figure. Thc CSS shown in the trace output (TodFtlteredPss) represents the Rerouting CSS of the
When no more potential matches exist, the trace shows the matching route pattem 0.0[ 1-9]! for
the outbound calling The trace would then continue with route list, route group, and gateway
selection and the call extension to the PSTN (not shown in the trace).
Troubleshooting Remote-Phone Calling Problems
Another set of problems relates to a user who calls the office by using a remote phone.
Troubleshooting Remote-
Problems
Typical remote-phone calling problems:
Complete PSTN number instead of associated office phone
number is presented.
Cisco Unified
Mobile Connect
Gateway
1 408 555-1234
Remote
Phone of
2001
The remote phone is the PSTN or mobile phone, so it is assigned the complete PSTN E.164
number. However, whenever a user calls from the remote phone through Mobile Connect, the
extension of the associated office phone should be presented to the called parties.
If Mobile Connect is configured inappropriately, the PSTN number of the remote phone,
instead of the extension of the associated office phone, is presented to the called party.
Or, in the worst case, internal phones cannot be reached at all.
2010 Cisco Systems, Inc Troubleshooting Cisco UnifiedCommunicalions Manager Features and Application Issues 5-89
Manager
Mobile Connect
Digit
CSS or Inbound Gateway
Gateway Problems
i 408 555-1234
Destination number.
Service Parameters.
Matching Caller IDwith Remote Destination1 Partial Malch
Number of Diqits for Caller ID. 10
--'""** A
4085551234 %H
. . 4085551234 J|
Admin disabled Mobile
Connect for end user
Remote Destn ation I ine
Not Associated with Office
Phone Line
Different Partitions
Line 2001 in Gateways
Partition
If the wrong number is presented, several configuration issues could be at fault.
First, verify that the remote destination line is associated with the office phone line. The Line
check box must be marked at the remote destination configuration page.
Also verify that the remote destination profile line is attached to the same partition as the office
phone line. If the partitions are different, the line is not being shared.
Alternatively, an administrator might have disabled Mobile Connect for thc end user. This
action would also cause the number-presentation issue.
If all the pre\ ious options have been checked and the PSTN number is still being presented,
then verify' how caller ID matching is set up in the Cisco CallManager service parameters.
The Matching Called ID with Remote Destination serviee parameter might have been set to
Complete Match. If the calling number was modified as it was being received for instance, if
the number globalization or localization process added a prefix then the number might not
exactly match the value that was set at the remote destination configuration page.
In situations such as these, set the Matching Called ID with Remote Destination service
parameter to Partial Match. This setting will enswe that only some determined portion of the
calling ID must match. Use the Number of Digits for Caller ID service parameter to configure
how many digits must match. The parameter is set by default to 10 digits, which suits most
situations.
5-90 Troubleshooting Cisco Unified Communications (TVOICE) vB.O 12010 Cisco Systems, Inc
Troubleshooting Problems with Call Redirection to Remote
Phone
Difficulties can also be experiencedwhen a mobility user wants to redirect to the remote phone
a call that was received at the office phone.
Troubleshooting Problems with Caii
Redirection to Remote Phone
Call redirection problem:
* User cannot redirect to the remote phone a call received at
office phone.
User gels error message on office phone display.
Cisco Umfied Communications
Manager Mobile Connect
2002 calls 2001.
' user answers and attempts lo redirect to remote pnone.
140B 555-1234
Remote
Phone of
2001
In the figure, the caller at 2002 places a call to 2001. The Cisco Unified Mobility user at 2001
answers the call at the office phone.
The user then decides to redirect the answered call to the remote phone. Perhaps the user is
leaving the office but wants to continue the conversation.
The user presses the Mobility buttonon the office phone and gets the message that the
redirection was not successful.
This behavior can have several causes.
2010 Cisco Systems. Inc. Troubleshooting Cisco UnifiedCommunications Manager Features and Application Issues 5-91
Many causes as before:
SottKey button template nol
associated or missing Mobility button
Mobile Connect disabled
RD 'umoer not configured properly
= RDP line nol in same partilion as
office phone line
RD line not associated with office
pnone line
Cisco Unitied
Mobile Connect
Gateway
Reroulmg CSS in RDP does not
permit calling to remote phone
Dial plan, globalization, or localization
issues.
Outbound calling issues on gaieway
i4oesr>r>-i2:s4
Remote
RD = Renxrte Debti rial ion
RDP = Remote Deslinalion Profile
The reason for the unsuccessful redirection is mostly the improper configuration of the Mobile
Connect configuration elements described earlier as follows:
The softkey button template might not be associated with the office phone device, or the
template is missing the Mobility button. In this case, the office phone user will not even sec
the button to perform the redirection.
Mobile Connect might be disabled for the user.
The remote destination number might not be configured properly, so the Cisco Unified
Communications Manager or PSTN does not know how to route the outbound call.
The remote destination profile line might not be in the same partition as the office phone
line.
The remote destination line might not be associated with the office phone line.
The rerouting CSS in the remote destination profile might not permit calling to the remote
phone. This issue is a typical class of service (CoS) problem.
The Cisco Unified Communications Manager dial plan, globalization, or localization
processes might not work properly.
Outbound calling issues on the PSTN gateway might prevent the outbound call from being
established.
An end user is nol an owner of" the phone.
Troubleshooting Cisco Unified Mobility Mobile
Voice Access
This topic describes howto identify and troubleshoot Mobile Voice Access incoming and
outgoing call problems.
Troubleshooting Mobile Voice Access
Incoming Call Problems
Cisco Unified
Mol^V!>.-.V..^:- 2000
Office
Phone
2001
Gateway Issues:
Inbound PSTN gateway calling problems
MVA directory number digit manipulation
MisconfiguredIVRon the gateway
Cisco Unified Communications Manager Issues:
Gateway inbound CSS blocks incoming Mobile Voice Access calls.
Cisco Unified Mobility Mobile Voice Access allows a caller to place external PSTNdestination
calls through the enterprise facility. A user calls the Mobile Voice Access resource number,
authenticates with the IVR, and then enters the destination number that thc user wants to call.
Cisco Unified Communications Manager then places a call to the destination, on behalf of the
user. When the call to the destination is established, the user can switch between thc office
phone and the remote phone.
Several problems could be encountered during this process. The first is the problem of inbound
calling to Mobile Voice Access.
The user calls the Mobile Voice Access resource number but either cannot reach the IVR or
cannot reach the destination number after entering it.
If the user cannot reach the Mobile Voice Access IVRsystemand no voice responds to thc call,
the issue is typically caused by a misconfigured IVR in the gateway or a problem with the
gateway inbound dial peer.
If thc user reaches the Mobile Voice Access IVR systemand enters the PIN and the destination
number but the call still fails, the problemis either the called-number digit manipulation on the
gateway or the configuration of inbound calling on the gateway. The gateway CSS needs access
to the Mobile Voice Access media resource number that typically is placed to a route partition.
The mediaresource numbcrmust be accessible by the gatewayInbound CSS.
Other Cisco Unified Communications Manager configuration issues can also cause an
outbound calling problem. The outbound call leg might fail because of a misconfigured CSS.
which is always the combination of Line CSS and remote destination CSS.
>2010 Cisco Systems. Inc. Troubleshooting Cisco UnifiedCommunications Manager Features and Application Issues 5-93
Troubleshooting Mobile Voice Access Outgoing Call Problems
Some issues can also be experienced in the outbound direction when a user uses Mobile Voice
Access to call an external destination.
Troubles!
Outnoinq
Mobile Voice Access 2000 H.323 IWR 2000
Gateway
vc-.
Manager Issues:
CSS blocks outgoing Mobile Voice
Access calls.
Gateway Issues:
Outbound PSTN gateway calling problems
Wrong number dialed
I
CSS determined by concatenation of Remole
Destination Profile line CSS and device-level CSS
In the figure, the user has established a call from the remote phone to the Mobile Voice Access
resource numbcr. authenticated by using the PIN, and entered thc destination number to call.
However, the user cannot call the outside destination by using Mobile Voice Access.
If the user is convinced that the called number is valid, the issues could be on the gateway. For
instance, the H.323 or SIP gateway might not know how to manage the outside calls. These
gateways use dial peers for outbound PSTN call routing and can be misconfigured, or the
gateway PSTN circuits might be exhausted.
Cisco Unified Communications Manager can block outgoing Mobile Voice Access calls
because of CSS issues. The CSS might not provide permission to call outside destinations.
Which CSS might cause the problem?
Cisco Unified Communications Manager determines the inbound call destination that is based
on the sen ice parameter Inbound CSSfor Remote Destination. This service parameter and the
choice between two options were introduced in Cisco Unified Communications Manager 7.(1,
Outbound call routing for Mobile Voice Access calls always uses a concatenation of the
Remote Destination Profile line CSS and device-level CSS. Therefore, it is important that you
configure these CSSs appropriately to provide access to any route patterns that arc necessary
for off-net or PSTN access. Precise configuration will ensure proper outbound call routing from
remote-destination phones.
Troubleshooting Enterprise Feature Access and
Dusting Feature Problems
This topic describes how totroubleshoot typical Enterprise Feature Access anddusting feature
problcms as user cannotpark, transfer, or resumethe call on remotephone.
Troubleshooting Mobile Voice
Enterprise Feature Problems
Cisco Unified
Mobile Voice Access: 2000
H.323
IVR: 2000
Cisco Unified Communications Manager Issues:
Enterprise Features Access disabled.
Wrong enterprise feature codes configured.
ClHlTXj
40f 555-,d00Q. A/ Remote
IN.Destinations'; Phone of
MVA = Mobile Voice Access
When the call to the destination is established by using Mobile Voice Access, the user might
want to switch phone devices.
The user entersspecial DTMF digit combinations, suchas *82 or *84, that shouldpark,
transfer, or resume the current call.
But nothing happens, and the user finds that the Mobile Voice Access systemdoes not
recognize these special DTMF codes. The phone call goes on as initially established.
Several configuration issues could cause this problem.
The most typical problem is usingthe wrongDTMFrelaybetween the PSTNgateway and
Cisco Unified Communications Manager. Onlyout-of-band DTMFrelay is supported in
Mobile Voice Access scenarios.
Cisco UnifiedCommunications Manager sets thc Enterprise Feature Access in Cisco
CallManager seniee parameters. Enterprise FeatureAccess is disabledby default, or the codes
that the user believes are correct might be configured differently.
When a remote phone is not a smart phone and a call to the remote phone is anchored through
Cisco UnifiedCommunications Manager, the user can hang up the remote phone and expect to
see a Resume softkey on the user office phone to resume the call. The user cannot resume this
call on the user desktop phone.
i 2010 Cisco Systems, Inc. Troubleshooting Cisco UnifiedCommunications Manager Features and Application Issues 5-95
If the calling party receives a busy, reorder, or disconnect tone after hanging up the remote
phone, then the remote phone provider probably did not disconneel the media. Cisco Unified
Communications Manager cannot recognize this circumstance because no disconnect signals
came from the provider. To verify whether this is the case, let the calling party wail 45 seconds,
when thc seniee provider will time out and send disconnect signals, after which Cisco Unified
Communications Manager can provide a Resume softkey to resume the call.
To solve this specific problem, you can take these actions:
Add the voice call disc-pi-off command to the gateway. This command enables the H.323
gateway to treat a disconnect message with progress indicator (PI) like a standard
disconnect without a PI. Use this command if the gateway is connected to a switch that
sends a release immediately after it receives a Disconnect with PI.
For the Cisco CallManager seniee, set the Retain Media on Disconnect with PI for
Acti\e Call seniee parameter to False. This parameter specifies the behavior of Cisco
Unified Communications Manager on receiving a PRI DISCONNECT or 11.225
ReleaseCompletemessage with a progress indicator of 8 (inband informationavailable) for
thc active call only. A value specifying True will hold on the media connection, so thc
other side will receive an announcement or inband tone. False will tear down the call
immediate! v.
Troubleshooting Dusting Feature Problems
This section describes what to verify if the dusting feature does not work as expected.
Troubleshooting Dusting Feature Pri
Cisco Unified
K.!i^~ -Communications Manager
2001
User attempts to
hand off the call.
Problem: Desk phone does not flash!
Check the following:
End User
Remote
Phone of
, , 2001
DTMF: *74 2i Mw.N&s
\h>? Crjll
Desk Phone Owner
j r""
[iMiFmdt PBrirtm rSr
thnt*rw* PirtnwtFn (&
k
m
Remote Destination
Profile
If a call that is being dusted from a mobile phone does not flash at the desk phone, verify the
following:
The Owner User ID for the desk phone matches the User ID of the associated Remote
Destination Profile.
The Enable Enterprise Feature Access service parameter is set to True, and the other
DTMF features (Hold-*81, Resume- *83)are working.
The dusting DTMF code Enterprise Feature Access Code for Session Handoff and the
Session Flandoff Alerting Timer service parameters have the correct values.
) 2010 Cisco Systems, Inc. Troubleshooting Cisco Unified Communications Manager Features and Application Issues 5-97
Summary
mt*Bx/i***na*j da**_j * >Mj?ii3&^&zii-ms&.Lt0L~. uae'.vB&ijasisaK"'! * -~ a a,; *.
,-iinimarv
Cisco Unified Mobility general issues include problems to
switch a call between mobile and desk phone, desk or mobile
phone does not ring, wrong numbers presented, reachability
and redirection problems, and problems with park, resume,
and hold
When troubleshooting Cisco Unified Mobility Mobile Connect,
venfy that CSS, remote destination profile, remote
destination, and end user are properly configured.
Troubleshooting Cisco Unified Mobility Mobile Voice Access
includes verification that CSS does not block outgoing Mobile
Voice Access calls and that a gateway has correct
configuration.
When troubleshooting Enterprise Feature Access problems,
make sure that the Enterprise Features Access is enabled
and that correct enterprise feature codes are configured.
In this lesson, you have learned to describe the common issues that are related to Cisco Unified
Mobility and how to identify thc most likely causes of these issues.
References
Cisco Svstems, Inc. Cisco Unified Communications Manager Features andServicesGuide
Release 8.0(1), April 2009 and updated April 2010.
hup. viwwciseo.com en I S docs voice ip comm cucm'niimin K_0_ I'ecml'cat
fsLid-M)j-=jrn html
Lesson 4
Native Presence Issues
Overview
The lesson reviews native Cisco Unified Communications Manager presence using Skinny
Client Control Protocol (SCCP) and Session Initiation Protocol (SIP). The two native presence
features are Busy Lamp Field (BLF) Speed Dial and the Call History List with presence
enabled. The lesson also reviews presence policies, which are like class of service (CoS), to
control who can see whom. Presence policies are the major cause of problems when you
implement native Cisco Unified Communications Manager presence. The lesson discusses the
issues of native Cisco Unified Communications Manager presence and outlines how to solve
them.
Objectives
Upon completing this lesson, you will be able to explain thc common issues related to native
Cisco Unified Communications Manager presence and identify the most likely causes of these
issues. This ability includes being able to meet these objectives:
Review native Cisco Unified Communications Manager presence operation and
configuration and describe the most common issues that arc experienced
Explain the most common causes of incorrect status indications on a line and describe how
to fix them
Explain thc most common causes of incorrect status indications on a trunk and describe
how to fix them
Explain the most common causes of incorrect status indications in phone directories and
describe how to fix them
Native Cisco Unified Communications Manager
Presence General Issues
This topic reviews the native Cisco Unified Communications Manager presence operation and
configuration and describes the most common issues that are experienced.
v U =. Miiiiu-J C<
Manacier Preser
HOUl h .i*t* h
Natively supported by Cisco Unified Communications
Manager
Allows an interested party (a watcher) to monitor the real
time status of a directory number (a presence entity).
* Administrator subscribes a watcher to receive status
information of the presence entity.
A watcher can show the status of a presence entity
Presence-enabled speed dials
Presence-enabled lists (call and directory lists)
" Three possible states of a watched directory number.
Entity is unregistered
Entity is registeredon-hook
Entity is registeredoff-book
Cisco Unified Communications Manager presence is natively supported by Cisco Unified
Communications Manager. No extra products or servers are required. It allows an interested
party, thc so-called watcher or subscriber, to monitor the real-time status of a directory number,
a so-called presence entity, or subscribee.
A watcher subscribes to the status information of one or more presence entities. The status
information of a presence entity can be viewed by using presence-enabled speed dials or
presence-enabled lists call lists such as placed, received, or missed calls and public directory
lists.
The status can be one of the following;
Unknown. This status is shown when the watched device is unregistered.
On-hook.
Off-hook.
Native Cisco Unified Communications Manager Presence
Configuration Review
This section reviews the BLF speed dial that is used to watch the status of another line on an
LED phone button.
Native Cisco Unified Communications
Manager Presence Configuration RevU
Cisco Unified Communications Manager supports the ability
for a speed dial to have presence capabilities via a BLF
speed dial:
BLF speed dials work as a speed dial and presence
indicator.
- Only the system administrator can configure a BLF
speed dial.
The administrator must configure the BLF speed dial with a
target directory number or a SIP trunk destination:
- BLF SIP line-side endpoints can also be configured with
a SIP URI for the BLF speed dial.
- SCCP line-side endpoints cannot be configured with a
SIP URI for BLF speed dial.
The BLF speed dial indication is a line-level indication and
not a device-level indication.
Cisco Unified Communications Manager supports the ability for a speed dial to have presence
capabilities via a BLF speed dial. BLF speed dials work as both a speed dial and a presence
indicator. Only the system administrator can configure a BLF speed diala system user is not
allowed to configure or modify a BLF speed dial.
The administrator must configure the BLF speed dial with a target directory number that is
resolvable to a directory number within the Cisco Unified Communications Manager cluster or
a SIP trunk destination. BLF SIP line-side endpoints can also be configured with a SIP Uniform
Resource Identifier (URI) for the BLF speed dial, but SCCP line-side endpoints cannot be
configured with a SIP URI for BLF speed dial. The BLF speed-dial indicator is a line-level
indicator and not a device-level indicator.
2010 Cisco Systems. Inc. Troubleshooting Cisco Unified Communications Manager Features and Application Issues 5-101
Cisco Unified Communications Manager Call History Presence
This section describes the Call History Presence feature that shows the status of a phone or line-
in the directories.
isco Unified uuminumcciiioii:*
anaqer Call Historv Presence
Cisco Unified Communications Manager supports presence
capabilities for calt history lists.
Call history list presence capabilities are controlled via the
BLF For Call Lists enterprise parameter within Cisco Unified
The BLF For Call Lists enterprise parameter impacts all
pages that use the phone Directories button, and it is set on a
global basis.
Call History Placed Calls
Missed
Received
Placed Calls
Personal Directory
Corporate Directory
CDPlaced Calls
I I To 914085551212
jjg&To 1005
H#To 1002
Cisco Unified Communications Manager supports presence capabilities for call history lists
(the Directories button on the phone). Call history list presence capabilities are controlled via
the BI.F For Call Lists enterprise parameter within Cisco Unified Communications Manager
Administration. Thc BLF For Call Lists enterprise parameter impacts all pages that use the
phone Directories button (Missed, Received, Placed Calls, Personal Directory, or Corporate
Directory), and it is set on a tjlobal basis.
Cisco IP Phones that Support Viewing Presence Status
Thetableshows which type of status information is supported onCiscoIPphone models.
Cisco IP Phones that Support V
Presence Status
794(125]. 796[125],
797[015]
SIP and SCCP
7914,7940,7960
SCCP
7914,7940,7960
SIP
Yes
Yes
No
KWj !-v *T"~--
sMs$&y"
Yes
No
No
As shoun in the table, Cisco Unified IP Phones 7914, 7940, and 7960 do not support presence
at all when running SIPwhenrunning SCCP they only support presence-enabledspeed dials
but no presence-enabled call and directory lists. Type B Cisco Unified IP Phone 794[125],
796[125], and 797[0I5] support both, presence-enabled call and directory lists and presence-
enabled speed dials regardless of the protocol (SIP or SCCP).
Note Cisco IP Communicator also supports both, presence-enabled speed dials and presence-
enabled call and directory lists.
Native Cisco Unified Communications Manager Presence
General Issues
Cisco Unified Communications Manager native presence is an interactive service that is
supposed to indicate presence status in real time.
?C U.;.o iLnic,
The following are general issues of native Cisco
Unified Communications Manager presence:
" Presence status not indicated properly at speed dial
* Presence status not coming over SIP trunk
* Historical presence status not indicated in call lists
The follow ing general issues can be experienced when implementing native Cisco Unitied
Communications Manager presence:
Presence status might not be indicated properly at the speed dial of a watcher.
Presence status might not be coming over a configured SIP trunk and distributed to a
watcher.
Historical presence status might not be indicated in call lists of thc watcher phone.
Troubleshooting Line Presence Indications
This topic explains the most common causes of incorrect status indications on aline and
describes how to fix them.
Presence Policies Using Subscribe CSS
Awatcher needs permission tomonitor thepresence entity;
SUBSCRIBE CSS
Presence group
. An IPphone ortrunk sendspresence requests:
SUBSCRIBE messages with theevent field sei toPresence
The SUBSCRIBE CSS is associated with the watcher and lists the partitions
that the watcher is allowed to see:
. This mechanism provides anadditional level ofgranularity.
- Can be assigned on adevice or user basis (Cisco Extension Mobility).
A can watch B
B cannot watch A
SUBSCRIBE CSS DEVICES
Partition DEVICES
SUBSCRIBE CSS NONE
Partition DEVICES
SCCP endpoints can request presence status of the indicated presence entity by sending SCCP
messages to Cisco Unified Communications Manager. There are no SCCP trunks.
If the presence entity resides within the Cisco Unified Communications Manager cluster, Cisco
Unified Communications Manager responds to the SCCP line-side presence request by sending
SCCP messages to the presence watcher that indicates the status of the presence entity.
If the watcher has permission to monitor the external presence entity-based on the
SUBSCRIBE calling search space (CSS) and presence group- -the SIP trunk will forward the
presence request to the external presence entity, await the presence response from the external
presence entity, and return the current presence status to the watcher.
Csco Unified Communications Manager provides thc capability to set policy for users who
request presence status. This policy can be set by configuring aCSS specifically to route SIP
SUBSCRIBE messages for presence status and by configuring presence groups with which
users can be associated that specify rules for viewing the presence status of users that are
associated with another group.
The first aspect of presence policies for Cisco Unified ^^^^1^^^
SUBSCRIBE CSS Cisco Unified Communications Manager uses the SUBSCKIBh L.^ to
determine how to route presence requests (SUBSCRIBE messages with the Event Ifield set to
Presence) that come from the watcher, which can be aphone or atrunk. The SUBSCRIBE CSS
is associated with the watcher and lists the partitions that the watcher is al owed to seeJh.s
mechanism provides an additional level of granularity for the presence SUBSCRIBE requests
to be routed independently from the normal call-processing CSS.
S2010 Cisco Systems, Inc.
Troubleshooting Cisco Unified Communications Manager Features and Application Issues
5-105
5-106
The SUBSCRIBE CSS can be assigned on adevice basis or on auser basis. The user settmg
applies for ongmatmg subscriptions when the user is logged into the device via Ci oUn ed
Communications Extension Mobditv or when the user ,s administratively ass.gncZhe
dies tlo, ^c T T ^'^B1F SP^ d'al Md " hi^ '"' Pe status
S BSCRIBF f^ Mi*W***& ,s rejected as "userunknown." When avalid
SESCRIBE CSS sspecified, the tndtcators work and the SUBSCRIBE messages are
accepted and routed properly. It is strongly recommended that you no. leave any CSS defined
V' h ^ 3CSS SCt t0 "*ne> C3n introducc P*<* *us or dahng plan
behavior that is difficult to predict. p
Troubleshooting Line Presence Indications with Subscribe
CSS
The figure illustrates a presence policyscenario that is basedon partitions and SUBSCRIBE
CSSs.
Troubleshooting Line Presence
indications with Subscribe CSS
Make sure that:
SUBSCRIBE CSS includes partition of the presence entity.
SUBSCRIBE CSS is assigned to the watcher phone.
Route Pattern: 8.1003
Partition P-3
Permitted
Permitted
CSSs
C-1 P-1 P-?
C-2 P-1 P-? P-3
C-3 P-1
Dented
Denied
Permitted
The configuration consists of threeCSSs: C-1 containspartitions P-I and P-2. C-2contains
partitions P-1, P-2, and P-3. C-3 contains partition P-I only.
Phonel has partition P-1 applied to its line, which is configured with DN 1001. CSS C-1 is
assigned to Phonel.
Phonc2 has partition P-2 applied to its line, which is configured with DN 1002. CSS C-2 is
assigned to Phone2.
ASIPphone withnumber 1003 canbereached through a SIPtrunk. Thecorresponding route
pattern 8.1003 is in partition P-3. CSS C-3 is assigned to the SIP trunk.
The effective permissions for presence subscriptions are as follows: Phone 1 is allowed to watch
the status of 1002but not that of 1003. Phone2 is allowed to watch both other phones. Phone 3
is allowed to subscribe to presence infonnation of 1001 but not of 1002.
Note The CSS in the figure refers to the standard CSS that is used for the implementation of
calling pnvileges. The CSSs are not relevant for the discussion of presence subscription
permissions, but because they also depend on the configured partitions, they are added
here to illustrate that they have to be considered in the overall configuration.
Partitions and SUBSCRIBE CSSs apply to both presence-enabled speed dials and
presence-enabled call lists.
While troubleshooting line presence indications when using SUBSCRIBE CSS, make sure that
the SUBSCRIBECSS of the watcher phone includes the partition that is assigned to the line of
the presence entity; otherwise, the access to its presence status is denied.
Presence Policies Using Presence Groups
The second option to implement the presence policy in Cisco Unified Communications
Manager is presence groups.
Presence Policies u^my t %eseuc&
Groups-
Apresence group controls the destinations that a watcher can monitor
> Presence groups can be assigned to devices, directory numbers, and users:
By default, all users are assigned to the Standard Presence Group.
When using multiple presence groups, consider this:
Set Ihe Inter-Presence Group Subscribe Policy service parameter
If one group has a relationship to another group via the Use System
Default selling this service parameter value will take effect.
If the Inter-Presence Group Subscribe Policy service parameter is set lo
Disallowed. Cisco Unified Communications Manager will block the
request even if ihe SUBSCRIBE CSS allows it
The Inter-Presence Group Subscribe Policy service parameter applies only
for presence status wiih call history lists and is nol used for BLF speed dials.
M can watch
E cannot watch M
BLF is still working
Presence Group Manage Presence Group Employee (E)
Devices, directory numbers, and users can be assigned to a presence group, and by default, all
users are assigned to the same Standard Presence Group. A presence group controls thc
destinations that a watcher can monitor, based on the association of a user with a defined
presence group (for example, employees watching managers is disallowed, but managers
watching employees is allowed). The presence group user setting applies tor originating
subscriptions when the user is logged in to the device via Cisco Extension Mobility or when the
user is administratively assigned to the device.
When multiple presence groups are defined, the Inter-Presence Group Subscribe Policy service
parameter is used. If one group has a relationship to another group via the Use System Default
setting rather than being allowed or disallowed, this service parameter value will take effect. If
the Inter-Presence Group Subscribe Policy service parameter is set to Disallowed, Cisco
Unified Communications Manager will block the request even if the SUBSCRIBE CSS allows
it. The Ititer-Presence Group Subscribe Policy service parameter applies only for presence
status with call history lists and is not used for BLF speed dials.
Presence groups can list all associated directory numbers, users, and devices via thc
dependency records. Dependency records allow the administrator to find specific information
about group-level settings. Use caution when enabling the Dependency Record enterprise
parameter because it can lead to high CPU usage.
When implementing presence policies, watchers and presence entities are put into presence
groups. Subscriptions can be allowed at an intergroup level.
Cisco Unified IP phones arc configured with two or more presence groups: one is applied lo thc
device (in the role as a watcher), and each line can be configured with a presence group in its
role as a presence entity.
On SIP trunks, only one presence group is configured, which is used in both roles: as a watcher
and as a presence entity. A presence group cannot be assigned to a route pattem.
Troubleshooting Cisco Unified Communications (TVOICE) vB.O 2010 Cisco Systems, Inc
Like SUBSCRIBE CSSs, presence groups can also beassigned toendusers. End users use
presence groups when they use Cisco Extension Mobility tolog into thephone or when the
users are associated with a device.
Note Presence groups only apply topresence-enabled call liststhey donot apply topresence-
enabled speed dials.
2010Cisco Systems, Inc Troubleshooting CiscoUnified Communications Manager FeaturesandApplication Issues 5-109
Troubleshooting Line Presence Indications with Presence
Groups
The figure illustrates a presence policyscenario that is basedon presence groups.
Troubleshooting i t\u Presence
Indications with Hênce Grouj:
Make sure that:
* Presence group members are assigned properly
* Permissions between groups are configured
Phonel j*
Presence Groups
G-2 to G-3 permitted
G-3loG-1 permitterl
Re&t denied
1003
.1 r
Permissions
From 1001
From 1002
From 1003
Permitted Permttled
Permitted Denied
Permitted Denied
The configuration uses three presence groups: G-l, G-2. and G-3. Inter-presencegroup
subscriptions are permitted from G-2 to G-3 and from G-3to G-l. All other inter-presence
group subscriptions are denied.
Phonel has presence group G-l applied to its line, which is configured with DN 1001. Presence
group G-2 is assigned to Phonel.
Phone2 has presence groupG-2 appliedto its line, whichis configured withDN 1002. Presence
group G-2 is also assigned to Phonc2.
A SIPphonewithnumber 1003 canbe reachedthrough a SIP trunk. Presence groupG-3 is
assigned to the SIP trunk.
Thc effective permissions for presence subscriptions are as follows: Phonel is allowed to watch
the status of 1002 and 1003. Phone2 is allowed to watch 1003 but not 1001. Phone 3 is allowed
to subscribe to presence information of 1001 but not of 1002.
While troubleshooting line presence indications when using presence groups, make sure that all
entities that arc supposed to watch each other's status are placed into correct presence groups.
If multiple presence groups arc used, make sure that correct inter-grouppermissions arc
configured.
imv
Troubleshooting Trunk Presence Indications
This topic explains the most common causes ofincorrect status indications ona trunk and
describes how to fix them.
Verify Presence Is Enabled on SI
System > SecurityProfile > SIP TrunkSecurity Profile
If presence indications arenottransported over a SIP trunk, most probably thetrunk is not
configured properly.
Ifpresence subscriptions should beavailable over a SIP trunk, presence needs tobeenabled on
the SIP trunk. Presence is not enableddirectlyat the SIPtrunkbut via a SIPTrunkSecurity
Profile. Therefore, configure a SIPtrunk security profile from System > Security Profile > SIP
Trunk Security Profile, where theAccept Presence Subscriptions and the Accept Unsolicited
Notification checkboxes are activated to givethe trunkpresencecapabilities. Thenapplythe
SIP trunksecurityprofileto the SIPtrunk, as shownin the figure.
2010 Cisco Systems, tnc
Troubleshooting CiscoUnified Communications ManagerFeatures and Application Issues 5-111
Verify Presence Policies Are Set on SIP Trunks
"fhis topic describes how toverify the presence policies that are configured on a SIP trunk.
jk tnt Sir -.e.ico Mi K*
Uunns
Device > Trunk
A1"^ - > i|i iiV
Make sure
SUBSCRIBE CSS,
if used, is applied
to SIP trunk.
"- = : * -= .-<-- u *...= ...
.-M,^^-..-
Device > Trunk
Make sure a
presence group, if
used, is assigned to
SIP trunk
Same presence
group is used in
watcher and
J") presence entity role.
Like phones, the tmnk implements presence policies to distribute presence indications properly.
The figures show how toverify if either of the presence policy options isassociated with the
trunk.
The upper figure displays the trunkthat is configured with SUBSCRIBR CSSand thc bottom
figure displays the presence grouppolicy,
Cisco Unified Communications Manager can send out subscribe messages on a SIP trunk
(when watching a presence entity that is located on the other side ofthe trunk) and can receive
subscriptions on a SIPtrunk(whena subscriber that is located on the other side of the trunk
watches a local directory numberover thc SIP trunk).
The trunk, therefore, can act inboth the subscriber and presence entity role. On a SIP trunk,
only one presence group can be configured, and therefore, this single presence group applies to
sending subscriptions as well as receiving subscriptions.
5-112 TroubleshootingCisco Unified Communications (TVOICE] v8 0
" Troubleshooting Historical Presence Indications
This topic explains the most common causes of incorrect status indications in phone directories
mm and describes how to fix them.
Troubleshooting
Indication
System >Enterprise Parameters
}.=--=!<=) st --' (* ciip (q<x<w) I Verify Historical
+-; -rf-jaftf.*'" cs*iM<.HiM si oscp igiunwi J presence indication is
LtîtiTtiir*- csxsnttattetiio&crniumys^ enabled.
jbi.a I
"i- r' *:'l<
Make sure that the phone model supports historical presence.
When browsing through adirectory or call list and historical presence indications are enabled,
each call list entry should display asymbol indicating its status.
If this is expected to work, but the presence indications do not show up in the call lists the
appropriate enterprise parameter might no. have been enabled. Make sure that BLF For Call
Lists enterprise parameter is set to Enabled and not its default Disabled.
After changing the BLF For Call Lists enterprise parameter to "Enabled," all phones that
support presence-enabled call lists have to be reset for the change to become effective.
Note In Cisco Unified Communications Manager, configuration presence-enabled call lists are
referred to as BLFcall lists. ,
For the historical presence to work, the phone also has to support it. Asubset of phone models
supports the historical presence, as shown earlier mthis lesson.
Troubleshooting Cisco Unified Communications Manager Features and Application Issues
5-113
Summary
This topic summarises the key points thai were discussed in this lesson.
- Cisco Unified Communications Manager native presence
general issues include presence status that is not indicated
properly at speed dial or calflists.
When troubleshooting line presence indications, make sure
that the presence policies are properly configured and speed
dial has BLF capabilities.
Presence indications can be transported over a SIP trunk
on y It must have presence capabilities enabled and must be
set to comply topresence policies.
For historical presence indications to work it has to be
enabled in enterprise parameters and the phone model must
support it.
In .his lesson, you have (canted to explain the common issues that arc related to Cisco Unified
Commumcations Manager native presence and how to identify the most likely causes oHl/esc
References
For additional information, refer tothese resources:
^koSystem,, ^c.OscoUniJiedConwnuucations
Re/ease *0(1,. April 2009 and updated April 2010. '" M(k
hup: >uui e1ô.eomenL-SdocsxoiL-e_ip comm euem.admin-X 0 1ccmlejit
r>gd-M0j-um.html
5-114 Troubleshooting Cisco Unified Communicati
oris (TVOICEl v8.0
'2010 Cisco Systems. Ir
r
Module Summary
lodule Summary
Cisco Device Mobility common issues are related to IP
subnets and DHCP, CAC and codecs, media resources, call
routing, or call privileges.
The most common issues of Cisco Extension Mobility
represent service login and logout problems, phone button
problems, and problems with call routing and call privileges.
Cisco Unified Mobility has the issues separately related to
Mobile Connect and Mobile Voice Access, inability to switch a
call between devices, presentation of a wrong calling ID, or
various call-routing and redirection problems.
Cisco Unified Communications Manager native presence
might have issues with incorrect presence status indicated on
speed dials or call lists.
In this module, you have learned how to troubleshoot issues that are related to Cisco Unified
Communications Manager features and applications.
References
Cisco Systems, Inc. Cisco Unified Communications Manager FeaturesandServicesGuide,
Release 8.0(1), April 2009 and updated April 2010.
http: wttw.eisco.com'en.'US'docs/voice ip comm'eucm/admin/8 0 1'ccmfeal
fsgd-801-cm.html
CiscoSystems, Inc. Troubleshooting Guidefor Cisco Unified Communications Manager,
http: uwu.ciêo.com en'US.docs'voiee_tp_conim/cucm/trouble/8_0_l/trbl80I.html
>2010Cisco Systems, Inc.
Troubleshooting Cisco Unified Communications Manager Features and Application Issues 5-115
y..
5-116 Troubleshooting Cisco UnifiedCommunications (TVOICE) v8 0 2010 Cisco Systems, Inc
Module Self-Check
Use the questions here to review what you learned in this module. Thc correct answers and
.. Ql) Device mobility can be used in multisite environments with distributed call processing.
h The feature supports any SCCP or SIP device that can be configured in Cisco Unified
Communications Manager. (Source: Troubleshooting Device Mobility Issues)
* A) true
M B) false
Q2) Cisco IP Communicator runs on a laptop. The operating system manages the network
settings. Also, the laptop might have static IP addressing or incorrect gateway
information, and Cisco IP Communicator running on that laptop wilt be unable to reach
Cisco Unified Communications Manager. (Source: Troubleshooting Device Mobility
Issues)
A) true
B) false
Q3) The Device Mobility feature maps various configuration elements to apply the correct
site-specific parameters to a phone when it roams in another site. Which three of the
following statements have correct elements mappings? (Choose three.) (Source:
Troubleshooting Device Mobility Issues)
A) The physical location maps with the device configuration page.
B) The physical location and device mobility info map to one or more device
pools.
C) The device mobility group maps with an end user owning the roaming device.
D) The roaming-sensitive parameters are configured under the device to which the
device mobility group and physical location map.
E) The roaming-sensitive parameters are configured under the device pool to
which the device mobility info and physical location map.
F) The device pool references the new site-specific default router.
G) The device pool references the new site-specific media resources by using an
MRGL reference.
Q4) A phone with a deviee CSS that routes calls through the IIQ gateway roams to a remote
site. The remote site sets the Branch gateway through the CSS, which is also assigned
as a deviee mobility CSS to the roaming device pool. Which gateway will be used by
the phone when calling from the roaming site to the PSTN? (Source: Troubleshooting
Device Mobility Issues)
A) HQ gateway
B) Branch gateway
C) either HQ or Branch, depending on the device mobility group
Q5) Which two statements about troubleshooting typical devicemobility call privilege
problems are true'.' (Choose two.) (Source: Troubleshooting Device Mobility Issues)
A) In the line dtnicc CSS model, the line CSS always overrides the device CSS.
I'he line CSS settings take precedence over the device CSS settings.
B) if home and roaming sites use different device mobility groups, then the
Device Mobility CSS from the roaming device pool replaces thc device CSS.
C) If home and roaming sites use the same device mobility group, then the Device
Mobility CSS from the roaming deviee pool replaces the line CSS.
D) If home and roaming sites use the same device mobility group, then the Deviee
Mobility CSS from the roaming device pool replaces the device CSS.
E) Neither device CSS nor line CSS is replaced when thc device mobility group is
the same.
Q6) What are two types of configuration parameters that are dynamically configured when
yoti use Cisco Extension Mobility? (Choose two.) (Source: Troubleshooting Cisco
Extension Mobility Issues)
A) Call routing parameters (such as gateway or trunk) and calling privilege
parameters (such as device CSS).
B) Phone-specific phone configuration parameters, such as SRST reference,
media resources, AAR, and AAR CSS.
C) User-specific phone configuration parameters, such as user music on hold,
audio source, phone button templates, softkey templates, user locales, DND,
privacy settings, and phone service subscriptions.
D) Cisco Extension Mobility updates softkey buttons according to the Cisco
Unified IP phone model.
E) Cisco Extension Mobility updates all phone buttons not only the button types
as specified in thc phone button template, but also the complete configuration
of the phone buttons.
Q7) When a user attempts to log out from Cisco Extension Mobility and presses the
Sen ices button, the Cisco Extension Mobility service is unavailable. Which is the most
likely cause? (Source: Troubleshooting Cisco Extension Mobility Issues)
A) mismatch between user locales
Bj user device profile that is not subscribed to the Cisco Extension Mobility
C) phone that is not subscribed to the Cisco Extension Mobility
D) VLAN mismatch
Q8) Cisco Extension Mobility replaces the CSS that is configured at the
phone \\ ith the CSS that is configured in the device profile of the user who is logged in.
(Source: Troubleshooting Cisco Extension Mobility Issues)
Q9) Which two of the following are the most likely causes when the wrong gateway is used
for PSTN calls after a user has logged in to Cisco Extension Mobility? (Choose two.)
(Source: Troubleshooting Cisco Extension Mobility Issues)
A) CSS problem
B) Cisco Extension Mobility service failure
C) local route group missing or incorrectly configured at phone device or route list
D) local route group missing or incorrectly configured at device pool or route list
E) Cisco Unified Communications Manager dial plan poorly designed
Tracing CCD Pattern Blocking
This section shows how to identify that the CCD blocking feature has blocked a pattern.
Tracing CCD Pattem Blocking
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tion* 0,0,0,0.0"'**
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Pattern-51Z2](XX 0,0,0,0.0**'*
nlnfa, LaarnsdPatts blc
The Blocked Learned Pattern configuration window supports the CCD feature by allowing you
to purge and block learned patterns (for example, learned patterns that you no longer want to
use). But you can also block the learned patterns so that Cisco Unified Communications
Manager ignores the patterns when they arc advertised by remote call control entities.
This trace output shows that the pattern 5122XXX has been received from the SAF layer and
decoded at the CCD layer.
But the learned pattem is not entered to digit analysis and the cache because it is being blocked
by the Blocked Learned Pattern configuration. You can access this configuration page by
choosing Call Routing >Call Control Discovery > Blocked Learned Patterns in Cisco
Unified Communications Manager Administration.
You can view purged and blocked learned patterns in the Find and List Blocked Learned
Patterns window in Cisco Unified Communications Manager Administration. These purged or
blocked patterns do not display in Cisco Unified RTMT. If you delete a blocked pattem from
Cisco Unified Communications Manager Administration, Cisco Unified Communications
Manager can relcarn those patterns if they are still available in the SAF network and if the
maximum number of learned patterns has not been reached for the cluster.
For a pattern to be releamed by Cisco Unified Communications Manager, the entire record for
the blocked learned pattern must be deleted from Cisco Unified Communications Manager
Administration. That is, Cisco Unified Communications Manager does not releam a pattem if
you delete only part of the blocked learned pattem configuration (for example, if you delete
only the Remote Call Control Entity or Remote IP configuration for thc record).
2010 Cisco Systems, Inc SAF and CCD Issues 4-53
Dialing to Learned Patterns Fails
This section describes howto troubleshoot when patterns arc learnedbut they are unreachable.
4-54
uirthny to tearoed Patterns rails
The problem:
* Patterns are learned but unreachable.
* If UnReachable shows in the RTMTlearned patterns report, the
reason is lost connectivity with the SAF forwarder or remote CCD
client PSTN failover might still work.
" Learned patterns are placed to a CCD partition that might not be
mentioned in the calling phone CSS.
Incorrect Learned Pattern Prefix applied at CCD requesting
service.
* Network connectivity issues exist.
if patients are learned from the SAF network but dialing to them fails, consider these causes:
If the Cisco UnifiedRTMT Learned Patterns report shows UnReachable for the pattern, the
reason is lost connectivity to either thc SAF forwarder or the remote CCD client that has
advertised the pattern (VoIP unavailable fortius pattern). If PSTN failover is enabled for
this pattern, the calling should proceed through PSTNduring the periods when thc SAF
forwarder or the remote CCD client are unreachable on-net. If PSTN failover does not
work, but it is configured, then look a few pages ahead in this lesson for additional
troubleshooting recommendations.
If the Cisco UnifiedRTMT Learned Patterns report shows Reachable for the pattern, then
this issue needs troubleshooting. Learned patterns are placed to a CCD partition when
entered to digit analysis. If the panem is learned, added to digit analysis, but not reachable,
then this CCD partition might not be mentioned in the calling phone CSS. Troubleshoot
this issue as a dial plan problem.
Another cause for an unreachable pattern could be an incorrect Learned Pattern Prefix
applied at the CCD requesting service. The learned pattern prefix is applied to the hosted
directory number pattem before the CCD requesting service registers with digit analysis.
When calling to thc pattern, the prefix must be dialed with the pattern for a successful call
setup. The learned pattern prefix does not show in the Cisco Unified RTMT learned pattern
report.
Caution Updating the Learned Pattem Prefixfield or Route Partitionfieldcan impact system
performance because the digit-analysis master routingtable automatically is updated when
these fields are changed. To avoid system performance issues, Cisco recommends that you
update these fields during off-peak hours.
Network connectivity issues, especially to remote CCD clients and PSTN gateways, might
also make the call inn unsuccessful.
Troubleshooting Cisco Unified Communications (TVOICE) v8.0 )2010 Cisco Systems Inc
QIO) Mobile Connect provides features for outgoing calls. When Mobile Connect isenabled,
a userwho is outside theenterprise canmake callsas if theuser wasdirectly connected
toCisco Unified Communications Manager. (Source: Troubleshooting Cisco Unified
Mobility Issues)
A) true
B) false
Qi 1) Whichtwo statements describe the dustingfeature of Enterprise FeatureAccess?
(Choose two.) {Source: TroubleshootingCisco Unified Mobility Issues)
A) The feature can movea singlecall, conference, or sessioncollaboration among
mobile phone, PC, anddeskphone bydialinga determined DTMF
combination.
B) Enterprise Feature Access, including thedusting feature, is enabled bydefault.
C) When themobile phone useris on anactive call, thcusercanusethedusting
feature to parkthe call by transferring the parkedpartyto the parkcode that the
user entered.
D) The original conversation can be continued until thc call is answered at thc new
terminating device.
E) The original conversation is terminated as soonas the user dials a determined
DTMF combination for the dusting feature. The call can be continued at the
new terminating device.
Q12) Theuser finds thattheMobility button doesnot existon theoffice phone. Which is the
most probable cause?(Source: Troubleshooting CiscoUnifiedMobility Issues)
A) Thc user who is enabled for Mobile Connect may not be property associated
with the office phone device.
B) The office phone device is not enabled for Mobile Connect.
C) The softkeytemplate withthe Mobility buttonis not attachedto the office
phone device.
D) The Mobile Connect service has not been activated.
Q13) Outbound call routingfor Mobile VoiceAccess calls alwaysuses a concatenation of
the remote destination profile line CSS and device-level line CSS. Therefore, it is
important thatyouconfigure theseCSSs appropriately to provide access to anyroute
patterns that arenecessary foroff-net or PSTN access. (Source: Troubleshooting Cisco
Unified Mobility Issues)
A) true
B) false
Q14) Which three stams indications are shown when using Cisco Unified Communications
Manager nativepresence? (Choose three.)(Source: Troubleshooting CiscoUnified
Communications Manager Native Presence Issues)
A)
Unregistered
B) Unknown
O Idle
D) On-hook
E) Standby
F) Busy
G) Off-hook
2010 Cisco Systems, Inc TroubleshootingCisco Unified Communicalions Manager Features and Application Issues 5-119
Q15) Ifthc watcher has permission tomonitor the external presence entity, based onthe
and __, theSIPtrunk will
forward the presence request tothc external presence entity, await the presence
response from the external presence entity, and return the current presence status to thc
watcher. (Source: Troubleshooting Cisco Unitied Communications Manager Native
Presence Issues)
Q16) Which two check boxes have tobechecked on the SIP Trunk Security Profile
configuration page forpresence subscriptions tobe available over a SIP trunk? (Choose
two.) (Source: Troubleshooting Cisco Unified Communications Manager Native
Presence Issues)
A) Enable Application Level Authorization
B) Accept Presence Subscriptions
C) Accept Out-of-Dialog REFER
D) Accept Unsolicited Notification
E) Accept Replaces Header
Q17) For browsing through adirectory orcall list, each call list entry can display a symbol
indicating its status. Forthis towork, the BLF for Call Lists enterprise parameter must
bechanged to Enabled from itsdefault of Disabled. (Source: Troubleshooting Cisco
Unified Communications Manager Native Presence Issues)
A ] true
B) false
5-120 Troubleshoolmg Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems Inc
Ql) B
q:)
A
03) B. -. G
Q4) C
Q5)
A.D
Q6I C. E
07)
B
QSi line
Q9) A.D
010)
B
Qll)
A.U
Q12| C
Q13) B
014)
B. D. G
Q15)
SUBSCRIBE CSS and presence group
016) B.D
017) A
2010 Cisco Systems. Inc Troubleshooting Cisco Unified Communications Manager Features and Application Issues 5-121
5-122 Troubleshooting Cisco Unified Communications (TVOICE) vB 0
Module 6
Voice Quality and Media
Resources Issues
Overview
Today Cisco Unified Communications system functionality requires the use of media
resources such as annunciator, transcoding, conferencing, music on hold (MOH), and media
termination. These media resources can be either software- or hardware-based. Both methods
have their own possible unique issues. These tools also increase the complex.ty of the Cisco
Unified Communications system and make it more challenging to troubleshoot. This module
will teach you how to troubleshoot both implementations effectively.
Isolating the source of voice quality issues is one of the most difficult problems that you can
face when troubleshooting aCisco Unified Communications system. Voice quality can be
subjective and data networks are not sensitive to the same impairments or limitations to which
voice over data networks are sensitive. This module also describes how to troubleshoot
situations when quality ofthe voice transmission declines.
Module Objectives
Upon completing this module, you will be able to troubleshoot voice quality issues and issues
that are related to media resources. This ability includes being able to meet these objectives:
. Explain the common lSsucs that are related to MOH and identify the most likely causes of
these issues
Explain the common issues that arc related to MTP and identify the most likely causes of
these issues
Explain the common issues that are related to conferences and identify the most likely
causes of these issues
Explain thc common issues that are related to transcoders and identify thc most likely
Explain the common issues that are related to RSVP agents and identify the most likely
Explain common voice quality issues and identify the most likely causes of these issues
6-2 Troubleshooting Cisco Unified Communications (TVOICEl v8.0
Lesson 1
Troubleshooting MOH Issues
Overview
Themusic onhold(MOH) feature letsyouplaceon-andoff-net userson holdwith music that
isstreamed from a streaming source. This feature includes the end-user hold and the network
hold, which includes transfer hold, conference hold, and park hold.
This lessonidentifies the most common issuesof MOHand explains howto troubleshoot the
issues.
Objectives
Upon completing this lesson, you will beable toexplain the common issues that are related to
MOH and identify the most likely causes of these issues. This ability includes being able to
Review situations whenMOHis usedto play music, reviewMOHimplementation options
and characteristics, and outlinegeneral issuesthat can be experienced with MOH
Describe how to monitor and troubleshoot MOH performance
Describe theMOH registration andnonresponsive software issues andexplain how to
troubleshoot them
Describe howto runevolume of playedmusicor tones to comfort
Describe the major causes ofplaying nomusic or playing TOH instead of MOH andhow to
troubleshoot them
Describe themajor issues that are related to multicast MOII andhowto troubleshoot them
Describe the major issues that arerelated to multicast MOH from a branch router flash and
outline how to troubleshoot them
MOH Review
6-4
This topic first reviews situations when MOH is used to play music. Then itdescribes MOH
implementation options and characteristics and outlines general issues that can be experienced
with MOH.
:i^H-VBJ.M(=SW.. (
The MOHfeature provides music to callers when their call is held
by features such as user hold, call transfer, Call Park, or
conference.
MOH provided by:
Cisco IP Voice Media Streaming Application service (software)
Cisco IOS router, when playing musicfrom branch router flash
(hardware)
-%/
For callers to hear MOH, Cisco Unified Communications Manager must be configured to
support the MOH feature. The MOH feature has two main requirements:
A MOH server to provide the MOH audio stream sources
ACisco Unified Communications Manager that is configured tousethe MOH streams that
are provided by thc MOII server when a call isplaced onhold
Theintegrated MOH feature allows users toplace on-andoff-net users on hold with music that
is streamed from a streaming source. This source makes music available toany on- oroff-net
deviee that isplaced onhold. On-net devices include station devices and applications that are
placed onhold, consult hold, or park hold byan interactive voice response (IVR) orcall
distributor. Off-net users include those connected through Media Gateway Control Protocol
(MGCP). Session Initiation Protocol (SIP), and H.323 gateways. Thc MOH feature isalso
available for plain old telephone service(POTS) phones that are connected to theCisco IP
network through Foreign Exchange Station (EXS) ports. Theintegrated MOII feature includes
media server, database administration, call control. Media Resource Manager (MRM), and
media control functional areas. TheMOH server provides themusic resources andstreams.
The Cisco Unified Communications Manager integrated MOH server supports multicast and
unicast forMOII streaming. The advantage ofusing multicast for MOII streaming over unicasl
is tosave bandwidth and toreduce load onthe MOH server. Saving bandwidth should notbea
major issue for campus LAN environments, but reducing load onthe MOH server by reducing
the number of media streams is advantageous, especially when the MOH server is coiocated on
thcsame serverwiththe CiscoCallManager service.
TroubleshootingCisco Unified Communicalions (TVOICE) v8.0
>H|lt
Unicast MOH
Unicast MOH consists of streams that aresentdirectly from theMOH server to theendpoint
that is requesting a MOHaudio stream.
Unicast MOH
Unicast MOH characteristics:
Streamsent directly from MOH server to requesting endpoint.
* Point-to-point, one-way audio stream.
Separate audio stream for each connection.
High bandwidth usage, increasing with number of phones.
UnicastMOH .*
Unicasl MOH
Aunicast MOH stream is a point-to-point, one-way audio Real-Time Transport Protocol (RTP)
stream between the server and the endpoint device. Unicast MOHuses a separate source stream
for each user or connection. Asmore endpoint devices goonhold viaa user or network event,
the numberof MOHstreams increases. Thus, if 20 devicesare on hold, then 20 streamsof RTP
traffic are generated over the network between the server and these endpoint devices. These
additional MOH streams can potentially have a negative effect onnetwork throughput and
bandwidth. However, unicast MOH can be extremelyuseful in thosenetworksin which
multicast is not enabled or in which devices are incapable of multicast.
Multicast MOH
6-6
Multicast MOII consists of streams that arc sent from the MOII server to amulticast group IP
address that endpoints that are requesting aMOI Iaudio stream can join as needed.
Multicast MOH characteristics:
* Streams sent from MOH server toa multicast group IPaddress.
* Endpoints request a MOH audiostreamandjoin as needed.
* Pomt-io-multipoint, one-way audio stream, shared byall users.
Conserves system resources and bandwidth.
- Network and endpoints must support multicast
f
Join Multicast Group 4
Amulticast MOH stream isa point-to-multipoint, one-way audio RTP stream between the
MOII server and the multicast group IP address. Multicast MOII conserves system resources
and bandwidth because it enables multiple users to use the same audio source stream to provide
MOII, Thus, if20 devices are on hold, then, potentially, only asingle stream ofRTP traffic is
generated over the network per enabled codec. For this reason, multicast is an extremely
attractive technology for the deployment ofaservice such as MOII because it greatly reduces
ihe CPU impact on thc source device and greatly reduces the bandwidth consumption for
delivery over common paths. However, multicast MOII can be problematic in situations in
which anetwork is not enabled for multicast or where the endpoint devices are incapable of
managing multicast.
*' MOH Sources
MOH audio files aregenerated automatically byCisco Unified Communications Manager
when .wav audio files are uploaded to the MOH server.
MOH Sources
MOH source characteristics:
Codecs supported by MOH are G.711, G.729, and wideband.
- G 729 is developed and optimizedfor speech and can drastically
reduce quality when used for music.
MOH sources
One fixed source using a Cisco MOHUSB audio sound card.
50 audio file sources
Audio files can be uploaded and are then converted (once per
codec)
* Upload has to be performed fromCisco Unified Communications
Manager Adminstration pages at each MOHserver.
Audio 1 (G711 a-law)
Audio 1 (G 711 mu-law)
Audio 1 (G 729)
Audio ! {wideband)
Audio 2 (G 71! a-Law)
Audio 2 (G.711 mu-law)
Audio 2 fG 729)
Audio 2 (wideband)
When theadministrator imports anaudio source file, theCiscoUnified Communications
Manager Administration window interface processes the file and converts the file tothe proper
formats for useby theMOH server. Therecommended format for audio source files includes
thc following specifications:
16-bit pulse code modulation(PCM) .wav file
Stereo or mono
Sample rates of 48, 32, 16, or 8 kHz
If recordedor live audiois needed, MOHcanbe generated froma fixedsource. For this typeof
MOII, a soundcard is required. The fixedaudiosource is connected to the audioinput of the
local sound card.
This mechanism enablesthe use of radios, CDplayers, or any other compatible soundsource.
The stream from the fixed audio source is transcoded in real time to support the codec that is
configured through Cisco Unified Communications Manager Administration. Thefixed audio
source can be transcoded into G.711 (a-law or mu-law), G.729 A, and wideband. It is thc only
audio source that is transcoded in real time.
The Cisco MOH USB audio sound card (MOH-USB-AUDIO=) must be used for connecting a
fixed or live audio source to the MOH server.
Note When uploading MOH files, use the IP address (or name) of the server that provides the
MOH servicethis is the server where the Cisco IP Voice Media Streaming Application
service is activatedin the URLfor accessing Cisco Unified Communications Manager
Administration. If multiple MOH servers exist, you must repeat the upload at each individual
server.
MOH Audio Source and Media Resource Selection
This figure describes howthe MOH audiosource fileand MOH audioserverare selected in
CiscoUnified Communications Manager.
urfdft ~J&,
TheMOH stream that an endpoint receives is determined by a
combination of these:
- User hold audio source ofthedevice placing theendpoint onhold (holder)
specifies wtiich audio source to use (audio source number!
MRGLof held endpoint (holdee) is used lo determine Ihe MOHserver to
use
Audiosources can be configured inservice parameters, devices, and lines.
Server
MOHB
"%>&
Audio 1
Audio 2
Audio 3
Audio 4
Phone B
puts
Phone A
on hold
Phone B
User Hold " '
1. Priority MOH Server I
Server
MOH A
I
Audio 1
Audio 2
Audio 3
Audio 4
Listen to
UseMRGLA
Phone A
'User Hold Audio 4
1. Priority MOH Server A
The basicoperation of MOII in a CiscoUnified Communications environment consists of a
holder and aholdee. The holder is the endpoint user ornetwork application that is placing a call
on hold, and the holdee is theendpoint useror device that is placed on hold.
The MOI 1stream that anendpoint receives isdetermined by a combination of the user hold
MOH audiosource of thedevicethat is placingthe endpoint on hold(holder) and the
configured, prioritized list ofMOH resources (Media Resource Group List [MRGL]) ofthe
endpoint that is placed on hold (holdee). The user hold MOH audio source that is configured
for the holder determines the audio file that will be streamed when the holder puts acall on
hold. The prioritized list of MOH resources of the holdeeis relevant to determine the server
from which the holdee will receive the MOII stream.
In simplest terms, the configuration of the holder determines which audio file to play, and the
configuration ofthe holdee determines which resource orserver will play that file. In the figure,
ifphones Aand Bare on acall, and phone B(holder) places phone A(holdee) on hold, phone
Awill hear thc MOII audio source that isconfigured for phone B(Audio 2). However, phone A
will receive this MOH audio stream from thc resource orserver that isconfigured for phone A.
6-8 TroubleshootingCisco Unified Communications (TVOICE) v6
MOH Service Parameters
The figure lists the relevant service parameters for MOH that arc usually verified during MOH
troubleshooting.
IOH Service Parameters
Cisco IP Voice Media Streaming Application service:
- Supported MOH codecs (G.711, G729A, wideband)
- QoS for MOH (signaling and audio)
Packet size for G.711, G.729, and wideband (20 ms)
Cisco CallManager service:
- Suppress MOH toConferenceBridge (True)
Default Network Hold MOH Audio Source ID (1)
Default User Hold MOHAudio Source ID(1)
- Duplex Streaming Enabled (False)
The default parameters are shown in parentheses. Configure these service parameters only if
there is a need to use nondefault values.
Note Theseservice parameters can beaccessedfrom System >Service Parameters. Note that
some of them are Cisco IPVoiceMediaStreamingApplication service parameters and
others are Cisco CallManager service parameters, as shown.
MOH Operation
The figure shows howMOII operate-
When aphone is held either because itis put on hold, it is parked by auser, or because
another feature such astransfer orconference is invoked - Cisco Unified Communications
Manager checks if a MOH media resource isavailable. Ittries toallocate a MOI I media
resource by usingthe MRGL of thedevicethat is held, the holdee.
IfaMOII media resource is unavailable, tone on hold (TOH) is played. Ifan MOII server is
allocated, Cisco Unified Communications Manager searches for a common codec for the MOH
stream that is based on the codecs that are supported by the MOII server, the holdee, and the
applicable regions. Ifno common codec for the MOH stream can be identified, TOH is plaved.
II acommon codec is found. Cisco Unified Communications Manager checks ifthe MOH
stream isadmitted bylocations configuration. This applies only to unicast MOH. Multicast
MOII isalways permitted, irrespective ofthe location that is configured. Ifthe MOH stream is
not admitted. TOH is played.
Ifthe MOH stream is admitted, Cisco Unified Communications Manager determines which
audio source to use. The audio source number is determined based on the configuration ofthe
holder.
The next decision is whether to use unicast MOH or multicast MOH. Multicast MOH will be
used only if the selected audio source is enabled for multicast, theselected MOH server is
enabled for multicast MOH. and multicast is enabled at the Media Resource Group (MRG) of
the holdee. In all other situations, unicast MOI I is used.
Itmulticast MOH is used, the holdee joins the multicast group as signaled by Cisco Unified
Communications Manager. Ifthere isany problem in this area (group cannot be joined, MOH
RI Ppackets are not received- for example, because ofatoo low maximum hop configuration
at the MOH server) theholdee will only heardead air. NoTOH andnoMOH will beheard.
Troubleshooting CiscoUnified Communications (TVOICE) v8 0
Common Issues of MOH
The list shows the most common issues of MOH.
Common Issues of MOH
Most common issues of MOH;
MOH resource is unavailable:
A MOH resource is not registered.
- All MOH resources are currently in use.
- Media Resources misconfigured, MOHnot listed.
TOH is heard instead of MOH.
* Call is disconnected when placed on hold.
MOHaudio has poor qualifyor is almost inaudible.
Multicast MOH is expected but unicast MOHtakes place.
MOH audio is inefficiently transferred via WAN when local
MOH resources are available.
These are the most common issues of MOH:
MOII resource is unavailable. An MOH resource might not be registered, all MOH
resources might currently beinuse, or MOH could beexcluded from theMRG and, hence,
unavailable to the endpoint.
TOH is heard instead of MOII.
Call is disconnected when placed on hold.
MOH audiohas poor qualityor is almost inaudible.
Multicast MOH is expected, but unicast MOH takesplace instead.
MOH audio is inefficiently transferred viaWAN when thelocal MOH resource is
available. This couldbe due eitherto misconfiguration or MOHserverunavailability.
Voice Quality and Media Resources Issues
Typical Issue: Calls Disconnected When Placed on Hold
The figure shows a typical issue of MOH ina multiregion Cisco Unified Communications
Manager environment: Acall is disconnected a few seconds after it isplaced onhold.
6-12
h/pical iss
Problem: Call is disconnected a few secondsafter it is placed
on hold:
- This can occurwhen the voice codec fora given device, as defined by
its region, is not in the list of codecs supported by the MOH server.
Enable codec and restartthe CiscoIPVoice Media StreamApplication
service
MOH
ServiceParameters > CiscoIPVoice Media StreamingApp
Region: G.729
Only G.711 mu-law
is allowed by the service
Ihis issue can occur when the voice codec for a given device, as defined by its region, is not in
the list ofcodecs that are supported by the server that streams the MOII stream. For example, if
a particular device is set touse only the G.729 codec, but the MOH service is only configured
to stream G.7] I mu-law. then this particularproblem can occur.
To resolve this issue, enable G.729 in the Cisco IP Voice Media Stream App service and restart
theservice. Choose System > Service Parameters and chooseCisco IPVoice Media
Streaming App from the Seniee drop-down menu. Under Clusterwide Parameters, choose
G.729 codec for Supported MOH Codecs parameter. To choose multiple codecs for MOH,
press Ctrl and select multiple codec values. The figure shows two codecs that are selected for
MOM: G.71 I mu-law and G.729. Ifusing G.729 codec, be aware that this codec is optimized
for speech, andtheaudio fidelity will be marginal.
Navigate toCisco Unified Serviceability, choose Tools >Control Center-Feature Services.
and restart the Cisco IP Voice Media Stream App service for the codec change to apply.
MOH Performance
This topic describes how tomonitor andtroubleshoot MOH device performance.
\ Performance in Cisco Uni
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Maximum available MOH streams
al the resource
This figure shows how MOII resources can bemonitored byusing theCisco Unified Real-
Time Monitoring Tool (RTMT).
CiscoUnified RTMTperformance counters allowthe displayof the activity of MOII devices
thatare registered withCisco Unified Communications Manager. TheCiscoMOH Device
group of performance counters contains variables for both unicast and multicast MOH. In
addition to a real-time activity, the total numberof available resources canbe shown.
This figure shows the two MOHperformance variables:
MOHTotalUnicastResources shows that the MOH device being monitored (M0H_2) can
supportmaximally 250MOHstreams (server platform dependent).
MOHUnicastResourceActive shows the real-time activity of the MOH device that, at the
moment, streams music to a single listener.
Voice Quality and Media Resources Issues
MOH Device Flapping Issue
This section describes the problem when the MOII resource performance variable fiaps
between two values.
..=*>*> *n**in a-=.a *.,
MOH Device Rappirtc! Issue
Problem:
MOH resource active variable flaps between 1 and 2.
Flaps between 1 and 2-
Possible causes:
MOH file is uploaded to publisher only.
MOH source filename contains space character.
on'^^^V^^^^^^B
Byusing Cisco Unified RTMf MOH device performance monitoring, youcanobserve an
interesting issue. If the number of currently active MOII devices fiaps between I and 2when
they stream music toasingle device, check these factors. Make sure that you have uploaded the
MOH file to all the servers in the Cisco Unified Communications Manager cluster, not justthe
publisher. Also, make sure that the MOII source filename does notcontain a space. Both of
these situations can cause the MOH device activity flapping behavior.
5-14 Troubleshooting Cisco Unified Communications(TVOICE] v8.i
Troubleshooting MOH Registration and
Nonresponsive Software Issues
This topic describes the MOH registration issues nonresponsive software issues, and explains
how to troubleshoot them.
Troubleshooting MOH Registration
Problem:
MOH resource does not register withCisco Unified CommunicalionsManager.
Possible causes:
There are networkconnectivityissues, insufficient QoS, or carrier issues.
Cisco Unified CommunicationsManager daiabase is corrupted.
* Cisco IP Voice Media Streaming Application service is not running.
MOH resource is disabled or software error has occurred.
Cisco Unified
Communications
Manager
Thefigure shows that despite being properly provisioned, a MOH resource does not register
with Cisco Unified Communications Manager, and hence, it cannot be used. MOHsoftware
resources can be collocated with Cisco Unified Communications Manager software, or the
MOH server can work as a standalone server because of higher performance requirements.
MOH resources canbe placed in thesame sitewithCisco Unified Communication Manager
servers, or they can begeographically distributed toother sites tosupport a growing number of
endpoints in remote sites. The MOH server, when registered, isan integral part ofthe Cisco
Unified CommunicationManager cluster and a member of its database.
If the MOII serverdoes not registerwithCiscoUnifiedCommunication Manager, therecould
be several possible causes:
The MOHserver mightbe unableto reachthe CiscoUnifiedCommunication Manager
because of network connectivity issues, especially if a standalone MOH server is connected
via an IP WAN. Evenif an IP WANis operational, issueswithcongestion or latency might
exist, andthelevel of quality of service (QoS) might not manage MOH server registration
attempts.
The Cisco Unified Communications Manager database might be corrupted, and database
updates like MOH registration might not besuccessfully propagated across all members of
the cluster.
The Cisco IPVoiceMediaStreaming Application servicemight not be running properly. If
this service is down, all MOH server registrations are rejected.
Alternately, MOH resource canbedisabled or a software errorcouldoccur(process
nonresponsive).
Voice Quatily and Media Resources Issues 6-15
troubleshooting w-
Control Center - Feature Services
, *,-- I, ,- , .-1.^, it (,ulf =,
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1- ,1
Is service running?
Is MOH enabled?
Use png from server-to-server toconfirm network reachability
Make surethatihe carrier network is operational andprovides necessaryQoS
level
Run CiscoUnified Communications Manager daiabase integrity checks.
Takethesesteps to troubleshoot MOH registration issues:
1. Make sure that the Cisco IP Voice Media Streaming Application service isactivated
andrunning. Navigate to Cisco Unified Serviceability andchoose Tools >Control
Center- Feature Services. The service shouldreportStarted" state. Restart the
service if it is not running,
2. Ensure that the MOH server is enabled. In Cisco Unified Communications Manager
Administration, choose Media Resources >MOH Server and clickthc MOH server
link. On the MOI I Server configuration page, check the RunFlay, and ensure that it is
set to Yes.
3. If thc service is running andtheMOH server is enabled, butthe MOH server still
cannot register, verifynetwork connectivity between the MOH serverand Cisco
Unified Communications Manager. Onthccommand-line interface (CLI) of the MOH
server, use theutils network pingcommand andmake sure that theresponse is shown.
If Domain Name System (DNS) names areused instead of IP addresses inthe cluster,
veritythat the names are resolved properly, because thc DNS server addsan additional
point of failure.
4. Ifthe MOI I server is remote, make sure that the IP WAN isoperational and the latency
is satisfactory, Around-trip delayof less than40 ms between the MOH server and the
Cisco Unified Communications Manager is required.
5. Finally, verify that the Cisco Unified Communications Manager database is not broken
and all clusterservers maintain the samedatabase copy.
3-16 TroubleshootingCisco Unified Communicalions (TVOICEl vS.O
Tuning MOH Loudness
This topic describes how totune thevolume of played music or tones tocomfort.
Tuning MOH Loudness
Problem:
MOH music is too loud or silent
Corrective action:
. Modify volume using service parameter in Cisco IPVoice Media
Streaming Application service.
- Restart the service when parameter changed.
The change affects only new imported files. Reimport the .wav files
to MOH server.
Cisco IPVoice Media Streaming Application Service Advanced Parameters
F;
If the music that is played through MOH is tooloud or toosilent, youcantune it tocomfort at
CiscoUnifiedCommunications Manager. An advanced serviceparameter withinthe Cisco IP
MediaStreaming serviceis referred to as the DefaultMOHVolume Level. This parameter
specifies the decibel (volume) level adjustment thatisapplied toMOH audio source files when
new audio source files areimported. Changes tothisparameter only affect audio files that are
imported after the change occurred. Thedefault is-2 dB, and theparameter isconfigurable
between minimum -10 and maximum 10 dB.
When you change this parameter, restart theCisco IPMedia Streaming service toapply it. Then
upload the MOH source files again toadjust their volume according tothenew settings.
Youcan alsouse tools that can modify thedecibel level of .wavfiles to adjust musicvolume.
Several such tools arc available. After a .wav file is modified, upload it as a MOHsource again.
TOH Instead of MOH
This topic describes the majorcausesofplaying no music or TOHinstead of MOH and howto
troubleshoot thc associated problems.
6-18
Common causes:
1Mismatch betweenthe codec configuration used bythe MOH
serverand the region in which the endpoint is registered.
" Location-based connection admission control is being used,
and no bandwidth is available (applies to unicast only).
' Audio streams are unavailable. MOH server has a finite
number of unicast streams it can generate.
>MOH server not operational.
Media resources misconfigured.
Several reasons could cause a user to hear TOII instead of MOH:
Amismatch can occur between the codec configuration that is used bythc MOH server and
theregion in which the endpoint is registered. For example, theMOH server advertises a
capability ofonly G.71 1mu-law. If anendpoint is inanother region that enforces the 0.729
codec, the Cisco Unified Communications Manager chooses TOHbecause a streamcannot
be played to this endpoint.
If location-based connection admission control is being used andthere is nobandwidth
available, then Cisco Unified Communications Manaucr instructs the phone togenerate
TOII locally.
TOH can be played if there are no available audio streams, because thc MOM server has a
finite numberof unicast streams that it can generate. If this limit is exceeded, it causes
Cisco Unified Communications Manager to play TOH. Check MOH performance inCisco
Unitied RTMT to see if streams are available.
Thc MOH server might not be registered, or thc Cisco IP Voice Media Streaming
Application service is not responding.
IfMROs are used, ensure that the MOH server is part ofthc MRG and that the group
belongs to an MRGL. Also check thatyourCisco IPphone references thecorrect MRGL.
Troubleshooting Cisco Unified Communications(TVOICEl v8 0
w Verify MOH Configuration
Thissection describes how to verify MOH configuration as a part of thetroubleshooting
procedure.
Verify MOH Configuration
Pnonty I
Device > Phone > Line
Will' T'*
Jw *vna MOi Autfro Spjn
1-1..
i
1- -i
| S'jnds=d Pno>(Kn=p
i
11-S*mD=>AudnSau' c
i=rfT-ofh Held Men Xud* Soured 11-k=npb=.=oV?S m -
I***.* .1
Device > Phone
H*4uHtH>m
U-.-HolJPWt
iOH==.j*o
BW..=|I
A
1 >S*mtJH F*.k=#eSs>v-<.* i
\
Device > Device Sellings > Common Device Configuration
ui*r Ka>a tK-H ki)Aa Sb&t*
Hfvt. i-tpld *QW iW*o Squtu
m*ibB -
*tc<
[ E'OfiQard u*.fr J
11 -S i=nr>l t Audiosomsa
d
11 -StmelvAudioSokfit
li
| <Hi J.
Service Parameters > Cisco CallManager
T*
aii...S;;
b=^
/
MOII audio settings cause theselection of anaudio source file to bestreamed to a holdee. The
audio settings of thc holder phone dictate which source file will beplayed totheholdee.
CiscoUnifiedCommunications Manageruses the User andNetworkHold MOHAudiosettings
toplay the MOH audio source totheusers. You canconfigure the MOH audio source at
various places inCisco Unified Communications Manager. These options are inthe order of
priority:
Line settings
Phone settings
Common Device Configuration
If all of these settings have a value of None for MOH audio source fields, then Cisco Unified
Communications Managertakesthe audiosourcethat is configured in the CiscoCallManager
service parameters page. Choose System > ServiceParameters andthen selecttheserver and
CiscoCallManager service. Make sure that the MOII audiosourcefieldsare configured with
valid values.
Voice Quality and Media Resources issues
MOH^mrgl
L-,,<! 4 ktiurfo Grb=a Wprm*
. ..
1 .
. ..It.
nJH_;i*^M, |
i
Device a Phone
Syslem> Device Pool
-Da*n==-lieni=tl=#=- >i<n,n4i.-
"* """-* [f'l.îi
i
"-' ' |t-F*^T
,-
"'"*"! -"* "' 1"-""='!
.
'-'-" 1 ---
--
..-..- .., .1. j w. ,
-
MOH 2
The MRGI. of thc endpoint beingheld(holdee) determines the MOII server that will stream
audio. There are two options for associating MRGL with the phone. MRGI. can be configured
at either the phone configuration page orthough the device pool. Ifboth are configured, the
phone-configured MRGLtakes precedence.
Make sure that the phone isassociated with the correct MRGL and thatthe MRGL points tothe
MRG that has thc MOHserver configured.
6-20 Troubleshooting Cisco Unified Communications (TVOICE)v8 0
* TOH to PSTN Caller
This figure shows an issue of a phone playing a TOH instead of a MOH to a public switched
telephone network (PSTN) holdee.
Problem:
- When putting a PSTN caller on hold. TOHis played to the caller
instead of MOH.
MOH
MRGL:
<noMOH>
Region: G.7JS
Gateway
TOH
PSTN
Possible causes:
* MOH server issues
MRGLmisconfiguration or holder without MOHaudio source
Codec mismatch, gateway in region not matching MOHcodec
* Gateway not enabled for MOH
When a PSTN caller is put on hold by an internal Cisco IPphone, TOH is played to the PSTN
caller (holdee) instead of the expected MOH.
Several reasons could cause this behavior:
Thc MOH server might not be operational, it might be overused, or service might be
running improperly.
The MRGL might be misconfiguredat the Cisco Unified Communications Manager. Either
the assigned MRGLdocs not contain any MOHserver, or there is no MRGL configured on
the gateway configuration page. Another cause could be related to the holder. The internal
Cisco IP phone might not provide any MOH audio source reference.
Acodecmismatch mightoccur whenthe gatewayis placedin the regionthat enforcesa
codec that does not match any of the MOHcodecs.
The Cisco IOS gateway might not be enabled for MOH. The ccm-manager music-on-hold
command must be configuredto enable MOH functionality. This is a required command
for MGCP Cisco IOS gateways. This command allows thc gateways to receive multicast
MOHsuccessfully, as instructedby the Cisco Unified Communications Manager.
Optionally, you might also need to use the ccm-manager music-on-hold bind interface
command to bind the multicast MOH feature to an interface type.
)2010Cisco Systems. Inc. Voice Quality and Media Resources Issues 6-21
Troubleshooting Multicast MOH
This topic describes the major issues that are related lo multicast MOH and how to troubleshoot
them.
I
MOH 2 ip multicast-routing
?r.S
MOH Audio Source
interface serial 0/0
ip pim aparoe-densa-mode
- ,1
1
^HtJ^|.
^
1* T^.^ " MU L"fl-L*
<-"-- *=. ....
-<,- w.t.. . ...,.
=.. ..* ._.,. t
_E_
WAN
HQ-1 '<- -:-- r
MOH Server
MRG
Multicast MOHcan suppon considerably more endpoints on hold than unicast MOII. and
therefore, multicast MOH is often preferred over unicast MOI 1.
Multicast MOIL however, needs additional configuration andIPmulticast routing functionalitv
in IP WAN.
If multicast MOH is expected, but unicast MOH is played instead of multicast, there could be
several causes of such behavior:
Multicast MOH might not be enabled. Multicast MOH must be enabled on the MRG, the
audiosource, and the MOI I server. In addition, multicast parameters must be specified on
thc MOH server configuration page.
The MOH serverconfiguration page lets you specifya Max Hopsparameter that is usedas
a timc-to-livc parameter for outgoing multicast traffic. If Max Hops is set to less than or
equal to thc real number of router hops along the multicast path, dead silence instead of
music at thebranch is heard when a call is placed on hold. In theexample with tworouters.
Max Hops should be set to a minimum of 3 for thc multicast traffic to reach the branch IP
phone.
Cisco IOS routers might not be enabled for IPmulticast. Each router on thepath must be
globally enabled for IP multicast routing by thecommand ip multicast-routing, andthe
router interfaces must beenabled to participate in multicast routing by theip pini or ip pim
sparse-dense-mode interface configuration command.
Troubleshooting IP Multicast Routing
This figure showshowto troubleshoot problems of IP multicastroutingon a CiscoIOSrouter.
Troubleshooting IP Multicast Routine
BR-l#debug ip pin
Apr & 18:58-22,448= PIMIOI: I'u : iiKng Prait rar,4i4=== fur J"<-5 .1.1 i.
'.4: no entries
Apr E 18:56:22.448. PIK(O): EiiilUiii) =. a;:'; SSi'sass in; 2.S4 .1.1.1,
10.1.1.1/32 CQU ot 1
Apr 6 18:58:22.418 PIH10): Send =.i fia! '. 1=3 i'i.i =. Hll (SerJ-jIij/'i/'S. 11 i!
Apr 6 18:58:22.452 PIlt))i Received v2 Orft-Ack = n Serial0/1/0.111 iron
10.1.6.101
Apr 6 18:58=22.452 Group 23S.1.1.It
10.1.1.1/32
HQ-l#r3bug ip pim
Apr 6 19:34:11.549: PIM(O): 'us TiK?- '.. -L . I . =. I /'. 2 }>'>. 1.1)
Apr 6 19:34:31.549: PIM(O): Wl Jfrui:/;/; 121/:\ . ; . C. C tl, ilC-.i 1.1
Apr 6 19:34:31. 549: PIH10): Send v2 Oratt-Ack oa Serial0/1/0.121 to
10.1.6.102
Use the debug ip pim command to monitor IP multicast routing. Whena call is placedon hold
by using multicast MOH, the heldIP phoneattempts tojoin a multicast group, in this case
239.1.1.1. IP multicast routers need to build the multicast forwarding tree for this group, where
the MOH server is at its root (10.1.1.1).
Thefigure shows two debugging outputs. TheBR-1 router is theclosest router tothebranch IP
phone that is being held, andthisroutertriggers thebuilding of thetree. It formulates thegraft
message and sends it to the HQ-1 routerup the tree. HQ-1 receives the graft messages and
placesthe tree for the root 10.1.1.1 and group239.1.1.1 intothe forwarding state.
If the debug command does not show any IP multicast routing events, make sure that IP
multicast routing is enabled at both routers by using the show ip multicast command.
BR-l(tshow ip multicast
Multicast Routing: enabled
Multicast Multipath: disabled
Multicast Route limit: No limit
Multicast Fallback group mode: Dense
Number of multicast boundaries configured with filter-autorp option: 0
2010 Cisco Systems, Inc Voice Quality and Media Resources Issues 6-23
6-24
BR-l#show ip mroute
IP Multicast Routing Table
Flags: 0 - Dense, S - Sparse. B - Bldir Group, a - SSM Group, C - Conr
L - Local, P - Pruned, H - HP-bit set, F - Register flag.
T - SPT-bit set, J - Join SPT, M - HEDP created entry, E - Exti
X Proxy Join Timer Running, a - Candidate for MSDP Advert! sen
U - Urd, I - Received Source Specific Boat Report,
Z - Multicast Tunnel, i - MDT-data group sender,
Y - Joined HDT-data group, y - Sending to HDT-data group,
V - PD I Vector, v - Vector
Outgoing interface flags: H - Hardware switched, A - Assert winner
Timers: Uptime/Expires
Interface state: Interface, Next-Hop or VCD, State/Mode
I", 239.1.1.11, 00:05: 14/a topped, RP 0.0.D.0, flags: DC
Incoming interface: Null, RPF nbr 0.0.0.0
Outgoing interface list:
S 4 --a fi
IPmulticast routing can bemonitored byusing the show ip mroutecommand that displays the
IPmulticast routing table. As soon as thetreeis placed intoforwarding state, youwill seethe
IPmulticast routing tabic entry that shows thebranch of thetreefrom the perspective of a
router where the table has been displayed. In this case, router BR-I shows the leaf of the
forwarding tree for the root 1(1,1.1.1 and group 239.1.1.1. The leafisconnected tothe router up
thc tree branch (IIQ-1) viaSenalOT/0.111. Thedownstream interlace GigabitEthernetO.0,3 14
connects to the IPmulticast trafficreceiverIPphonethat is beingheld.
Ifthere isan issue with the building of the forwarding tree, and the IPmulticast routing table
does not showexpected entries, you can verify if the interfaces are enabled for multicast
routing by using the show ip multicast interface command. This command also shows how
many IP multicast packets were received so far.
BR-l#sh ip multicast interface SerialO/1/Q.111
Serial0/1/C.Ill is up, line protocol is up
Internet address is 10.1.6.102/24
Mult icast routing: enabled
Multicast switching: fast
Multicast packets in/out: 93904/0
Multicast TTL threshold: 0
Multicast Tagswitching; disabled
Multicast MOH Trace
This section describes the Cisco UnifiedCommunications Manager trace output for a multicast
MOII. Theoutput shows thetraceof a callfrom theheadquarters phone 2001
(SEP0024C4454AD8) to the BR phone 3001 (SEP0024C4455561) when 2001 places 3001 on
hold.
Multicast MOH Trace
Stationlnit : <O0Q0006 ) SoftKey Event soft K=rvlfv=s:.it,-i !!6;n','li; linelnstance-O
callReference-0. 1,100,4 9,1.2707'10.1.2.12*SEF0024C4454AD8
. . . truncated. . .
DialedHumber -i ,".-.tl.;-=-r-r - 2C; ; lineInstance- 1
callReference-32334506. 1,10 0,49,1.2712*10,1.2.12"SEP002 4C44 54AD8
. . . truncated...
Stationlnit: (000 0005) SoftKey Event ?-'-tKe/F-'a'it- 1 t *ii-v-==-i lineinstance.l
callP.eterenqe.3 2334 507. I 1,100,49,1.2713 *10.1.4.14"SEP0024C4455561
. . .truncated, , .
StationD: 1000 0006) =_=.: *-K. 3s=iTi. .rsi'3'.i.L i conerenceID-3233450 6
pa BBThruPartyiD=ciS7 772 94 remoteIpAddres a-IpAddr. type :0
ipAddr, OiOa0104000000000000 0000000000000 110.1.4.14) remoteportNumber-1665
mi HiSecondPacketSi ie-20 compresBType-6 (Kf.<,i i nylosrt V!22 '.ikl
RFC2B33PayloadType-0 qualifierOut.?. mylP: IpAddr.typa:0
ipv4Addr:0Da01020c 110.1.2.12) I1,100,49,1.2715"10.1.4.14"SEP0024C4455561
.. . truncatsd. ..
Stationlnit: 1000 0006) Sof MeyEvent softKayEvent.3 (Hold) lioemstance=l
callReferenCS-32334506. 1,100,49,1.27 18*10.1.2 .12~SEP002 4C44 54ADB
StationD: (0000006) CloseReceiveChannel conferenceID-32334506
pasaTtiruPaityID-16777294. myIP: IpAddr.type;0
ipv4Addr:OxCa01C20c(10.1.2.12) |1.100,4 9,1.2718*10.1.2.12"SEP0024C4454AnB
StationD: (0000006) StopMediaTransmisslon confaranceID-32334506
passThruPartyID.167 77294. myIP: IpAddr.typa: 0
ipv4Addr:Ox0a01020c (10.1.2.12) 1, 100,49, 1. 2718* 10 . 1 .2. 12*SEP0024C4454AD8
The initial sectionof thc outputshowsstandard call setup(most call setupoutput is truncated as
irrelevant for MOH) between the two registered IP phones. The call is answered, and the RTP
traffic starts flowing between the phonesthat are using the G.722codec(onlythe entry for
2001 is displayed in the figure).
Then headquarters phoneputs the call on hold. The RTPtraffichas stoppedflowing, and the
RTP channels are closed down.
icasl
HediaResourceManager: :waiting HrmAllocateMohRe
CREATING CHILD USING HP.GL AND DEFAULT LIST'
1,100,45,1.2718"10,1.2.12*SEP002 4C4454ADB
HSM, <ge UfohDevi ceGivenMrgl I>eviceName=MOH 2 DeviceType =7Q Group*0 Counter.0
Capability-." Multicast.! MRGL- 1,100,IS,1.2718"10.1.2.12"SEP0024C4454AD8
SMDMSharedData::findLoealDevice - Name-MOH 2 Key=.cddcdf la-3951-46ec-ad0 5-
c5af8df954s4 isActvie-1 Pid=(1,112 , 7) found!
1,100,49,1.2718*10.1.2.12'SEPC024C4454AD8
MediaReaourceCdpc (16) : ! f indDevjeeGivenList - Multicast Flag=l
1.100,49,1.2713"10.1.2.12*SEPC02 4C445 4AD8
MonDControl - tindMulticastSourceGivenSourceNum - SuccsbhI Device Name =
MOH 2, PayLoadType=4 1,100,49,1.2718"10.1.2.12 -SEP0024C4454AD8
MonDControl - handleMotiSucceas - Call Id . 32334508 AudioSourcelD = 1
M=j ticastPlag = 1 1,100,49,1.2718*10.1.2.12*SEP0024C4454AD8
StationD. (000OOOS) StartMu1 ticastHedisReception mcantlp: IpAddr.type:0
ipv4Addr,0xef0101O1 (23 9.l.i l) , myip, IpAddr.type:0
ipv4Addr;0x0a0104 0e(10.1 .4.14) ,1,100,4 9.1.2718*10.1.2.12 *SEP0024C4454AD8
StationD: (0000005) conferencelD-323 3450 7 maecPacketsize-20
compreeeionType: (4)Medii Payload C-711Ulan 64k 1,100,49,1.2718*10.1.2.12"SEP00
24C4454AD8
Stationlnit: 10000005) MulticastMediaResept ion 1,100.49,1.2719*10.1.4.14
'3EP0034C44 5S51
EReq - MRGI, SEARCH, TBV
This trace starts with the MRM searching the MOII server, fhc MRM uses thc MRGL lo locate
an appropriate MOH server.
The MOH server that is found is MOI I_2. This server has multicast capabilities. The next entry
shows that theMOI12 is registered (active), andit is partof the MRG list that has multicast
enabled (Multicast Flag-!).
Then MRM hits been allocating MOII resource MOII 2, This process evaluates theresource-
usage and determines if it can take on additional load. Ihe allocation is successful. MOII
resource MOH 2 is allocated as a multicast resource.
The bottom of the trace output shows thc process of joining the IP multicast group 239.1.1.1.
The IP phone SEP0024C4455561 (3001) with the IP address 10.1.4.14 has joined the group and
starts receiving the IP multicast stream that uses the G.711 mu-law codec.
Troubleshooting Multicast MOH from Branch
Router Flash
This topicdescribes the major issuesthat are relatedto multicastMOHfromthc branchrouter
flashthat is used during Cisco Unified Survivable Remote Site Telephony (Cisco Unified
SRST) and outlines how to troubleshoot them.
lulticast MOH from Branch Router Flash
Multicast MOH is configuredfor remote site users in Cisco Unified
Multicast MOHpackets generated by ttie Cisco UnifiedCommunications
Manager MOHserver are kept away from the IP WAN.
Branch router generates identical multicast MOH packets.
Cisco Unified
Communications
Manager MOH
Main Site
WAN
Multicast MOH from the branch router flash is a feature that allows multicast MOH streams to
be generated by gateways that are located at remote sites instead of streaming MOHfrom the
main site to the remote site over the IP WAN.
Because this feature is based on multicast MOH, Cisco Unified Communications Manager must
be configured to use multicast MOH instead of unicast MOH. This is recommended anyway to
reduce load at the MOHserver by just multicasting one stream that can be received by all
devices instead of streaming MOHindividually for eachendpointin separate RTPsessions.
The multicast MOII from the branch router flash feature is part of the Cisco Unified SRST
configuration. Therefore, the remote site router that will generate the multicast MOII stream for
the devices that are located at the remote site must be configured for Cisco Unified SRST.
Cisco Unified SRST does not need to be active (there is no need for a fallback scenario)
because a Cisco Unified SRST gateway that is configured for multicast MOH streams MOH alt
the time, regardless of its state (standby mode or Cisco Unified SRST mode). It works the same
with Cisco Unified Communications Manager Express in SRST mode.
Since Cisco IOS Release 15.x, the Cisco Unified SRST router can stream multiple files and use
multiple multicast addresses. By providing different MOHfiles, different MOII can be played
per phone.
)2010 Cisco Systems, Inc. Voice Quality and Media Resources Issues 6-27
Considerations When Using Multicast MOHfrom Branch
Router Flash
Consider the following when using multicast MOH from the branch router flash.
6-28
from Br.
that it lislens to locally generated multicast MOH stream
instead of stream generated by Cisco Unified CommunicationsManager MOH
server
Onlypossible if , ...-<..:< MOH := -:; ,->,. ::;.- Ihe following ''tlriO.*-.of the
streams that are generated by Cisco Unified Communications Manager MOH
server
Same destination IP address (multicast group address)
Same destination port number
Same codec
Multicast MOH (rom branch router flash supports only G 711.
G 711 must be used also for the stream generated by Cisco Unified
Communicalions Manager MOH server(s):
Put CiscoUnified Communications Manager MOH server intodedicated region
AlowG 711 between region of MOHserver and region used by branch phones
Limit all other inters lie calls to G 729
When using multicast MOH, IP phones andCisco Unified Communications Manager are
unaware that theIPphones listen to locally generated MOH streams. From a signaling
perspective, the IPphone is instructed to listen to a certain multicast stream, and the local
SRST gateway is generating a multicast MOII stream that is using identical settings such as
destination address (multicast group), destination port, and codec.
Multicast MOII in CiscoUnified SRSTgateways onlysupports the 0.71 I codec. Therefore,
G.71 1must alsobeconfigured between the Cisco Unified Communications Manager MOH
server and the branch IPphones Otherwise, because of a codecmismatch (signaling said
G.729. but the received RTPstream is G.711), the CiscoUnified Communications Manager
will signal a different codec tothe IP phone, and the IP phone will notplay the locally
generated MOH stream.
Toensure that Cisco Unified Communications Manager sends signaling messages to the phone
instructingthe phone to listen to a G.711 stream, configure regions like this:
Put the CiscoUnified Communications Manager MOII servers intoa dedicated region (for
example. MOH).
Put all branch devices intoa site-specific region (for example, Branch-1).
Allow G.7| I between regions MOH and Branch-1.
Make sure that region Branch-1 is limited toG.729 for callsto andfrom all otherregions.
Multicast MOH from Branch Router Flash Example
The figure shows a sample scenario for implementing multicast MOHfrom a branch router
flash.
Multicast MOH
Example
Cisco Unified
Commumcal=ons
Manager MOH
Configuraliorr
DA 239.1 1 1-4
DP 16384
a) Ma> Hops
(TTL) 1
WAN
ip access-list extended drop-mob
deny udp any host 239.1.1.1-4 eq 16384
inter's
ip ace
rial CO
roup drop-s
rail-manager-fallback
man-ephonee 1
majc-dn 1
lp source-address 10.1.5.102
moh mah-fils.su
multicast moh 239.1.1.1
port 16 384
The figure shows the configuration of a sample scenario.
Assume that the baseline configuration provides multicast routing in thc entire network, and the
Cisco Unified Communications Manager MOHserver is already configured for multicast
MOH. The branch phones arc configured to use multicast MOH with the G.711 codec.
The multicast MOI I stream that is sent toward the remote site should be blocked, and thc
multicast MOH from the branch router flash should be implemented at thc remote site.
The Cisco Unified SRST configuration of the remote site router is extended to include
multicast MOH. It uses the same multicast IP address and port that is configured at the Cisco
Unified Communications Manager MOH server that is located at the main site.
To stop the multicast MOHthat is generated by the main site Cisco Unified Communications
Manager MOHserver frombeing sent over the IP WAN, choose one of two options:
Set TTL to a low enough value at the Cisco Unified Communications Manager MOH
server: By setting thc TTL value in the IP header of the generated multicast MOH packets
to a low enough value, you ensure that the packets are not routed out to the IP WAN.
However, if the IP WAN link is one hop away from the Cisco Unified Communications
Manager MOH server and the main site phones are also one hop away from the server, this
method cannot be used because the main site IP phones would also be affected by the
dropped packets. In the example, Time to Live (TTL) is set to 1, and it is assumed that the
IP phones are in the same VLAN as the Cisco Unified Communications Manager MOH
server.
Filter the packets by an IP access control list (ACL): At the main site router, configure
an ACL that drops the multicast MOH packets at the IP WAN interface.
>2010 Cisco Syslems. Inc. Voice Quality and Media Resources Issues 6-29
Note When configuring multicast MOH inCisco Unified Communications Manager, you must
specifyhowto increment multicast streams: based on IPaddresses or based on port
numbers. Depending on the setting (increment that is based on IP addresses is
recommended), the IPACL must be configured appropriately to include all possible IP
addresses and port numbers. Note thateach enabled codec has a separate MOH stream.
At the branch router, the multicast MOH stream is sent out on the interface that is specified in
the ip source address command. This command is under the cail-manager-fallback
configuration mode (or telephony-serviceconfiguration mode when using Cisco Unified
Communications Manager Express in Cisco Unified SRSf mode). Therefore, the multicast
MOII stream that is generated at the branch router does not need to be blocked at the branch
router WAN interface.
6-30 Troubleshooting Cisco UnifiedCommunicalions (TVOICE) v8 0
wmm
Multicast MOH from Branch Router Flash Operation
The figure shows the operation of the multicast MOH from the branch router flash feature.
Multicast MOH from
Flash Operation
fittUdUttriBndUfttfutwtusut&A
' Unified CM = Cisco Unified Communications Manager
First, Cisco Unified Communications Manager signals the holdee to join its multicast MOH
group. Now, the question is whether multicast MOH is enabled at the branch router with the
following conditions:
The codec that was signaled to the holdee by Cisco Unified Communications Manager is
G.711.
The destination IP address (multicast group) and port are identical with those that are
configured at thc branch router Cisco Unified SRST multicast MOH feature.
If the mentioned conditions are met and the MOH stream IP packets that are generated by the
Cisco Unified Communications Manager MOH server are blocked toward thc holdee by one of
the following methods, the holdee at the branch will listen to multicast MOH that is generated
by the branch router:
The maximum hops value that is configured at the multicast MOH server is set to a low
enough value so that the packets that are generated by the Cisco Unified Communications
Manager MOH server are not sent out to the IP WAN.
An outbound access list that is applied at the headquarters router at the IP WAN interface
filters the multicast MOH packets that are generated by the Cisco Unified Communications
Manager MOH server.
Multicast routing is disabled at the IP WAN interface of thc headquarters router.
If multicast MOH is incorrectly enabled at the branch router (see preceding conditions), but thc
multicast MOH stream that is generated by the Cisco Unified Communications Manager MOII
server is correctly blocked (see preceding conditions), then thc holdee does not hear anything:
neither TOH. nor MOH.
>2010 Cisco Systems. Inc. Voice Quality and Media Resources Issues 6-31
Ifmulticast MOH is incorrectly enabled at the branch router and the multicast MOH packets
that are generated by thc CiscoUnified Communications ManagerMOH serverare not
blocked, then you are back at standard multicast MOII from the Cisco Unified
Communications Manager MOH server. In this case, the holdee hears MOH; however, the
corresponding packets have traversed the IP WAN.
Finally, if multicast MOH is enabled at the branch router correctly bul the packets that are
generated by the Cisco Unified Communications Manager MOI I server are not blocked,
packets of two streams are received by the holdee. Typically, this results in distorted audio.
6-32 Troubleshooting Cisco Unified Communicalions (TVOICE) v8 0 2010 Cisco Systems, Inc
1^^^
Troubleshooting Multicast MOH from Branch Router
The procedure of troubleshooting multicast MOH from a branch router feature and the
troubleshooting procedure of thc standard MOH are similar because of their similar set of
causes.
Troubleshooting
Branch Router
lulticast MOI
Issues are similar to standard multicast MOH.
In case of issues, verify these conditions:
Audio source file is stored in router flash.
Configurations of router and Cisco Unified
Communications Manager mutually match.
IP multicast routing works at the Cisco Unified
Communications Manager server and router sites.
G.711 is used toward the phones.
You can use all the commands and tools that were mentioned in the troubleshooting multicast
MOH section also for multicast MOH from branch router troubleshooting: show ip multicast,
show ip pim, debug ip pim, Cisco Unified RTMT Performance Monitor, and system
diagnostic interface (SDI) traces.
However, some of the causes could be unique to this feature. If issues are experienced, make
sure that the following conditions are met:
Use the show flash command to verify that the audio source file that is referenced in the
Cisco IOS configuration is actually stored in the router flash, and make sure that it contains
the expected music. Also, verify that the MOH configuration at the branch router has no
issues.
Ensure that the Cisco Unified Communications Manager MOH server and thc router share
the following attributes: same destination IP address (multicast group address), same
destination port number, and same codec (G.711).
Ensure that the multicast MOFI packets that are generated by the Cisco Unified
Communications Manager MOH server are blocked from entering the IP WAN. Ensure
also that the branch router generates identical multicast MOH packets. IP multicast routing
must be functional at the central and branch sites if branch IP phones and the branch router
are not mutually connected by Layer 2 connectivity only, but an additional router is in
between them.
If multicast MOH from the branch router flash supports only G.711, make sure that region
configuration at Cisco Unified Communications Manager sets this codec for branch
phones.
12010 Cisco Systems, Inc. Voice Quality and Media Resources Issues 6-33
Summary
This topic summarises the key points that were discussed in this lesson.
Summary
The MOH feature provides music to callers when their calls
are held by features such as user hold, call transfer, Call
Park, or conference.
MOH performance can be checked by using Cisco Unified
RTMT performance counters The MOH audio source file
needs to be uploaded to all servers of Cisco Unified
When troubleshooting MOHregistration issues, make sure
that the Cisco IP Voice Media Streaming Application service
is running, the MOH server is enabled, and no network
issues exist.
MOH loudness can be tuned by using the advanced
parameter Default MOH Volume Level within the Cisco IP
Voice Media Streaming Application service.
IfTOH is played instead of MOH, check for a mismatch
between the codec that is used by the MOH server and the
endpoint registration region. Check also if a device is not in
place, if the CAC rejects the MOH streaming, if the MOH
server is operational but running out of resources, or if media
resources are configured improperly.
For multicast MOH to work, the multicast MOH must be
enabled on MRG, the audio source, and the MOH server.
Multicast MOH frombranch router flash supports only G.711
and possible only if the locally generated MOHstream shares
the attributes of the streams that are generated by the MOH
server.
In this lesson, you have learned to explain the common issues that are related to MOH and to
identifv the most likclv causes of these issues.
6-34 Troubleshooting Cisco Unified Communications (TVOICE) v8 i 2010 Cisco Systems, Inc.
*mm
itt
References
For additional information, refer to this resource:
Cisco Unified Communications Manager Features and Services Guide, Release 8.0(2),
Music On Hold at
http: www.ci.NCo.com enUS partner'docs'voice ip comm/cucnr'admin.'S 0__2<
ccmfeat fsinoh.html
12010 Cisco Systems. Inc. Voice Quality and Media Resources Issues 6-35
6-36 Troubleshooting Cisco Unified Communicalions (TVOICE) v8 0
i 2010 Cisco Systems, Inc
feM
Lesson 2
Troubleshooting MTP Issues
Overview
A Media Termination Point (MTP) device is required in scenarios in which Cisco Unified
Communications Managermust relaycalls that use special signalingmethods like H.323 fast
start, Session Initiation Protocol (SIP) early offer, H.323 supplementaryservices, or
repackctization of audio stream.
MTP, a Ciscosoftware application, installson a serverduringthe software installation process.
MTPis part of the CiscoIPVoiceMediaStreaming Application serviceon the Cisco Unified
Objectives
Upon completing thislesson, youwill beabletoexplain thecommon issues that are related to
MTP and identify the most likely causes of these issues. This ability includes being able to
Reviewthe major scenarios in which MTP is used and outline general issues that can be
experienced with MTP
Describe the MTP registrationand nonresponsive software issues and explain how to
troubleshoot them
Describe the major reasons why MTP allocation might fail at call setup
MTP Review
This topic reviews ihemajor scenarios inwhich MTP is used, describes MTP requirements for
named telephony events (NT hs), andoutlines general issues thatcanbeexperienced with MTP
MTPs are used in different scenarios:
p Repacketization of an audio stream:
Transcode a-law to mu-law and vice versa.
Bridge two connections by using the same codec but
different sample sizes.
H.323 supplementary services:
Extend supplementary services to H 323 endpoints that do
not support the H.323v2 Open Logical Channel (OLC) and
Close Logical Channel (CLC) request features of the Empty
Capabilities Set (ECS).
Add features such as hold, transfer, park, and conferencing.
If no MTP is available, the call will proceed but willbe unable
to invoke supplementary services.
MTPs have many possible uses:
Repacketization of a stream: An MTP can be used to transcode a-law to mu-law and vice
versa, or it canbeused tobridge twoconnections that use different packetization periods
(ditferent sample sizes).
H.323 supplementary services: MTPs canbeused to extend supplementary services to
H.323 endpoints that do not support thc H.323v2 Open Logical Channel (OLC)and Close
Logical Channel (CLC) request feanues of the Empty Capabilities Set (ECS). Ihis
requirement occurs infrequently. Cisco H.323 endpoints support KCS, andmost third-pariv
endpoints have support as well. When needed, an MTP is allocated and connected to a call
on behalfof an H.323 endpoint.
Once inserted, the media streams are connected between thc MTP and the 11.323 device,
and these connections are present for the duration of the call. The media streams that are
connected to the other side of the MTP can be connected and disconnected as needed to
implement features such as hold, transfer, and so on. When an MTP is required on an
H.323call and none is available, the call will proceed but will not be able to invoke
supplementary services.
MTP Review (Cont.)
MTPs are used in different scenarios:
H.323 outbound fast start:
For inbound fast start, no MTP is required.
Outbound calls on an H.323 trunk do require an MTP
when fast start is enabled.
Named telephony events (NTEs) RFC 2833:
Method of sending DTMF.
The tones are sent within RTP packets but not as digitized
audio tones.
Digital data is used and is distinguished from audio
payload by the RTP packet type field (NTE).
- MTP required when two endpoints do not have a method
in common for sending DTMF between them.
H.323 outbound fast start: H.323 defines a feature that is called fast start, which sends
H.245parameters withinH.225. This feature reducesthe numberof packetsthat arc
exchanged dunng a call setup, therefore reducingthe timefor mediato be established. It is
useful when two devices that are using H.323 have high network latency between them,
because the time to establish media is dependent on that latency.
Cisco Unified Communications Manager distinguishes between inbound and outbound fast
start that is based on the direction of the call setup. The distinction is important because the
MTPrequirements are not equal. For inboundfast start, no MTPis required. Outbound
calls on an H.323 trunk do require an MTP when fast start is enabled. If only inbound calls
are problematic, youcan use inboundfast start to solvethe issuewithout alsoenabling
outbound fast start. Atypical scenario wouldbe an H.323 gatewaythat supports only
H.323 fast start, but it is used for inbound calling only.
NTEs (RFC 2833): Dual tone multifrequency (DTMF) tones can be used during a call to
signal to a far-end device for navigatinga menu system, entering data, or other
manipulation. These DTMF tonesarc processed differently than DTMF tones that are sent
during a call setup as part of the call control.
NTEs defined by RFC 2833 are a method of sending DTMF from one endpoint to another
after the call media is established. The tones are sent as packet data by using the already
established RTP streamand are distinguished fromthc audio by the RTP packet type field.
For example, the audioof a call canbe sent on a sessionwithan RTPpacket type that
identifies it as G.711 data, and the DTMFpackets are sent with an RTP packet type that
identifies them as NTEs. The consumer of the stream uses the G.711 packets and the NTE
packets separately.
MTP for NTEs
Determine when MTPs are required for NTEs.
Two non-SIP endpoints do not require MTPs:
IP phones send SCCP to Cisco Unified Communications
Manager.
H.323 gateway sends H.245 signaling events to Cisco
Two Cisco SIP endpoints do not require MTPs:
All Cisco SIP endpoints support NTEs.
DTMF is sent directly between the endpoints using NTE.
No MTP required for G.711 calls.
Combination of a SIP and a non-SIP endpoint might
require MTPs:
This depends on the endpoint.
Cisco Unified Communications Manager dynamically
allocates MTPs on a call-by-call basis.
InCisco Unified Communicalions Manager, anM'fP is required when two endpoints do not
have a method incommon for sending DTMF between them or when the system configuration
specifies that one MTP must be allocated. Verify the types ofendpoints that arc planned for the
system by using the following rules:
Two non-SIP endpoints do not require MTPs. All Cisco Unitied Communicalions
endpoints other than SIP sendDM'IT toCisco Unified Communications Manager via
various signaling paths, and Cisco Unified Communications Manager forwards the DTMF
between dissimilar endpoints. For example, anIPphone might use Skinny Client Control
Protocol (SCCP) messages toCisco Unified Communications Manager to send DTMF.
which then is sent toanH.323 gateway via 11.245 signaling events. The two endpoints have
a common method of sending DTMF toCiscoUnitied Communications Manager.
Two Cisco SIP endpoints donot require MTPs. All Cisco SIP endpoints support NTEv
therefore. MTPs can beavoided entirely. DTMF is sent directly between the endpoints by
using NTF. If all endpoints areCisco SIPdevices, then no MTPs are required for DTMF
conversion.
SIP calls that use G.71 1never require an MTP.
Acombination of a SEP endpoint and a non-SIP endpointmight require Ml Ps. RFC 2833 is
not limited to SIP andcanbe supported indevices with othercall control protocols. For
example, Cisco Unified IPphones that run either anSCCP or SIP stack can support NTEs
in bothmodes. Somedevices support DTMF via multiple methods. Otherdevices suchas
the CiscoUnified Wireless IPPhone 7920can sendonlySCCP, and someotherscan send
only NTF (such as the Cisco Unified IP Phone 7960 with a SIP stack). Cisco Unified
Communications Manager can allocate MTPs dynamically on a call-by-call basis thai is
based on the capabilities of the pair of endpoints.
hm
MTP Requirement Review
The figure demonstrates when there is a need for MTPs.
MTP Requirement Review
"<
-T--1 i-^
-TV* L-- -
ÎSII^r 1--V
MTPsare required whentwoendpoints cannot agreeon the samepacket size or whenG.711 a-
law to mu-law conversion is required.
In addition, MTPsare required for H.323 devicesthat do not supportsupplementary services or
when H.323 outbound fast start is used.
Another reason for the need of an MTP is that no common DTMF method is found. In a Cisco
Unified Communications Manager environment, this occurs only on some calls fromnon-SIP
devices to SIPendpoints thatarereached viaa SIPtrunk. If theSIPtrunk is usedonly for
inbound calls, no MTPis required. If the SIPtrunk is used for outbound calls and all devices on
theotherendsupport a dynamic negotiation of NTE, then MTPs areallocated onlyif needed on
a call-by-call basis.
If not all devices on the other end of the SIP trunk support dynamic negotiation of NTE, in
otherwords, they require Session Description Protocol (SDP) tobeusedwith the first INVITE,
then the MTPRequired checkboxmust be set at the trunk. Otherwise, calls to devices that
require SDP to be used with the first INVITE will fail.
If the MTPRequired checkbox is set, then all calls must use thc G.729codec. If G.729is not
supported onthe other end, the call fails. Otherwise anMTP isallocated for the outgoing call.
Voice Quality and Media Resources issues
General MTP Issues
6-42
This section names the general issues that might beexperienced when using MTP.
General problems related to MTPs;
' No supplementary services available in H.323.
* Call setup fails when MTP required (gateway or trunk).
" DTMF issues when NTE not supported by an endpoint.
Issues with mixed SIP endpoints.
General MTP issues:
* MTP cannot register because of network issues or
misconfiguration.
MTP registeredbut not available to calls because of running
out of resources or misconfiguration.
MTP is used in various call situations. If MTP does notwork as expected, consider these
problems that are directlyrelated to MTP unavailability:
Supplementary services are unavailable in an H.323 environment.
Call setup fails when MTP is required ina scenario that uses gateway or trunk. If MTP
cannot beallocated and it isrequired for acall, thc Cisco CallManager service parameter
Fail Call IfMTP Allocation Fails sets thebehavior for thecall. If the parameter is set to
false, the call scnip coniinues without MTP, but if it isset toTrue, the call fails together
with the failed M'fP allocation process.
When NTEis not supported by an endpoint, different DTMFmechanisms must be
translated for incompatible endpoints. If MTP is unavailable to perform thistranslation.
DTMF might not work as expected.
Various problems occur ina SIP environment with mixed SIPendpoints.
All of these problems canbecaused by improper MTP functionality ina cluster. These are the
most common MTP issues:
MLP cannot register because of network issues or because of misconfiguration at the Cisco
Unitied Communications Manager or the Cisco IOS MTP, if used.
MT Pcanbesuccessfully registered but is unavailable tosupport calls. MTP can run out of
resources, or its association with a device that requires MTP services might be
misconfigured.
TroubleshootingCisco Unified Communicalions (TVOICE) v8 0
Troubleshooting MTP Registration and
Nonresponsive Software Issues
This topic describes MTP registration and nonresponsive software issues and explains how to
troubleshoot them.
MTP Registration
HQ-ltshow accp
SCCP Admin State: UP
Gateway Local Interface; LoopbackO
IPv4 Address; 10.1.250.101
Port Number; 2000
IP precedence; 5
User Masked Codec list: None
Cell Manager: 10,1.1.1. Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 1
Trustpoint; N/A
HTP oper State; ACTIVE - Cause Code; NONE
Active Call Manager: 10,1.1.1, Port Number* 2000
TCP Link Statue: CONNECTED, Profile Identifier: 1
Reported Max Streams: 8, Reported Ma* OOS Streams: 0
Supported Codec g711ulan. Maximum Packetiaation Period. 30
Supported Codec rfc2833 dtmf, Mauimum packetiiatioo Period 30
Supported Codec rfc2B33 pa as-thru. Maximum Packetiiatioo P riod: 30
Supported Codec
inband-dtmf to rfc2633 conversion. Maximum Packet!zation
Period; 30
TLS ENABLE 0
CiscoUnifiedCommunications Managersupports the software and hardware MTP. The figure
shows how to verify that hardware MTP (Cisco IOSEnhanced Software MTP) is registered
withCiscoUnified Communications Manager. Thetypical configuration of Cisco IOSMTPis
like this:
voice-card 0
dspfarm
dsp services dspfarm
i
interface LoopbackO
ip address 10.1.250.101 255.255.255.255
i
seep local LoopbackO
seep can 10.1.1.1 identifier 1 version 7.0
seep
i
seep ccm group 1
associate ccm 1 priority 1
associate profile 1 register MTP-HQ1
i
dspfarm profile 1 mtp
codec g711ulaw
maximum sessions hardware 4
associate application SCCP
no shutdown
Ihe show seep command shows that the SCCP application on the Cisco IOS router is up, and
the MTP has been registered uith the Cisco Unified Communications Manager at the IP
address 10.1.1.1. The MTP operational state is ACTIVE. This MTP supports the codec G.711
mu-law. Ifthc operational state shows anything other than ACTIVE, you have experienced an
MTPregistration issueas seenin these examples:
HQ-inshow accp
SCCP Admin State: "JF
Gaieway Local Interface. LoopbackO
IPv4 Address: 1 0 .'. .250 .10 !
Pert Kumber; 200u
IP Precedence: :
User Masked Codec "! ist : None
Ca:: Manager; zz . I .1 ,: , Port Number: 20OD
Priority: N/A, Version: 7.C, Identifier; 1
Tit.Stpoir.t : N/A
MT? Oper State: ACTIVE_ zr, PROGRESS Cause Code: TCP_CONN_ERROR
Active Call Manager: KONK
TC? Link Statas: CONNECT^PENDING, Protile 2dentifier: 1
Seportea yax Streams: S, Reported Max OOS Streams: 0
Supported l.cdec: g/llulav. Maximum Packetization Period: 30
Supported Cede;: : il;:2R33 dtmf , Max imum Packetization Period: 30
Supported Codec. rfc2333 pass-thru, Maximum Packetization Period: Kl
Supported Coder: i:;band-dtmf to rtc:>833 conversion, Maximum Packetizacicn
Period: 30
HQIwshow seep
SCCP Adm;r. State: 'JP
Gateway Local Interface: LoopbackO
IPv4 Addiess: 10.1.250.101
Fort KjTier: 20G0
IP Precedence: 5
User Masked Coac.z 1 _st : !.'one
Call Manaqer. :c. 1.1.1, Port Nmber: 2000
Priority: N,'A, Version; ". 0, Identifier: 1
"I rustpoir.t : N/A
M1F Oper State: ACTIVE IK_PROGRESS - Cause Code: KEEPAL1VE_FAII,!-;d
Active Ca". 1 Manager: NGNF
TCP Link Status: NCT_CCNKECTED, Profile Identifier: 1
Reported y.&x Streams: 8, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period; 30
Supported Codec rl.j-2833 dtmf. Maximum Packetization Period: iO
Supported Codec; rfc2833 pass-thru. Maximum Packetization Period: 50
Supported Codec: ir.band-dtmt tc rfc2833 conversion, Maximum Packet izat ior.
Period: 30
TLS : ENABLECJ
6-44 Troubleshooting Cisco Unified Communications (TVOICE! v80 2010Cisco Systems. Inc
MTP Registration (Cont.
HQ-l#sfaow seep ccm group 1
CCM Group Identifier: 1
Description: Nona
Bindad Interface: None
Associated CCM Id; 1 Prioritv in this CCM Group: 1
Associated Profile: 1, Registration Nam : MTP-HQ1
]
Registration Retries 3, Rgiotration T neoutr. 10
Keepalive Retries: 3 Keepalive Timeout 30 sec
CCM Connect Retries; 3, CCM Connect Interval; 10 sec
Switchover Method: GRACEFUL, Switchback Method: GRACEFUL GUARD
Switchback Interval: 10 sec. Switchback Timeout: 7200 sec
Signaling DSCP value co3, Audio DSC? value; e
Media Resources > MTP
- * DUPHfAfi B#c# Pb"I
fcQi*T*rftd wibin 10.1,11
*Werd *Jrhlit 1 11
Service Parameters > Cisco IP Voice Media Streaming Application
-NMu TlrtMMtm PBIM (MTP) PeW4in*trr>-
For hardware MTPto register properly, its registration namethat is configured at thc CiscoIOS
router must match the registrationname that is configured at the Cisco Unified
Communications Manager. The figure showsthat thc name MTP-HQ1 matches as required.
This is the most common error when configuring hardware MTPs. Also ensure that the
hardware MTP is not shut down at the Cisco IOS router.
Thc software MTPis provided by theCisco Unified Communications Managercluster. Its
name is generated duringthe installation process. The figure showssoftwareMTPMTP_2
collocated with Cisco Unified Communications Manager software. If you experience
registration problcms withsoftware MTPs, makesure that the MTPis enabled. In Cisco IP
VoiceMediaStreaming Application serviceparameters, RunFlag for MTPshouldbe set to
True. Another service parameter Call Count sets the number of resources that MTP can
support. This is important during the MTP allocation process.
For both hardware and software MTPs, make sure that the Cisco IP Voice Media Streaming
Application service is running. Verify thisby navigating to Cisco Unified Serviceability >
Tools > Control Center - Feature Services. It is rare, but the software can also stop responding.
If the serviceis reportedas not running, restart the CiscoIP VoiceMediaStreaming
Application service.
MTP Registration Issues
This listsummarizes themost common causes of why MTP registration fails.
The common causes for MTP registration issues:
* Cisco IPVoice MediaStreaming Application not running.
- Hardware MTP shutdown or its name does not match.
Network connectivity or DNS issues.
* Access list in the path filtering SCCP.
* Software MTP deactivated.
* Cisco Unified Communications Manager database broken.
Depending on the MTP implementation, if hardware or software MTP has been used, these
could be the most common causes of failed registration:
The Cisco IP Voice Media Streaming Application service is notrunning properly. It has
been deactivated or is down due to software error. Restart the service in Cisco Unified
Serviceability.
Hardware MTP is shut down or its name does not match with the Cisco Unified
Communications Manager configuration. Make sure that the no shutdown command is,
appliedwithinthe digital signal processor(DSP) farmprofile for MTP. For most Cisco IOS
versions, the noshutdown command does not show when displaying the configuration.
(However, the shutdown command does show in the configuration.)
Hardware MTPis usuallylocated at close proximity to the Cisco Unified Communications
Manager server, but IPnetwork connectivity issues could alsobe present, preventing
successful registration. Software MTPs arc implemented by selected servers ina Cisco
Unified Communications Manager cluster. IP connectivity issues between individual
cluster members will also affect MTP registration within the cluster. If names are used
instead of IP addresses, verify that Domain Name System(DNS) name resolution works as
expected.
If hardware MTP is behind a firewall or access list, ensure that theTCP port 20(10 (SCCP)
is enabled for the communications between the MTP and the Cisco Unified
If software MTP is implemented, be sure that it is activated in Cisco IP Voice Media
Streaming Application service parameters.
If the MTP registration status reported is inconsistent across Cisco Unitied
Communications Manager servers, verify that theCisco Unified Communications Manager
database is not broken. Restore the database replication between the cluster servers.
6-46 Troubleshooting Cisco UnifiedCommunications (TVOICEl v8 0
MTP Allocation
This topic describes the major reasonsthat MTPallocation might fail at call setup.
iVITP Allocation on SIP Trunks
Ifa SIP trunk is used for inbound calls only, no MTP
resources are required.
For outbound calls, MTPs are dynamically allocated if
required (one endpoint does not support NTE).
Dynamic allocation not supported by all devices.
MTPRequired check box (deactivated by default) can be
activated on SIP trunks in such cases.
Allows communication to devices that require SDP to
be used in first INVITE.
* G.711 or G.729 options exist when MTP Required is
activated.
WhenMTPsare requiredon a SIP trunk, youcan manageMTPallocation in severalways:
No MTPis required on a SIPtrunk if the trunkis used for inboundcalls only. Unless the
configuration requires MTPs to be allocatedfor each call, no MTPis allocatedon incoming
calls.
For outbound calls, MTPs are allocated dynamicallyby default. This means that MTPs are
allocated as needed on a call-by-call basis.
Some devices do not support a dynamic allocation of MTPs. If such devices should be
reached through a SIP trunk, the SIP trunk always must be configured to allocate an MTP.
Enforce this by activatingthe MTP Required check box at the SIP trunk configuration
page. Now, calls will alsowork for devicesthat do not supportdynamic MTPallocation.
However, Cisco Unified Communications Manager will now allocate an MTP for all
calls includingthose in which, because of a common DTMF method, no MTPs are
needed. In addition, either G.711 or G.729 must be used for all calls that go through this
trunk.
Note SIP uses SDP for establishing the parameters of a session, and SDP is embedded into SIP
messages. When an MTP is forced, the SDP is sent with the INVITE message that initiates
the call. When an MTP is not forced, it is allocated (if needed) after the INVITE message. In
that case, the INVITE message does not contain SDP, and it is sent later in the call setup. If
the far-end device supports only INVITE messages with embedded SDP (early offer), then
the MTP Required parameter must be checked (enabled).
MTP Allocation Failure Decision Tree
Thedecision tree that is shown inthe figure explains what happens when MTPallocation fails.
6-48
Assume that, because of an 11.323 outbound fast startor toprovide supplementary services to
anH.323 gateway that doesnot support these services on itsown(11.323 before version 2), you
attempt an MTP allocation that fails. The following happens:
li the Fail Call 1/MTP Allocation Fails Cisco CallManager service parameter is set to
True, the call fails.
It that service parameter is set to False, then thecall is set up without using theappropriate
feature.
If the reason for the MTPallocation is an activated MTP Requiredcheck box ai a SIP trunk and
the allocation fails, then the following happens:
If the Fail CallOver SIPTrunk if MTP Allocation Fails CiscoCallManager service
parameter is set to True, the call fails,
If that serviceparameter is set to False, call setupdepends on whetherthecall actually
requires an M1P if yes. the call fails; if no, the call is set up without an MTP.
For all other calls that require an MTP(for example, different packet sizesor the needfor
G.711 a-law to mu-law conversion), an MTP allocation failure will make the call fail.
Troubleshooting Cisco Unified Communications (TVOICEl u8 0
* MTP Allocation Issues
This section lists the most common causes of MTP allocation issues.
MTP Allocation Issues
The common causes for MTP allocation issues:
No MTP registered at Cisco Unified Communications Manager.
MTP registered but running out of resources.
MRG and MRGLmisconfigured, MTP not available to endpoint.
Network connectivity issues to MTP.
If the Cisco Unified Communications Manager fails to allocate MTP resources when required,
consider these reasons:
No MTP is registered at Cisco Unified Communications Manager. Make sure that the
appropriate MTP is successfully registered before calls are placed.
MTP is registered, but it might be running out of resources. MTPs can support a finite
number of sessions, depending on whether hardware or software MTP is implemented.
Make sure that more resources are deployed if the MTP frequently runs out of resources.
The media resource might be misconfigured when you use Media Resource Group (MRG)
and Media Resource Group List (MRGL), making an MTP unavailableto the endpoint
when needed. An MRGL that contains MTP resources should be associated with a device
that requires MTP.
MTP seems to be registered, but it is no longer accessible, and it cannot be allocated.
Keepalives are exchanged between MTP and Cisco Unified Communications Manager with
a default 30-second period. If network connectivity issues appear between two keepalives,
this kind of situation can be experienced. In this case, MTP would fail and will start
renewing its registration.
2010 Cisco Systems. Inc. Voice Quality and Media Resources issues 6-49
Verify MTP Utilization
6-50
Ifproblems with MTP allocation areexperienced and MTP is properly registered with Cisco
Unified Communications Manager, make sure that the MTP does not run out of resources.
;t:nty M
Hardware MTPperformance can also be verified by using
show dspfarm profile.
-...1.-.
t-c3fli
-f,.
* |TJ .'is:
>=.=:.
a =..=.=.
nci=.
=:.(..
?:n-
Dc i:ftf=o,='i
[V *.=1P =..'=.*'.
The figure showsthe CiscoUnified Real-Time Monitoring Tool (RTMf) performance counters
for the MTPdevice. You can select fromthe counters to display variousvariables. View how
many resources are available, how many resources are used at the moment, which MTP runs
out of resources, howmanyresources couldnot be allocated due toopeningpott errors, and so
on. I he counters can be displayed per individual MTP.
The figure shows two \ariables for the hardware MTP named MTP-IIQI that has thc total
capacity of four resources. The figure shows that one resource has been used and three are
available for next calls.
Performance of the CiscoIOSM'l'P can also be verified by using the showdspfarm profile
command. The following command output showsthat four resources have beenconfigured in
total, but one is used and three are available:
HQ-liishow dspfarm profile 1
Dspfarm Profile Configuration
Profile ID = 1, Service = MTP, Resource ID =. 1
Prof ile Description :
Profile Service Mode : Non Secure
Profile Admin State : UP
Profile Operation State : ACTIVE
Application SCCP Status : ASSOCIATED
Resource Provider : FLEX_DSPRM Status : UP
Number of Resource Configured : 4
Number of Resource Available : 3
Hardware Configured Resources : 4
)2010 Cisco Systems, inc
Hardware Available Resources i 3
Software Resources : 0
Codec Configuration
Codec : g711alaw, Maximum Packetization Period : 30
Check also the Cisco CallManager service parameter MTP and Transcoder Resource Throttling
Percentage. This parameter defines a percentage of the configured number of MTP (or
transcoder) resources. When the numbcr of active MTP resources is equal to or higher than thc
percentage that is defined in this parameter, Cisco Unified Communications Manager throttles
(stops sending) calls to this MTP.
This throttling allows Cisco Unified Communications Manager to check the availability of
resources in the next MTP in the MRGL that has matching codecs on both sides in an attempt
to locate an available resource.
If Cisco Unified Communications Manager has checked every possible MTP in the MRGL and
all have been throttled, Cisco Unified Communications Manager will check all MTPs once
more and attempt to make a best match from even the throttled MTPs. If a resource remains
unavailable after Cisco Unified Communications Manager has extended the call to the MTPs
that Cisco Unified Communications Manager determined to be the best match, the call will fail.
In some cases, Cisco Unified Communications Manager may perceive that a resource on
hardware MTP is available, yet the actual port on the hardware has not been released. This
parameter allows Cisco Unified Communications Manager to make the best-match judgment to
extend the call to an MTP that offers the best chance of successfully connecting the call. This
service parameter is set to 95 percent by default.
)2010 Cisco Systems, Inc Voice Quality and Media Resources Issues 6-51
MTP Allocation Trace
The trace output of successful hardware MTP allocation is shown in this section.
i -i~ ^3c
StationD: (0000003) DialedNumber ; =.. : ,-s. : . .-;. i,:.> 1 inel instance* ]
callReerence=17183516. 1,100,49,1.975*10,1.2.13"SEP0024C445 4AD8
StationD: (0000003) (1.100.9,9) calllnfo callingPartyKamc =' '
callingParty=?=13 555 2 00: eg pnVo i ceMa ilbox. alternateCallingparty=
ca11edPartyName =' ' calledParty.12125550120 cdpnVoiceMailbox=
origiDalCalledPartyName='' originalCal1edParty.12125550120
callType=2( i linelnstai,ce=l callReerence = 171B351=S . version;
3572 0013 1, 100, 49,1.979" . - - 'SEP 0024 C4454 ADS
. . . truncated. .
SMDHSha redData: : i . ,.;.' Key=b4b42cbc-c 8al-l 999-
dlO8-ddb8ic0efe7e . '. Pld= (1 ,173 , 3)
found 1,100,49, 1.979 "10. 1.2. 13"SEP0024C44 54ADB
truncated.. .
MediaResourceHanager; =waiting MrmAUocaleMtpResoarceHeo. CI= 1718 3518 ,
Count=I 1,100,49,1.979*10.1.2.13"SEPO024C4454AD8
MediaResourceManager, =a iting MrmAllncateMtpKesourceReq - CHEATING CHILD
USING MRGL LIST 1,100,49, 1.97 9 * 10. 1.2.13*SEP0Q24C4454ADB
. . .truncated. . .
MEM:=getMtpDeviceSlvenMrgl HRGL=MTP mrg.1,100,49,1.979*10.1.2.13
'SEP0024C4454ADB
MHM= :getMtpDevi ceGivenMigl Devi ceNau-ie=MTF-HQl DeviceType-83 GroupÔ
Counter=D Capability=0 Multicast.0 NRGL.9--11 961t>- 95f d-7f 9 9 -4d 09 -
37 6de7d6a26 0 1, 100, 49,1. 979"10.1.2.13"SEP0034C4454AD8
The calling IP phone (SEP0024C4454ADIS) with the IP address 10.1.2.13 places a call to the
PSTN through an 11.323 gateway. The gateway is configured for 11,323 fast start, and M'fP is
required for all outbound calls that are placed through this gateway (the MTP required check
box is marked on the gateway configuration page).
The beginning of the trace shows the calling IP phone (SEP0024C4454AD8) dialing
912125550120. All trace output showing digit analysis and path selection was truncated to
focus on MTP allocation solely.
The H.323gateway with IP address 10.1.250.102 was selected. The gateway has been
registered and ready to take calls (isActive- 1).
Media Resource Manager (MRM) starts searching for M'fP to support this call. The gateway
has been associated with the MRGI. (its name is not show in the output) that points to the MRG
that is called MTPmrg. The MRG lists the hardware MTP that is named MTP-HQ1.
6-52 Troubleshooting Cisco Unified Communications (TVOICE) v8 2010 Cisco Systems, Inc
MTP Allocation Trace (Cont/
MRKi igetHtpDeviceQiveaHrgl 'J?TT'E7, T(?r FF.3H !=BF=V. T.T
Lis: 1,100,49,1.979"10.1.2.13'SEP0024C4454AD8
MEM: :ge tHtpDevicaGivanMrgl !:< : ".!.s,i.MS!ii 2 SJe; "(=;; !) =; 10 Gtoup-1 Counter.O
Capability!) MultiCast-0 KRGL.9all961S-8Sfd-7f99-4309-
37 6de7d6a2G0'1,100,49,1.979"10.1.2.13"SEP0024C4454AD8
. , . truncated. . .
MadiaRe aourcecdpr (121 ::waiting MnnAiiocateKtpResourceReii - CI=171B3 51B
Count-1 TryPflSBThru.D 1.100.49.1.979*10.1.2.13"SEP0024C4454ADB
SMDMSharedData: =findLoealDevica - Name-HTP-HQl Key-9fac2eeS-Bb29-lea4-la37-
c413153343e8 isActivo.l Pid-II,119,2)
found 1,100,49, 1.97 9"10. 1.1. 13 "SEF002 4C44 54ADB
. tr ited.
MediaRaourcaCdpc(12) : :f iodDevicaGivanLlat - Nama.MTP-HOl typs=-l isTRPMuat-1
-eight-0 1,100,49,1.979"10.1.2.13'SEP0024C44S4=DB
MediaResourceCdpc(121::seadHtpAllocatsBequeBtToDBiice HtpResourco-MTP-H01
Capo9fac2ee6-BbZ9-lsa4-la37-
=r413153 3438'1. 100, 49,1. 97 9" 10 . 1 . 2 . 13 ' SEP0024C4454AD8
HedisTerminationPolntControU2) : .waiting AllocataMtpRasourceReq -
IcBpCount.rogion) . A(0,) , B11, Default) , regDsvCap-OnO, eupDevCap-0x10b,
passthru-0, reflourcsCount-II 1,100,49,1.97 9"10.1.2.13"SEP0024C4454AD8
. . . truncated, . .
MedlaTermlnatioaPointCoiitrol (2) : : logResourceStatuainTrace - - Device
Name.MTP-HQI RsBourceAvai l8t=le*4
ReaourcaUbUd.O 1, 100,49, 1.97 9" 10.1.2. 13"5EP002 4C445 4AD8
There is also software MTP named MTP 2 that is configured in Cisco Unified
Communications Manager, but this MTP is not part of the MRG and, therefore, it is not taken
into consideration.
The allocation process of MTP-HQI is shown in thc figure. The MTP is registered
(isActive=l). The call requires the MTP (the requirement was set at the gateway).
Cisco Unified Communications Manager has sent the allocation request for a single resource to
the MTP. The MTP has responded, indicating that it is available for the call.
6-54
,WI
MRM: ;updateMtpCounter davNaae-MTP- HQl. count Chang e=l | 1,100 , 49 , 1. 97 9
'10.1.2 .13'SEP0024C'14S'1ADB
HediaResourr;eCdpc(12) :: shutting down MrmChildStopConf 1 ,100 .49.1 . 979'10 .1 .2
.13"SEP0024C4454ADB
. . .truncated. . .
HediaTerminationPointControl (2) ::atai MediaExchangeAgenaOpenLogi calChannel
- Partyld - 1,100,49,1.979"10.1.2.13"3EP0024C4454AD8
MediaTaroinationPnintControl(2) ::star StationOutpu tOpenReceiveChannel -
TCPPid = [1.100.9.6] mylP: OuSSfaOlOa (10.1,250.101) Conference ID; 16779226.
MediaPartyld: , msecPackeLSizet 20 compressionType;
4 1,100,49, 1.979'10. 1.2. 13"SEP00 24C44 54ADB
MediaTerminati onPointControl (2)::star StationOutputStartHediaTransmisaion -
TCPPid- 11.100.9.61 n=,IP: 0x65fa010a (10.1.250.101) |1,100,170,10.1"** *
MediaTerminationPoictControl(2) :;Btar StationOutputStartHediaTransmisaion -
ConferencelD: . _ , MediaPartyId: .="" .'-. RemotelpAddr: 0x66fa010a
(10. 1.250.102) RemoteRtpPortNumber : 17236 msecPacketSiie: 20
COffipreSEicnType : 4 1,100,170,10.1"*"'
.. .truncated.. .
StationD: (00 0000 3) Cp e:iRe L-eiv eChanne 1 confer enceID= 17183 516
pasBThiuPai tyID=-16777255 m: 11 is econdPackstsi :==; 0
;orrpre=3BionType=4 iHedia_ Payload G711Ulau6 ik) RFC2833PaylaadType=0
qualifierln.? sou roe IpAddr-IpAddr. type : 0
ipAddr:0x0a01 fa65000000000000000 000000000 (10,1.250.101) . mylP: IpAddr.type;0
ipv4Addr:Ox0a01020d (10.1.2.13) 1,100,211,38.1"*"*
The MTP allocation process is complete when the number of resources that are used has been
incremented and the search for other MTPs has stopped.
The MTP has been allocated now, and ihe Cisco UnifiedCommunications Managerstarts
setting up the RTPconnections. The figure showshowRTPconnections are beingset up at the
M'fP (10.1.250.101). The trace output also shows howthc RTPconnections are being set up at
the H.323 gateway (10.1.250.102) by usingpaeketization of20msandthe codectype4 (G.71 I
mu-law). When the RIP connection is set up, the media transmission starts at MTP. The H.323
gateway RTP port number is 17236.
Now, the IP phone starts with the RTP connections setup process (StationD). The RTP
connections at thc IP phone 10.1.2.13 are setup by using the same parameters: packetizatiouof
20 ms and the codec type 4 (G.711 mu-law).
The figure also shows several identifiers that will be shared with the MTP device. When
troubleshooting, it is worth noting them, especially in high call volume environments, to locate
the proper debugging output on the hardware MTP side:
Partyld - 16777253 is the identifier that names a single leg of the MTP session. In this
case, it is the leg between MTP and the H.323 gateway.
Partyld - 16777254 is the leg between MTP and the IP phone (shown in the next figure).
ConferencelD - 16779226 identities uniquely the entire MTP session (both legs).
Troubleshooting Cisco Unified Communications (TVOICE} v8.Q 2010 Cisco Systems, Inc
Mm
MTP Allocation Trace (Cont.]
Stationlnit: (000 00021 openRecBiveOiannalAck Status0, IpAddr-IpAddr.type:C
ipAddr: CxDa01fa6560dlfb0Sa35f971g5018f35 (10.1.250. 1011 . Port.lB434.
PartylD" :;""-;" s 1, 100,49,1. 982" 10 .1. 2E0.101'HTP-HQl
StationD: (0000003) startMadiaTransmiSBion conferencaID-17183516
passThruPartyID-16777255 remoteIpAddresa.IpAddr.type:0
ipAddr:OxCaClfa650000000000000000O00O0000 110.1.250.101)
remoter ortNusiber-18 434 milliSacondPaclietSiie-20
compresBType-4 (HediaPayload G71 lUlw64)t) RPC2B32PayloadType-0
qualifiarOut=7. mylP: IpAddr. type: 0 lpv4Addt :0x0a01020d (10 .1. 2 .13)
1,100,49,1.982*10.1.250.101"MTP-HQI
Stationlnit; (0000003) OpenReceivaChannalAck Btatus-0, IpAddr.IpAddr. type: 0
ipAddr:0*Oa010'0d000000000000000000000000 (10.1.2.13}, Port-30334,
PartylD-167 7725 511, 10D,49,1.883*10.1.2.13'SEP0 024C4 454ADB
. . .truncated. . .
MediaTermlnationPointControl(2) i: star MediaExchangeStartTalking -
CI.16779226, DECPValue - 184; 1, 100, 4.1.983~10.1 .2.13"SEP0024C4454ADB
MediaTenninationPointContiol (2) ; :Btar StationOutputStartHediaTranamission -
ContorencelP; '":.?>., KediaPartyld: .ii>'"7" =, RemotelpAddr: 0xd02010a
(10.1.2.131 RemoteRtpPortHumber: 30334 msecPackatSlie: 20 compreBSionType:
4 1,100,49,1.9B3"10.1.2. 13"SEP00 24C44 54ADB
This trace output shows that the setup of RTP connections is completed. The RTP port number
at the MTP is 18434. The media transmission at the IP phone has started toward the MTP RTP
port 18434.
The media transmission stage has started at the MTP to the calling IP phone as well. The RTP
port numberof thc IPphone is 30334. Notethe same mediaparameters: packetization of 20 ms
and G.711 mu-law (compression type 4).
Note the leg and session identifiers in this figure.
MTPAllocation Debug
5-56
This section shows a debug of a successful MTP allocation from the perspective of the
hardware MTP.
Thedebugseepmessages command is used to generate thisdebug output, which shows the
SCCPmessages beingreceived and sent at the hardware MTP. The first message relatesto the
H.323 gateway leg. Note thesame identifiers as seen inthetrace output, inwhich thcidentifier
16777253 means thegateway. Cisco Unified Communications Manager requests theMTP to
set up a leg(open RTP channel) byusing a packetization period of 20msand compression type
4 (071 I mu-law). The MTPacknowledges with OpenReceiveChannelAck.
Thc bottomof the debug output relates to the IP phone. Cisco UnifiedCommunications
Manager requests the MTP toset upa leg(open RTP channel) toward the IPphone by using
the same media parameters.
mW
'Mm
MTP Allocation Debug (Cont.
SCCPircvd StartMediaTranBniisaion I H.32J gateway related
BtartMediaTransmiaaionMag Info:
conference id - 1S779226, passthroughpartyid - !>.: "iS ;
msec pktBiie = 20, comiression type - 4
ranwte ip addr . 10.1.250.102. remote port . 17236
qualifier out.precedents value - 184, quallfiarout.BBvalue *
qualifier out .majtf rames per pkt - 0, g723_bitrate - 0, call_
stream pass through id 16777216 rfc2833 payload type 0
co dec dynamic pay load 0, codecmode 0
Encryption Info i: algorithm!d 0. kiyleo Oaalt len 0
SCCP;send OpenReceiveChannelAck
SCCPircvd startHediaTransmiss ion / IP phone related
StarttredlaTranBmiSBiortMsg Info:
conferenceid 16779226, pass_ through partyid - \', "'?',!
msec pkt siia 20, ctn=prission_typt 4
remote ip addr 10.1.2.13, remote port 30334
qualifierout.precadence value - 184, gualifierout.bbvbIu* -
qualifierout .max frames per pkt 0, g723 bitrat* - 0, call_
atrsampaBathroughld 16777216 rfc2B33_payload type - 0
codecdynamic payload - 0, codacmoda 0
Bncryption Info ;: olgorithmid 0, key len Oaaltlan 0
This debug outputshowsthe start of mediatransmission. The upper sectionshowsthe media
transmission start at the leg towardthe H.323gateway. Note the remoteRTPport number
17236, which is the port number at the H.323 gateway 10.1.250.102.
The bottom section shows the media transmission start at the leg toward the IP phone. The IP
phone RTP port number is 30334.
>2010 Cisco Systems, Inc. Voice Quality and Media Resources Issues 6-57
Hardware MTP Allocated
6-58
This figure shows theMTP legs that have been set up.
fcif&M
Hardware MTP
HQ-Uahow accp connections
seas id conn id stype mode codec sport rport ripaddr
ntp sendrec v g711u 1S434 30334 . ' 1
mtp sendrec j gTllu 17062 17236 .='; ;'
Total number of active sessio (a) 1. and connection(s) 2
The show seep connections command displays the session and both its legs. The first row of
the command output shows ihelegtothcIPphone and thc second row shows the legtothe
H.323 gaieway. Note the meaning of identifiers. The M'fP has two connections (legs) butonly
a single MTP session. The identifiers, ihe same asthose inthe trace and debug output, can be
seen here as well.
Troubleshooting Cisco Unified Communications (TVOICE) vtf 0
mr
w
Summary
This topicsummarizes the key pointsthat were discussed in this lesson.
MTPmight be required to support H.323supplementary services,
H.323 outbound fast start, DTMFexchange ifother than NTE, and
mixed SIP environments.
Causes of MTPregistration issues include the Cisco IPVoice
Media Streaming Application service not running, MTPbeing
deactivated, or network connectivity issues.
The common causes for MTP allocation issues include no MTP
being registered, MTP runningout of resources, media resource
misconfiguration, or network connectivity issues.
In this lesson, you have learned to explainthe common issuesthat are relatedto MTPand
identify the most likely causes of these issues.
References
Cisco Unified Communications Manager System Guide, Release 8.0(2), Media
Termination Points at
http: \\\\u cisco com en US.partner.'docs'voice_ip_comm/cucm/admin'8_0_2/
ccmsys a05mtp.html
2010 Cisco Systems, Inc. Voice Quality and Media Resources Issues
6-60 Troubleshooting Cisco Unified Communications (TVOICE) v8 0 >2010 Cisco Systems. Inc
Lesson 3
Troubleshooting Issues with
Conferences
Overview
The conference bridge for Cisco Unified Communications Manager, asoftware orhardware
application, allows both Ad Hoc and Meet-Me voice conferencing. Each conference bridge can
host several simultaneous, multiparty conferences.
Both hardware and software conference bridges can beactive at the same time. Software and
hardware conference devices differ inthe number ofstreams and the types ofcodec that they
support.
Objectives
conferences and identify the most likely causes ofthese issues. This ability includes being able
to meet these objectives:
Review major characteristics ofAd Hoe and Meet-Me conferencing, review thc major
characteristics of software and hardware conference bridge implementations, and outline
general issues that can be experienced with conference bridges
Describe conference bridge registration issues, nonresponsive software issues, and explain
how to troubleshoot them
Describe the major issues that are experienced while using Ad Hoc conferencing and how
to troubleshoot them
Describe the major issues that are experienced while using Meet-Me conferencing and how
to troubleshoot them
Conferencing in Cisco Unified Communications
This topic reviews major characteristics ofAd Hoc and Meet-Me conferencing. It also reviews
the major characteristics ofsoftware and hardware conference bridge implementations and
outlines general issues that can be experienced with conference bridges.
3-62
tujfck.
CiscoUnified Communications Manager supportshardware and
software conference bridges
Thesoftware-based conference bridge supports only single-mode
conferences, using the G.711 codec.
Some Cisco Unified IPphoneshave a built-in conferencebridge
that can be used bythe barge feature only.
Software Conference
Bridge in Cisco Unified
Communications
Manager Server
"^S/ *t*7^J
Hardware
Conference Bridge
in Cisco IOS Router
PSTN
%/
Hardware Conference
Bridge in Swtch Chassis
| (Cisco Communication Media Module)
Cisco Unified Communications Manager supports hardware and software conference bridges.
The software-based conference bridge that is implemented as a Cisco Unified Communications
Manager Service supports only single-mode conferences by using a single codec (G.711).
Some hardware conference bridges can support multiple low-bit-ratc (LBR) stream types such
as G.729, Global System for Mobile Communications (GSM), orG.723- ihis capability
enables these hardware conference bridges to manage mixed-mode conferences. Ina mixed-
mode conference, the hardware conference bridge transcodes G.729, GSM, and G.723 streams
into G.71 I streams, mixes them, and then encodes the resulting stream into the appropriate
stream type for transmission back to the user. Some hardware conference bridges support only
G.711 conferences.
Meet Me and Ad Hoc Conferencing
Cisco Unified Communications Manager supports bothMeet-Me andAdHoc conferences.
Meet Me and Ad Hoc Conferencing
Meet-Me and Ad Hoc conferencing characteristics:
Meet-Me
Allocate directory numbers.
Manual distribution of meet me number.
- Access control by partitions and CSS.
Basic Ad Hoc
Conference originatorcontrolsthe conference.
- Originator can add and removeparticipants.
* Advanced Ad Hoc
Any participant can add and removeother participants.
- Multiple AdHoc conferences can be linked.
Meet-Me conferences allow userstodial intoa conference. Ad Hoc conferences allow the
conference controller to add specific participants to the conference.
Mcct-Me conferences require that a range ofdirectory numbers beallocated for exclusive use
of theconference. When a Meet-Me conference is set up, theconference controller chooses a
directory number and advertises it to members ofthe group. The users call the directory
number to join the conference. Anyone who calls the directory number while the conference is
active joins the conference.
Ad Hoc conferences comprise two types: basic and advanced. In basic Ad Hoc conferencing,
the originator ofthe conference acts asthe controller of the conference and isthe only
participant who canaddor remove otherparticipants.
Inadvanced Ad Hoc conferencing, any participant can add or remove other participants; that
capability is not limited tothe originator ofthe conference. Advanced Ad Hoc conferencing
also allows linking ofmultiple Ad Hoc conferences. Set the Advanced Ad Hoc Conference
Enabled elusterwide service parameter to True togainaccess toadvanced AdHoc
conferencing.
Linear Ad Hoc Conferencing
Advanced Ad Hoc conferencing allows linking multiple Ad Hoe conferences by adding one Ad
Hoc conference to another Ad Hoe conference as ifit would be an individual participant.
jnea
J
Conference
Bridge 1
Conference
Bridge 2
Conference
Bridge 3
No more thantwo Ad Hoc conferences can link directly toany
participating conference.
Avoid loops, otherwise participants in all conferences will
hear echoes
If attempting tolink multiple conferences when the Advanced Ad Hoc Conference Enabled
service parameter is set tnFalse, the IPphone displays anerror message. Use the methods that
arcavailable for adding individual participants toanAd Hoc conference toaddanother
conference to an Ad Hoc conference.
Invoke Ad Hoc conference linking for Session Initiation Protocol (SIP) phones only by using
the Conference and Transfer functions. The system does not support Direct Transfer and Join,
Supported Cisco Unified IPSIPphones comprise Cisco Unified IPPhone 7911 7941 7961
7970, and 797 I.
The participants inlinked conferences can all hear and talk with one another, but the
conferences are not merged into asingle conference. The conference list (ConlTist) softkey
displays an added conference as Conference and does not display the individual participants in
the added conference. Each participant can sec only the individual participants in their own
conference bridge.
In linear Ad Hoc conference linking, no more than two Ad Hoc conferences can link directly to
any participating conference. With linearconference linking, no limitation exists to the number
of Ad Hoc conferences that can be added, as long as no more than two conferences link directly
to any one conference.
IfConference Bridge I links directly to Conference Bridge 3, it results in a looped conference.
Looped conlercnces donot add any functionality avoid them because participants inall the
conferences can hear echoes.
Nonlinear Ad Hoc Conferencing
When three ormore Ad Hoc conferences link directly to another conference, nonlinear linking
results.
Nonlinear Ad Hoc Conferencing
Three or more Ad Hoc conferences linkdirectly to another
conference.
Anegative impact onnonlinear conferences istherelease of
resources.
Conference Conference
Bridge 1 Bridge 2
Conference Conference
Bridge 4 Bridge 5
1
Conference
Bridge 3
Conference
Bridge 6
The system does not permit this type oflinking by default because potentially negative impact
onconference resources exists. To enable nonlinear conference linking, setthe Nonlinear Ad
Hoc Conference Linking Enabled clusterwide service parameter toTrue. Nonlinear Ad Hoc
conference linking will not work unless both the Nonlinear Ad Hoc Conference Linking
Enabled andAdvanced AdHocConference Enabled service parameters areset toTrue. Access
the Nonlinear Ad Hoc Conference Linking Enabled service parameter in the Advanced view of
the Service Parameters Configurationwindow.
When conferences arelinked innonlinear fashion, theconference resources might notbe
released when all Teal participants have dropped out ofthe conference, which leaves the
conference bridges that are connected toeach other when no one isusing them. This can
happen because each conference only recognizes the participants that connect directly to its
own conference bridge. They cannot detect when all the real participants in the other
conferences have dropped out. Toreduce the risk of tying upunused conference resources,
restart conference bridges more frequently when the Nonlinear Ad Hoc Conference Linking
Enabled service parameter is set to True.
Hardware and Software Conference Media Resource
The figure summarizes implementation differences between software and hardware conference
media resources.
Software conference media resources are provided by Cisco Unified Communications Manager
servers. They only support the Ci.711 (and wideband) audio codec. Ifa software conference
bridge should be used for other codecs, transcodcrs are required. In amultisite Cisco Unified
Communications Manager deployment with centralized call processing, all software conference
bridges are centralized. Only when using multiple Cisco Unified Communicalions Manager
clusters or choosing the clustering over the WAN deployment model, software conference
bridges can bedistributed tomultiple sites.
To use software conference bridges, activate the Cisco IP Voice Media Streaming Application
service, verify that the Run Flag service parameter is set to True, and change the Call Count
service parameter ifdesired. In addition, verify the configuration ofthe media resource in Cisco
Unitied Communications Manager.
When implementing hardware conference media resources, you can selectively enable and
disable multiple audio codecs so you arc not limited to using G.71 I only. As hardware
conference media resources are provided by Cisco IOS gateways, they can be deployed in a
distributed fashion: you can have a local hardware conference atany site.
Configure the media resource at the Cisco IOS device and set the permitted codecs and the
maximum number ofsessions. Add the media resource in Cisco Unified Communications
Manager.
" General Issues Related to Conference Bridges
This is alist ofthe most common issues when you use the conferencing feature ofCisco
Unified Communications.
General Issues Related to Conference
Bridges
General conferencing issues:
Conference originator cannot addparticipants toAd Hoc
conference.
Ad Hoc conferences cannot be linked together.
Conference originator cannot set upa Meet-Me conference.
Conference participants cannot join a Meet-Me conference.
Conference participants are unexpectedly dropped off their
conference.
Regardless of the conference bridge type that is used, thc general conferencing issues are the
following:
Conference originator cannot add participants toAd Hoc conference.
Two or more Ad Hoc conferences cannot be linked.
Conference originator cannot setupa Meet-Mc conference.
Conference participants cannot joina Meet-Me conference.
Conference participants are unexpectedly dropped offtheir conference.
This lesson inspects each ofthese issues and names their most possible causes and remedies.
Troubleshooting Conference Bridge Registration
and Nonresponsive Software Issues
This topic describes the issues of conference bridge registration and nonresponsive software
and explains how to troubleshoot them.
6-68
Con nun
HQ-l#aJiow accp
SCCP Admin State: UP
TPv4 Address: 10.1.250.101
Port Numbec . 2 OQ0
IP Precedence. 5
Uaer Masked Codec list: None
Call Manager: 10.1.1.1. Port Number; 2C
Priority: N/A, Version:
Trustpoint: N/A
1.0, Identif lei
conferencing Cper state: ACTIVE - Cause Code; NONE
Active Call Manager: 10.1.1.1, Port Number: 2000
TCP Link status: CONNECTED, Profile Identifier: 1
Reported Max Streams: 16, Reported Max OOS streams: 0
Supported Codec: gTllulav, Maximum Packetiiation Period; 30
Supported Codec: gTllalaw, Maximum Packetization Period; 30
rldgt lith) I
Thc figure show, hou 10 verify That aconference bridge is registered with Cisco Unified
Ifahardware conference bridge is deployed, use the show seep command at the Cisco IOS
conference bridge, or choose Media Resources >Conference Bridge in Cisco Unified
The command output shows that thc Skinny Client Control Protocol (SCCP) application on the
Cisco IOS router is up. and the conference bridge isregistered with the Cisco Unified
Communications Manager at 1P address 10.1.1.1. The conference bridge operat.onal state is
ACTIVE, and the TCP link for SCCP is connected tothe Cisco Unified Communications
Manager.
The typical configuration of aCisco IOS conference bridge is like thc following. This
conference bridge can support thc conferencing legs that use thc G.711 and G.729 codecs:
voice-card 0
dspfarm
interface LoopbackO
ip address 10.1,250.10: 255.255.255.25b
seep ccm 10.1.1.1 identifier 1 version 7.
seep
2010 Cisco Systems
seep ccm group 1
associate profile 1 register CFB-HQ1
i
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
For ahardware conference bridge to register properly, its registration name that isconfigured in
theCisco IOSSoftware mustmatch thename that is configured at theCiscoUnified
Communications Manager. Older Cisco IOS versions might use the name that isbuilt from the
interface MAC address, like CTBmac_address.
Use the show seep ccm group command todisplay the name that isused for the hardware
conference bridge registration. This output shows that the name CFB-HQI isused as shown
registered in the figure.
HQ-lttshow seep ccm group 1
CCM Group Identifier; 1
Description: None
Binded Interface: None
Associated CCM Id: 1, Priority in this CCM Group: 1
Associated Profile: 1, Registration Name: m?)Wl
Registration Retries: 3, Registration Timeout: 10 sec
Keepalive Retries: 3, Keepalive Timeout: 30 sec
CCM Connect Retries: 3, CCM Connect Interval: 10 sec
Switchover Method: GRACEFUL, Switchback Method: GRACEFUL_GUARD
Switchback Interval: 10 sec, Switchback Timeout: 7200 sec
Signaling DSCP value: cs3, Audio DSCP value: ef
Ifthe software conference bridge isdeployed, it must beenabled (default setting) toregister
properly. Asoftware conference bridge can be enabled atthe Cisco IP Voice Media Streaming
Application service parameters page by setting the Run Flag in the CFB parameters section to
True. Another service parameter Call Count setsthenumber of resources that thesoftware
conference bridge can support.
For a software conference bridge, make sure thatthe Cisco IPVoice Media Streaming
Application service isrunning. You can verify this bynavigating toCisco Unified
Serviceability > Tools > ControlCenter - Feature Services. If theservice is reported as not
running, restart the Cisco IP Voice Media Streaming Application service.
Troubleshooting Ad Hoc Conferencing
This topic describes ihe major issues that are experienced while using Ad Hoc conferencing
*25fc *.(!,
Cisco IPVoice Media StreamingApplication
(G 711 only)
' "Y; .-. (mixed codecs)
This figure shows the setup ofathree-party basic Ad Hoc conference. Ihe headquarters IP
phone 2001 isa conference originator inthis figure. This IPphone isassociated with Media
Resource Group fist (MRGL) that refers to two media groups and, hence, two conference
resources: hardware conference resource CFBJiw, implemented ona Cisco IOS router and
software conference resource CFB_2 that is running ona Cisco Unified Communications
Manager server. For the headquarters and branch phones, regions arc configured that way when
acall stays within aregion. G.711 is used, but when a call is set up between headquarters and
the branch. GJ29 is used.
The conference originator places apoint-to-point call to the branch phone 3001, then presses,
Confm softkey, and dials 2002. Inthe meantime, 3001 is placed onnetwork hold. One more
pressing of the Confm softkey would set up a conference of three participants (3001 onhold
would be connected to the conference).
At this point, it is very important toknow which conference bridge the conference originator is
using for this conference. IfCFB2 is used, the branch phone would beunable lojoin the Ad
Hoc conference, because the software conference bridge can support G.711 only.
On the other side, the hardware conference bridge could beconfigured tosuppon mixed-codec
conferences (including G.729), and3001 would beable tojoin thisconference if such a
hardware conference bridge is used. If your MRGL has the MRG for the hardware conference
bridge configured (CFBhw) before the MRG that has thc software conference bridge
configured (CFB2), then thc conferencing will always use the CFBJiw first.
Note that thc conference participants do not need tobe associated with any media resources to
join the conference, only thc originator.
Another alternative tosolving thc issue of mixed codecs isto provide a transcoder that supports
thebranch phone andtranscodes from G.729 toG.71 1to join the G.711-only conference of
CFB 2.
70 Troubleshooting Cisco Unitied Communicalions(TVOICE) v8 0
1^^^-
Setting Up Ad Hoc Conference
HQ Region: G.711
CFBJiw (Mixed Codecs)
Location: CFB
| Point-to-Point Call
3001
Call Dropped Out
2002
Location: BS
BR Region: G.7?;:
This figure shows a situation that is like the previous one. Headquarters phones are within the
headquarters location, and the region allows G.711 within it.The branch phone has been
assigned with location branch and theregion dictates using G.729 for intersite calls. Locations-
basedCall Admission Control (CAC)has beenconfigured in CiscoUnifiedCommunications
Manager as follows:
The headquarters is unlimited.
The branch allows 72 kb/s.
CFB allows 16 kb/s.
The conference originator has been associated with the hardware conference bridge CFBJiw
that is able to mix variouscodecs, including G.711 and G.729, so all three phones wouldbe
able tojointhc conference that is supported bythis conference bridge. The conference bridge
CFB hw is assigned to thelocation CFB, citherdirectly at theconference bridge configuration
page, or through thedevice pool with which it hasbeenassociated.
Whenthe conference originator (2001)wants to invitethe twoconference participants, it sets
upa point-to-point call tothe first participant, forinstance the 3001 phone. TheCAC permits
thecall as long as at least 24 kb/sis available between thesites(theG.729 codec allocates static
24 kb/s from location). Then the conference originator uses the Confrn softkey to invite another
headquaners phone 2002, white 3001 is automatically placed onhold.
Now, the conference originator wantsto finishthe AdHoc conference setupand pressesthe
Confrnsoftkeyagainto bring3001 into the conference. But, the 3001 participant is suddenly
dropped out, and2001 and2002 have been connected ina point-to-point call. Thereason for
this behavior is the lack of bandwidth at the CFB location that allows up to 16 kb/s. CAC
blocks the calls between the conference bridge (part of the CFB location) and the phones.
Ad Hoc Conferencing Issues
This section lists the major issues of Ad Hoc conferencing and their possible causes.
Zffek&ZZ^ma, =f*J,fi&J,^ =mf.J *'?i&U&i'it&!r -=,
OC <-OMHTnCl
Major issues of Ad Hoc conferencing and (heir possible causes:
* Conference participants cannot be added by =.=<=..; <.,!.-:
No Confrn softkey on the phone.
If -No Conference Bridge" messageshown onphone, theoriginator phone
s notassociatedwith any conference bridge (MRGL), or itis not registered
or down.
Maximum numberof participants reached, check Maximum AdHoc
Conference service parameter.
Conference bridge cannot dothe codec ofthe participant being added.
CAC rejectsthe conference bndgeleg to a participant.
Conference bridge out of resources.
Cisco IPVoice Media Streaming Application servicenot running.
- Conference participants cannot be added by ,'>.-;.<'.n!. j
In addition, check if Advanced Ad Hoc has been enabled.
Conference participants can beadded either by anoriginator (basic Ad hoe) or by any
participant (advanced Ad hoc). If conference participants cannot beadded bya conference
originator, here arc possible reasons:
TheConfm softkey might not be seen onthe phone. TheConfrn softkey is located onthe
next screen that is accessible bypressing the More softkey at a Cisco IPphone. If it cannot
belocated onany of the screens, the softkey template that is used bythe phone might not
contain this softkey. Look at thcphone contiguration page tosee which softkey template is
used, and ensurethat the template contains the softkey.
If an IPphone shows themessage "NoConference Bridge," the conference originator
phone is not associated with any conference bridge through MRGL, or the conference
bridge is not registered, or. although rare, it might be down due to a software error. Make
sure that the MRGL contains the appropriate conference bridge and that the conference
bridge functions properly.
Each conference bridge supports a certain number of sessions andparticipants. The
maximum numbcr of participants could bereached andnoadditional participants arc
accepted. Check thc Cisco CallManager service parameter MaximumAd Hoc Conference
to see how many participants are supported per conference session. The value of Maximum
Ad Hoc Conference depends on the capabilities of the software or hardware conference
bridge.
Thcconference bridge might not support thecodec of theparticipant that is being added.
Remember that the software conference bridgesupports onlyG.711. Youcanjoin a
conference that is hosted by a software conference bridge that uses a codec other than
G.711 through a transcoder.
If bandwidth resources arc usedup, a CACmight reject theconference bridge leg toa
participant. Allocate more bandwidthresources or alleviate the existing usage.
Troubleshooting Cisco Unified Communications (TVOICE)v8 0
The conference bridge might be running out of resources. Deploy additional conference
bridge resources.
Cisco IP Voice Media Streaming Application service is needed for the software conference
bridge type. It acts as asoftware conference bridge. Ifthe service is not running, no
software conference resources are available.
Conference participants can be added by aparty other than aconference originator (controller)
only in case of an advanced Ad Hoc conference. By default, advanced Ad Hoc conference is
disabled.
For this functionality, enable the advanced Ad Hoc conferencing by setting the Cisco
CallManager service parameter Advanced Ad Hoc Conference Enabled to True, Advanced Ad
Hoc conference features include the ability for conference participants other than the
conference controller to add new participants to an existing Ad Hoc conference. Itcan also
drop other participants from the conference via the ConfList and RmLstC softkey.
Ad 1loc conferencing can also link Ad Hoc conferences by using features such as Conference,
Join Direct Transfer, and Transfer. To enable this parameter, you might have to restart the
conference bridges periodically to free all conference resources, because some resources might
not be freed even though the conference is terminated. For this reason, use this parameter with
extreme caution.
Thc Cisco CallManager service parameter Non-linear Ad Hoc Conference Linking Enabled
detennines whether more than two Ad Hoc conferences can be linked directly to an Ad Hoc
conference in anonlinear fashion. For this parameter towork, the Advanced Ad Hoc
Conference Enabled service parameter mustbe set toTrue.
6-74
ui Hoc conferencing Issues (Cont.)
Conference established, but suddenly dropped.
Service parameter Drop Ad Hoc Conference canforce the
conference termination.
If Advanced Ad Hoc enabled, this may happen when
periodically releasing a conference bridge.
Network connectivity to Cisco Unified Communications
Manager or to conference bridge tost.
Cisco IP\feice Media Streaming Application orsoftware
conference bridge is not responding.
Conference linking does not work.
Service parameter Nonlinear Ad Hoc Conference Linking
Enabled must be set to True.
The list of issues continues. Aconference being successfully established but being suddenly
dropped can also be experienced. This, however, can be the anticipated behavior.
The Cisco CallManager service parameter Drop Ad Hoc Conference determines how an Ad
Hoc conference terminates. Thc parameter can be set to the following values:
Never: The conference remains active a) after the conference controller hangs, and b) after
all on-net parties hang up. Be aware that choosing this option means that ifon-net parties
conference in off-net parties and then disconnect, thc conference stays active between the
oft-net parties, which could result inpotential toll fraud.
When Conference Controller Leaves: Terminate the conference when the conference
controller hangs up or when the conference controller transfers, redirects, or parks the
conference call and thc retrieving party hangs up.
When No On-Net Parties Remain in theConference: Terminate thc conference when
there arc noon-net parties remaining in theconference.
Aconference can suddenly drop ifAdvanced Ad Hoc conferencing is enabled. As mentioned
earlier, this type of conference is associated with the risk of aconference bridge periodically
releasing all conference resources.
Aconference will drop also ifnetwork connectivity to the Cisco Unified Communications
Manager or to aconference bridge is lost. Make sure that the IP connectivity is maintained and
ifDomain Name System (DNS) is used, that name resolution works as expected.
The Cisco IP Voice Media Streaming Application service includes the software conference
bridge. Although extremely rare, asoftware failure can occur and the process can stop
responding. Restart the service.
Troubleshooting Cisco Unified Communications (TVOICE) v8i
You might encounter vet another type of issue when conference linking does not work. As
menttoned earlier, this is possible only with advanced Ad Hoc conferencing. Ensure that the
Cisco CallManager service parameter Non-linear Ad Hoc Conference Linking Enabled is set to
True This parameter detennines whether more than two Ad Hoc conferences can be linked
directly to an Ad Hoc conference in anonlinear fashion. Nonlinear conference linking occurs
when three or more Ad Hoc conferences are linked directly to another Ad Hoc conference.
I mear conferencing linking occurs when one or two Ad Hoc conferences are linked directly to
another Ad Hoc conference. For this parameter to work, the Advanced Ad Hoc Conference
Enabled seniee parameter must be set to True. Valid values specify True (allow nonlinear
conference linking so that three or more Ad Hoc conferences can be linked to asingle other
conference) or False (do not allow nonlinear conference linking; only linear linking is
permitted).
Note Keep this parameter set to the default value unless aCisco support engineer instructs
otherwise.
Caution Enabling this parameter might require you to restart the conference bridges periodically to
free all conference resources, because some resources might not be freed, even though the
conference is terminated. _
VoiceQuality and Media Resources Issues 6-75
Verify Conference Bridge Performance
This figure shows how to verify the performance of conference bridges, namely, to ensure that
they have adequate resources.
3-.10*.
(
L3** ,
^ " =>'
1C'F-iTBiiT'lAJ.Irifl
- H*Sfc.*AtW
The Cisco Lmficd Real-Time Monitoring Toot (RTMT) provides performance counters for
both hardware and software conference bridges. The upper-left graphs show the real-time
statistics for thc hardware conference bridge CFB-IIQ1 that can support amaximum of 16
participants (two conferences with up to eight participants each). Three participants are
currently being served in a conference.
The two remaining graphs show the capacity and the current utilization ofthe software
conference bridge.
Troubleshooting Cisco Unified Communicalions (TVOICE] v8 0
Tracing Ad Hoc Conference Setup
This section describes the trace output of asuccessful basic Ad iloc conference setup. Only the
relevant events are shown and the rest are excluded tosave space.
Tracing Ad Hoc Conference Setop
Station
callRef
IpAddr
(0000011) SoftKeyEvent softKayEvent-2 (NewCall) linelt
.0.. 1,100,49,1.2385 "10.1.3.12'SEP0024C4454ADB
Ststionlnit-. (0000011) BnblocCall calledParty-3001. 1
1,10 0,49,1-23 SB"10.1.2.13-SEF0024C4454ADB
S;^lo"f'i0000013) 0H*. 11.100.4,.1.23,2-10.1.4.12'SBP0024C44SS561
,;[^ln"d"(OOD00111 OpenKeceiveChaanelAC* St.tus-0. IpAddr-IpAddt. type, 0
oiosocoooooooooooooooooooooooouc. 1.2.121, Port-mso,
r . j,*- 1, 100.49.1.23S3-10.1.2.12"SEPO024C4454ADB
StationD- (0000012) startHediaTransiDiBaion coofBrencelD-lTBSBBTS
BBBThrUPrtylD=1677T28B remotelpftddresa-lpAddr. type;0
ipAddr-0x0801030=00 0000000000000000000000(10.1.2.131 reBOtePortNumber-19180
oilll3ecopdP.cH-tSi.s-20 conprBBBType-6 (Media PaylOBd_G732 64k)
nnr2tmpavloadTyps-0 quali fierOut-T . mylP. IpAddr .type:0
"^Si".ii'..u.i ,i'io-"'i-239rio-i-2a2"rrLT5^ o
Stationlnit- (0000012) op.nlWC.lv-Cto.lAck Bt.tul-0, IpAddr,IPAddT. typ.-O
ipAddr-0n00104OcOOOOOOOOOOO0000000O0O00O 110.1.4.12), Port-23266,
rf, -,-. <,.; l 100,49, 1.2394-10.1. 4.12'3BP0024C44555 61
StationD- (0000011) st.rtHedi.TrsnBini.sion coQfer.nCeiP-1785BB75
paaBThruPartylD. 16777287 r.n=ot.iIpAddreS8-IpAddr.typ. =0
IpMdr, 0x0.01040=000000000000000000000000(10.1.4.12) reitePortNUii1b.2 326
millisecondPacket
e.O
Three phones will participate in the conference: the headquarters phones with IP addresses
10.1.2.12 (directory number 2001)and 10.1.2.11 (directory number 2002) and the branch
phone with the IP address 10.1.4.12 (directory number 3001).
The upper part of the output shows the headquarters phone 2001 (SEP0024C4454AD8) calling
the branch phone 3001 (SEP0024C4455561). The branch phone 3001 answers the call. The
bottom part of the output shows how Real-Time Transport Protocol (RTP) media is established,
and when done, the transmission starts on this point-to-point call.
Voice Quality andMedia Resources Issues 6-77
6-78
: racma
Station Ir
callRefei
. . .[run ci
StationD:
ipv4Addr:
Stations :
ipv4Addr:
S t a t i on D :
;pv4Addr:
StationD:
ipv4Addr=
StationD:
callRef en
ated
Tit :
ated
Stationlr
callRef ei
(0000011) SoftKeyEvent EoftKeyEvent.13(Confrn, linelnatance-1
:e=l7858875. 1,100,49,1.2401*10.1.2.12-SEPC024C4454ADB
(0000011) Cla.eB.Calv.Ch.nn.l confeCMfeID,,7B5a8,5
. mylP: IpAddr.type:0
01020C (10.1.2.12) ILIDO.49,1.2401-10.1.2.12-BBP0024C44S4AOB
10000011) S,Fx.d1.Trnl..loo conference!^ 17B5 B875
. mylP; IpAddr.type:0
"-So^r^^^.i^^L^nn^c^.:^^^4-"^
mylP: IpAddr.type;0
.01040=110.1.4.12) :l,iOD.4.l.al-l0.1.2.12-SW0024C4454ADB
10000012) StopMediaTtan3mia3ion conferencbid,!7B58876
-.-. mylPi IpAddr.type:0
.01040c 110.1.4.12) 1,100,49, 1.2401-10.1.2.12-SW0O24C44S4ADB
(0000011) DialedNumber dial edMumber^ 2002 linelnstance-1
B=17B5B87B. 1.100, 49, 1,2407'IO. 1.2. 12-SEPO02 4C44S4AD8
(0000003> OftHook.1.100,49,1.2410-10.1.2.11-8BP002497A90D33
(0000011) SoftKeyBvent softKeyEvent, 13 (Confm) linelnetance-1
,-.7858878. 1,100, 49. 1.2415-10. 1.2. 12 "S EP002 4C44 S4AD8
\ nm tP ' 'S cstabhshcd- ^ C'f ftkCy is pressed a. thc headquarters
phone 2001. The poim-to-poim RTP media is closed, and the branch phone 1001 is
automatically placed on hold (ihe MOII allocation is excluded from .he trace).
Ihe headquarters phone 2001 then dials the headquarters phone 2002, and the 2002 answers the
call. The point-to-point call RTP media from 2001 to 2C.02 is set up (it was eliminated from the
traceoutput because it is like 2001-3001).
Now. the user al the headquarters phone 2001 presses the Confrn softkey again to brine the
three parties into the conference.
Tracing Ad Hoc Conference Sei
KRH: LConvcrtScmStringToStdSttiog CFB mrg]
. :: i. - : ;: : 2 ;;'sbpo024C4454AD8
MRU; :gt>tUcbDeviceGivtnMxgl DeviceName-CFB
Capability,.! HultiCaat.O MRGL. I : ,1'iS r- , I M : I s
BType50 Qroup=0 Counter
.].!..;"SEP0 024C4454ADB
. tru Ited.
MsdiaBsaourceCdpc(7}:tuaieing HrmAllocateUcbReBourceReq -
CI.17858880 1,100,4 9,1.2415'10.1.2.12'SEP0024C4454AD8
SMDMSharedData::findLoealDevice - Name-CFB 2 Key-3494336d-bal9-4f82-bea2-
f21d9a04ecf isActive-1 Fid-11,42,1) found]
1. 100,4 9,1.2415"10.1.2.12'SEP0024C445 4ADB
KediaResourceCdpc (7 I s : BendUcbAllocateRequeBtToDovice UcbReflource-CFB 2
. . .truncated. . .
Unicast Br idgeControl; :getAvailableBrldge RequestedStreamO
Current StreamsA vail ablec48
MaxStraamParConf.O 1.100.49,1.2415'10.1.2 .12'SEP002 4C44S4AD8
ited.
OnicaBtBridgeControl: j-haadlaSetupSuccass - Device name
17 85888 1 1, 100, 49, 1. 2415 '10. 1.2. 12"SBP0O2 4C445 4ADS
. . .truncated. . .
StationD; (0000011! CloeBRecaivaChannel conferencID.17B5887B
paasThruPartyID.167772B9. mylPi IpAddr.typei0
ipv4Addr:0x0aO1020c(10.1.2.12) |1,100,49,1.2415"10.1.2.12'SEP0024C4454AD8
StationD: (0000003) CloseReceiveChannel coaterencelD17B58879
pa ssThruPartyID=i 167 77290 . mylP : IpAddr . type :0
ipv4Kddr;00aO1020b(10.1.2.II) |1,100,49,1.2415'10.1.2.12'SBP0024C4454AP8
CFB party Ci
When the Confrn softkey is pressed at 2001, Media Resource Manager (MRM) looks up the
MRGI. at the 2001 phone. The MRGL is found (the name is not shown in the output, only its
descriptor) as pointing to the Media Resource Group (MRG) that is called CFB mrg.
The MRG CFBmrg lists the software unicast conference bridge (Ucb) that is named CFB_2.
MRM attempts to allocate this conference bridge resource. The CFB_2 has been registered with
Cisco Unified Communication Manager (isActive=l). The allocation request is originated
toward CFB_2.
The conference bridge CFB_2 supports a maximum of 48 streams, and three streams are being
requesied for this conference. CFB 2 is successfully allocated. The point-to-point RTP media
between 2001 and 2002 are removed.
The stop media transmission event for these two phones was removed from the trace output to
save space.
2010Cisco Systems, Inc. Voice Quality and Media Resources Issues 6-79
:IW
' > . . '. OpenReceiveChannel conferenceID=17 858876
paBaThruPartyID = l7 772 92 milliaecondPacketSize=2 0
compressionType.4 (Media Payload G71iUl8w6 4M RFC2833PayloadType=0
qualifierln-? sou reelpAddr-lpAddr.type:0
ipAddr:OxOaOlOl0100000000000000000000 0000 (10.1 .1. 1) . mylP; IpAddr.types0
ipv4Addr:OxOa0104 0c(10.1.4.12) !1,100.4 9,1.2 415"10.1.:.12'SEP0024C445 4ADB
. . - - < . - sta tionOutputOpenReceiveChannel TCPPid = 11.100. 9.5]
mylP: IC 1010a (10 . 1.1.1) ConferencelD: 1677 9 222 MediaPartyId: 16777293,
ecPacketSiiei 20, couple ssionType: 4, security: O 1,100.211,49.1****
OpenUeceiveChannel conferences. 17 S588 79
passThruPartyID=167 772 94 mi llisecondPac)ietSiie=2 0
co=EpressionType>4 (Media Payload G71iulaw=j 41t) RFC2 833PayloadType=0
qualifierln*? sou reelpAddrolpAddr.type:0
ipAddri0x0a 010101000000000000000 000000000(10.l.l.l) . myIP: IpAddr.type:0
ipv4Addr:0*Oa01Q20b (10.1.2.11) 1.100,49,1.2415'10.1.2 .12'SEP0024C4454AD8
' .-- .-- - - sta UonOutputOpenReceiveChannel TCPPid [1.100.9.5]
mylP: 101010a (10.1.1.1) ConferencelD: 16779222 MediaPartyId: 16777295,
ecPacixetSiie, 20, compre esior.Type; 4, security: 0 \ 1, 100, 211. 50 .1 '***
- :.. .. OpenReceiveChannel conierenceID=17B58875
ssTbruFar tylD =167 7 72 96 milliaacondPac>.etSiie-2 0
ETpregaionType =4 (Media Payload Q713Ulaw=; 4k.) RFC2 833PayloadType=0
qualifierln-? sou reelpAddr*IpAddr.type:0
ipAddr:0x0a010101000000000000000000000000(10.1.1.1). mylP: IpAddr.type:0
ipv4Addr:OxOa01020c(30.1.2.32) ,1,100,4 9,1.2415'10.1.2.12'SEP0024C445 4AD8
'I his trace output shows how Cisco Unified Communications Manager requests the setup of
RTP media for thc conferencing legs:
Setup of the RTP media between the conference bridge CFB 2 and the branch phone
10.1.4.12 by using thc codec G.7I I
Setup of the RTP media between [he conference bridge CFB 2 and the headquarters phone
10.1.2.11 by using ihe codec G.711
Setup of the RIP media between the conference bridge CFB 2 and the headquarters phone
10.1.2.12 bv using the codec 0.71 I
6-S0 Troubleshooting Cisco Unified Communications (TVOICE) vB0 2010 Cisco Systems. Inc
Tracing Ad Hoc Conference Setup
*-.'. .','''-'": OpenReceivBChannslAck Statua-0, IpAddr.IpAddr. type=.0
ipAddr: 0x0s010101000000000000000000000000(10.1.1.1) , Port-24610,
PartyID>167 77 2 9 1! 1. 100,4 9,1, 2416 '10 .1.1.1 'CrH 2
-. = - .. ~,Z',"''. : startHsdiiTransDiBsiori conferancelD.1785BS76
piiaThruPr tylD-lri? 77292 remota IpAddrao s-IpAddr . type =0
ipAddr: 0x03010101000000000000000000000000 110.1.1.1) remotePortNumber-24 610
.. .truncated. ,.
ipv4Addr;0x0s01040c (10.1.4.12) 11,100,49,1. 2 416" 10 .1.1.1 *CFB 2
- - !-:.- .:;-;;. OpanRac aivaChannelAc It Status.0. IpAddr.IpAddr .typa: 0
ipAddr:OxOaOlOl01000000000000000000000000 110.1.1.1) , Port=2 4612,
PartylD-16777293|1,100,49,1.2417"10.1.1.1"CFB 2
'-.---.-?. . ( ;o -'''': l ; startMadiaTransaiiBBion conference ID-17 8 58 87 9
passThruPartylD-16777294 remoteIpAddreas-IpAddr.type:0
ipAddr:OxOaO1010100 00000 0000 0000 000000000110.1.1.1) ramotePorbNumber-24 612
.. .truncated. ..
ipv4 Addr :0x0a01 020b (10.1.2.11) 1,100, 4 9,1. 2 417" 10 .1.1.1'CFB 2
-':-.- ; ".;'",( i Opa nBeceive Channel Ac k StatusaO, I pAddn IpAddr. type:0
ipAddr:0x0a0101010000000000OO0OD0OO0OOO0O(10.1.1.1). Fort-2 4614,
PartyID-167 7729 5i1,100,4 9.1.2418"10.1.1.1"CFB 2
<:-: !,:: :-:'!";: startMeaiaTransmission eonfarencelD. 17858875
paaaThruPartyID-167 77296 remoteIpAddraBs-IpAddr.type:0
IpAddr:OxOaO10101000000000000000000000000 110.1.1.1) remote PortNumber.24614
ipv4Addr:0x0a01020c 110.1.2.12) |1,10D.4 9.1.2 41B"10.1.1.1"CFB_2
The conference bridge CFB_2 acknowledges the requests and offers the RTP ports to IP
phones. IP phones open their RTP media for transmission toward CFB 2 by using thc ports that
are provided.
The trace output on the figure continues with the following events that open up the RTP
channels for the transmission at the CFB_2:
The branch phone 10.1.4.12 acknowledges the request to open the RTP channel and
provides the RTP port number.
Stationlnit: (0000012) OpenReceiveChannelAck Status=0,
IpAddr=IpAddr.type:0
ipAddr:0x0a0104OcOO00000000 00000000000 000(10.1.4.12), Port=24656,
PartyID=16 7772 92|1,100,49,1.2419*10.1.4.12"SEPQ024C4455561
Thc CFB opens the RTP channel toward the branch phone 10.1.4.12 and starts the media
transmission.
UnicastBridgeControl - stationOutputStartHediaTransmission TCPPid =
[1.100.9.5] mylP: 101010a (10.1.1.1), security:
0.[1,100,49,1,2419*10.1.4.12*SEP0024C4455561
UnicastBridgeControl - star_StationOutputStartMediaTransmission -
ConferencelD: 16779222 MediaPartyld: 16777291, RemotelpAddr: c04010a
(10.1.4.12) RemoteRtpPortNumber: 6050 msecPacketSize: 20
compressionType: 4, TCPPid =
[1.100.9.5]|1,100,49,1.2419*10.1.4.12*SEP0024C44 55561
The headquarters phone 10.1.2.11 acknowledges the request to open the RTP channel and
ipAddr:0x0a01020b000000000000000000000000(10.1.2.11) , Port =20782,
PartyID=16777294|1,100,49,1.2421*10.1.2.11*SEP0024 97A90D32
) 2010 Cisco Systems, Inc Voice Quality and Media Resources Issues 6-81
The CFB opens thc RTP channel toward the headquarters phone 10,1.2.11 and starts the
media transmission.
UnicastBridgeControl - stationOutputStartMediaTransmission TCPPid =
[1.100.9.5; mylP: 101010a (10.1.1.1), security:
0. I 1,100,49,1.2421*10.1.2.11*SEP002497A90D32
ConferencelD: 16779222 MediaPartyld: 16777293, RemotelpAddr: b02010a
(10,1.2,11) RemoteRtpPortNumber: 512E msecPacketSize: 20
[1, 100. 9. 5] I 1,100,49,1,2421*10.1.2.11*SEP0024 97A90D32
The headquarters phone 10.1.2.12 acknowledges the request to open the RTP channel and
Stationlnit: SOOOOOlli OpenReceiveChannelAck Status=0,
IpAddr=IpAddr.type;0
ipAddr:0x0a01020c0O000QQ000C0OOO00000OOO0(10.1.2.12), Port-20604,
PartyID=167 772 96|1,100,49,1.2427*10,1.2.12*SEP0024C44 54AD8
The CFB opens thc RTP channel toward the headquarters phone 10.1.2.12 and starts the
media transmission.
UnicastBridgeControl - stationOutputStartMediaTransmission TCPPid -
[1.100.9.5] mylP: 101010a (10.1.1.1), security:
0. |1,100,49,1.2427*10.1.2.12"SEP0024C4454AD8
110.1 .2 .12) RemoteRtpPort
6-82 Troubleshooting Cisco Unified Communications (TVOICE) vB.O 2010 Cisco Systems, Ir.c
Troubleshooting Meet-Me Conferencing
This topic describes the major issues that are experienced while using Meet-Me conferencing
letting Meet-Me Conference
Meet-Me: 2Y.
F..HG
C:-B
HQ Region: G.711
CFBJiw (G.711)
NewCall: 2100
NewCall. 2100
2002
3001
3002 3003
Location: BR
BR Region: G.729
This figure shows howa Meet-Me conference is being set up, and it outlines two reasons why
the branch phone 3001 cannot join the established conference.
The Meet-Me pattem 2100 was configured at Cisco UnifiedCommunications Manager. The
headquarters phones and the conference bridge are assigned to thc headquarters location with
unlimited bandwidth and the headquarters region with the G.711 codec. The branch phones are
assigned to the location branch that allows two G.729 calls (2 * 24 kb/s - 48 kb/s) and the
region branch with the G.729 codec.
The hardware conference bridge can performthe G.711 codec only, and it has been properly
registered with Cisco Unified Communications Manager.
The conference originator, the headquarters phone 2001, by using the Meet-Me softkey, dials
thc Mect-Me conference pattem and establishes thc conference. Another headquarters phone,
directory number 2002. dials into the established conference.
Now, branch phone 3001 wants to join the conference. The user dials 2100, but the reorder tone
is heard. The reorder tone is heard because the conference bridge does not support the G.729
codec that is enforced between the headquarters and the branch regions.
The branch phone can still establish a point-to-point call with any of the headquarters IP
phones, because Cisco Unified IP phones support both G.71 1 and G.729 codecs by default.
You can deploy a transcoder at the headquarters to support branch phones to solve this
particular codec issue.
This figure also illustrates an issue that is caused by the locations. If, in or out of the branch
site, two G.729 calls are already set up, the third call, and it also includes the calls to join thc
conference, would fail because of a CAC rejection. The branch region allows maximally 48
kb/s that have been consumed by the two earlier G.729 calls.
Notethat onlythe conference originator is associated with the conference bridge that is using
the MRGL and the MRG. The conference participants use the conference resource that is
determined by thc conference originator.
6-84 Troubleshooting Cisco UnifiedCommunications (TVOICE] v8.i
Meet-Me Conferencing Issues
This section lists the major issues and their possible causes when using a Meet-Me
conferencing feature.
Meet-Me Conferencing Issues
Major issues of Meet-Me conferencing and their possible causes:
Meet-Me softkey unavailable:
Possibly excluded from Softkey template
Meet-Me pattem cannot be dialed lo pin a conference (New Call), reorder lone
heard:
Meet-Me conference has not yet been set up by a conference originator or
Meet-Me pattern is in partition not accessible by the caller CSS.
CAC rejects the call.
Conference bridge does not support a codec of the participant phone or running
out of resources.
When setting up a conference using Meet-Me softkey, "No Conference Bridge"
displays and reorder tone is heard.
Not associated with any conference bridge, media resource misconfigured.
Conference bridge not regstered or dcwn or out of resources.
The Meet-Me softkey is part of the phone off-hook state. It is located on the phone screen that
is accessible by using the More softkey. The Meet-Me softkey is part of the Cisco Unified
Communications Manager preeonfigured softkey templates (for instance, the Standard Feature
and Standard User). If your phone uses a customized softkey template, make sure that the
Meet-Me softkey appears among the selected softkeys of the off-hook state.
If the Meet-Me pattern cannot be dialed as a new call to join a conference, and a caller gets a
reorder tone, the most likely cause is that the Meet-Me conference is not yet set up. This is
normal behavior. The Meet-Me conference must be set up by a conference originator before
participants can dial in. The conference originator takes the phone off-hook, presses the Meet-
Me softkey (gets a ZipZip tone to confirm), and enters the Meet-Me pattern (for instance,
2100). If the pattern is valid and the conference resources are available, the phone displays "To
Conference (2100)."
Another possible cause could be that the CAC is rejecting the call to the Meet-Me pattern at
either the phone or the conference bridge location. If so, the "Not Enough Bandwidth" message
displays at the calling phone.
The joining call will also fail if thc conference bridge does not support the caller codec or if the
conference bridge resources are exhausted.
The Cisco CallManager Maximum MeetMe Conference Unicast service parameter specifies the
maximum number of participants that are allowed in a single unicast Meet-Me conference that
is hosted by a software conference bridge. When a conference is created, the system
automatically reserves a minimum of three streams, so specifying a value less than three allows
a maximum of three participants.
)2010 Cisco Systems. Inc. Voice Quality and Media Resources Issues 6-85
When a conference originator wants to set up a Meet-Me conference and dials thc Mcet-Mc
pattem by using the Meet-Me softkey, thc message"No Conference Bridge" displays on the
phone and the reorder tone is heard. These are the most likely causes:
Thc phone is not associated with any conference bridge or MRG, or the MRGL might be
misconfigured.
The phone is associated with a conference bridge, but the conference bridge is not
registered or it is down. This bridge failure can occur if the conference bridge is
misconfigured, if network connectivity issues exist, or if the DNS does not resolve the
names (if names instead oflP addresses arc used).
Make sure that an appropriate conference bridge is associated with the Meet-Me conference
originator phone and that the conference bridge is registered and available. The Cisco IP Voice
Media Streaming Application service includes the software conference bridge.
leet-Me Conferencing issues {C<
When setting up a conference using Meet-Me softkey,
reorder tone is heard:
- Meet-Me conference already in progress, set up by
another originator.
- Meet-Me pattern in partition not in CSS of the originator
phone (reorder tone or annunciator).
CAC rejects the call.
Conference bridge does not support a codec of originator
phone or running out of resources.
Auser cannot join a conference using Meet-Me softkey:
- This is normal, join the conference by standard dialing.
A conference originator attempts to set up a new Meet-Me conference. However, as soon as the
Meet-Me pattern is dialed by using the Mect-Me softkey, the reorder tone is heard. This issue is
most probably caused by one of the following:
The Meet-Me pattern has already been used, a Meet-Me conference is in progress at this
pattern, set up by another originator. To minimize thc probability of this occurrence,
provide a range of Meet-Me patterns.
If a reorder tone or annunciator message is heard, thc Mcct-Me pattern that is being dialed
is inaccessible from the calling search space (CSS) that is associated with the conference
originator phone.
The CAC might reject the call to the Meet-Me pattern at either the phone or the conference
bridge location. If this is the cause, the "Not Enough Bandwidth" message displays at the
calling phone.
The call that is attempting to set up the Meet-Me conference will also fail if the conference
bridge does not support the conference originator codec or if the conference bridge
resources are exhausted.
If a user attempts to join an existing conference in progress by using the Mect-Me softkey, thc
reorder tone is heard. This is normal behavior. The conference in progress is joined by the
standard dialing of the Meet-Me pattem, not by using the Meet-Me softkey.
2010 Cisco Systems, Inc. Voice Quality and Media Resources Issues
Tracing a Meet-Me Conference Setup
This section describes the trace output that is taken during a Mcet-Mc conference setup with
three participants.
&& ****&**&*-****- if*
riicir
Stationlnit: 10000001) SoftKeyEvent softKeyEvent 2 (NewCall) llneinstance=0
cailEeference-O. 1, 100 49,1.445*10.1.2.12"SEP0024C4454BD8
Stationlnit : (000 0001 ) SoftKeyEvent softKeyEvent 18 (MeetMe) linelnat nce = l
callReferenceiG2:7819 . 1,100,49,1.448*10.1.2.12 SEPO024C4454AD8
.. . truncated. ..
Stations! (0000001 ) DialedNumber dialedNumber 2100 linel stances. 1
callflefetence.lB22T819 1,100.49,1.452*10.1.2.12 SEP0024C4454ADB
MRM:jconvartScmStringTQStdString hwCFB mrtj i. .. : . . '- "
MRM: :getUcbDevi ceGivenMigl DeviceNatne=CFB-HQl DsviceType=52 Oroup-0
Counter=0 Capability=1 MultiCaat.O
.. .truncated. ..
MediaResojiceCapc [41:: se r.dUnBAl locate RequestToDe Jice UcbRes urce=tCFB HQ1
Cepn=>2d 3 f 47 84-8 10=?- Obf d - ddeb -
47535E621daf 1, 100, 49, 1.452'10.1.2.12 *SEP0024C=H54AD8
UnicastBridgeControl: : getAvailableBridge Requestedstream=1
Currents tie ansAvailabl e=48 NaxStreamPerconf-8
1, 100,49,1.452'10.1.2. 12 'SEP0024C44S4AD8
. . .truncated.. .
UnicaHtBridgsCantrol:: hsnaieSetupSuceess - Device Name * cfb-hqi, Party CI
18227822 1,100,118,3.1 *
Three phones, two at the headquarters site and one at the branchsite, participate in the Mect-
Me conference at thc Meet-Me pattern 2100. The conference is hosted by the hardware
conference bridge that is called CFB-IIQ1, which is an enhanced conference bridge that runs on
Cisco IOS Software.
At the beginning of the trace, the conference originator phone (SEP0024C4454AD8) with the
IP address 10.1.2.12 dials the Mect-Me pattern 2100. The MRM checks if the phone is
associated to any conference bridge via the MRGL. Thc MRGL is found (only its descriptor is
shown in the output) as pointing to thc MRG hwCFBjnrg that lists theCFB-IIQl conference
bridge. The MRM successfully allocates the conference bridge that can support up to 48
streams.
Trace output for the setup of RTP media between the conference bridge and the IP phone that
uses the G.711 codec is shown here:
StationD: (0000001) SEP0024C4454AD8 ,
star_MediaExchangeAgenaOpenLogicalChannel packetSize=20, codec=4,
ci=18227819|l,100,211,13.1****
UnicastBridgeControl - stationOutputOpenReceiveChannel TCPPid =
[1.100.9.1] mylP: 65fa010a (10.1.250. 101) ConferencelD: 16781219
MediaPartyld: 16777228, msecPacketSize: 20, COmpxessionType; 4,
security: 0|l,100,211,14.1****
StationD: (0000001) OpenReceiveChannel conferenceID=l8227819
passThruPartyID=1677722 9 millisecondPacketSize=2 0
compressionType=4 (Media_Payload___G711Uiaw64k) RFC2833PayloadType=0
qualifierln=? sourceIpAddr=IpAddr.type:0
ipAddr:0x0a01fa65000000000000000000000000(10.1.250.101). mylP:
IpAddr. type: 0 ipv4Addr: 0x0a01020c(10.1.2.12) |l,100,211,13.1***'*
Stationlnit: (Ooooooi) OpenReceiveChannelAck Status=0,
Troubleshooting Cisco Unified Communicalions (TVOICE) v8 0 2010 Cisco Systems, Inc.
ipAddr:0x0a01fa65000000000000000000000000(10.1.2 50.101), Port=1682 0,
PartyID=16777228|1,100,49,1.453*10.1.250.101*CFB-HQ1
StationD: (0000001) StartMedaTrattemiSSi>n COnferenceID=182 27819
passThruPartyID=1677722 9 remoteIpAddress=IpAddr.type:0
ipAddr:0x0a01fa65000000000000000000000000(10.1.250.101)
remotePortNumber=1682 0 milliSecondPacketSize=20
COmpressType=4 (Media_Payload_G7UUiaw64kJ. RFC2833PayloadType=0
qualifierOut=?, mylP: IpAddr.type:0 ipv4Addr:0x0a01020cf10.1.2.12)
|l,100,49,1.453*10.1.25Q.101*CFB-HQl
Stationlnit: (0000001) OpenReeeiveOianhelAGfc Status=0,
ipAddr:0x0a0102OcOOO0 00000000000000000 000(10.1.2.12), Port=230 60,
PartyID=16777229|l,100,49,1.454*10.1.2.12*SEP0 024C4454AD8
[1.100.9.1] mylP: 65fa010a (10.1.250.101), security:
0.|1,100,49,1.454*10.1.2.12*SEP0024C4454AD8
UnicastBridgeControl - star^StatiemOutputStartMedlaTranSffiissiori -
(10 .1 .2 .12) RemoteRtpPortNumber: 5A14 msecPacketSize: 20
congressionType: 4, TCPPid =
[1.100.9. llll,100,49,1.454"10.1.2.12~SEP0024C4454AD8
iraciti'
Stationlnit; (000 0004) SoltKeyEvent softKeyEvent=2(NewCall) linelnatan
ca11 Reference =0 . 1.100.49,1.4 57"10.1.2.11"SEF0 024 97A9DD32
StationD: (0000004) DialedNumber aialeaHumt>er=210D linelri3tance= 1
callReferecce=lS227S23. 1,100,49,1.462'10.1.2.11*SEP0024 97A9 0D32
. . , truncated. . .
SKDKShaiedData: : findLoealDevice - Name=CPB-HQl Key-2d3 47B4-810e-Otlll-
47 535b=i21daf isActive=l Pi d= (1 ,12, 11
found 1.100,19, 1.46 2 '10. 1.2.ll'SEP002 497A90D32
UnicastBridgeControl::allocatest ream - Device Name.CPB-HQ1,
StreamAvailable ==40 StreamUsed=-1
Maxtreaa-,s-4S 1, 100, 43,1 ,462 '10. 1 .2 . 1 l"SEP0024 97A90D32
...truncated...
Stationlnit; 10000005) SoftKeyEvent softKeyEvent-2 (NewCall) linelnstan
callRefsreace=Q. 1,100,19.1.169*10,1.4,12'SEP0 024C4455561
StationD: 10000005) DialedNumber dialedNumber=2100 linelnstance=1
callReferenoe=l3227626. 1,100,49,1.477"10 .1.1. 12"SBP0024C4455S61
SHDHSharedData: : f mdLocalDevi.ee - Hame=CPB-HQl Key=2d3 f 47=34-810e-0bfd-
47535b621daf iaActiv=s = l Pid=[l,42,l)
Eound 1,100,4,1.477'10.1.4.12"SEP002 4C4455561
UnicastBridgeControl : =al loca teStream - Device Hame=.CI?B-HQ1 .
StreaicAvaliable-4 0 StreadUsed= 2
Ma*Str=3ams=4B I ,100, 19,1, 477* 10. 1.4.1"!*SEP0024C4455 561
This trace output starts with showing the second phone (SEP0024l)7A90D32) with the IP
addres?N 10.1.2.11 joining thc established Meet-Me conference. The IP phone dials the Meet-Me
pattem 210(1. the conference bridge is found (determined by the originator, not the MRGL). and
the conference bridge is successfully allocated.
Trace output for the setup of thc RTP media between thc conference bridge and the IP phone
that uses thc G.711 codec is shown here:
UnicastBridgeControl::handleSetupSuccess - Device Name = CFB-HQ1,
Party CI = 18227825|1,100,49,1.462*10.1.2.11*SEP002497A90D32
StationD: (0000004) SEP002497A90D32 ,
star_MediaExchangeAgenaOpenLogicalChannel packetSize=20, codec=4,
ci=18227823|1,100,211,15.1****
UnicastBridgeControl - stationOutputOpenReceiveChannel TCPPid =
[1.100,9.1] mylP: 65faQl0a (10.1,250.101) ConferencelD: 16781219
MediaPartyld: 16777230, msecPacketSize: 20, compressionType: 4,
security: 0(1,100,211,16.1****
StationD: (0000004) OpenReceiveChannel conferenceID=182278 23
passThruPartyID=16777231 millisecondPacketSize=2Q
compressionType=4(Media_Payload_G711Ulaw64k) RFC2 833PayloadType=0
qualifierln-? sourcelpAddr-IpAddr.type:0
ipAddr:OxOa01fa650000000000C0000000000000(10.1.250.101). mylP;
IpAddr.type:0 ipv4Addr:0x0a0102 0b(10.1.2.11) |l,100,211,15.1****
ipAddr:0x0a0Ifa6500 0000000 00 0000000000000(10.1.250.101), Port=16630,
PartyID=16777230|1,100,49,1.463*10.1.250.101*CFB-HQ1
StationD: (0000004) startMediaTransmission conferenceID==18227823
passThruPartyID=167772 31 remoteIpAddress=IpAddr.type:0
ipAddr;0x0a01fa65000000000000000000000000(10.1.250.101)
remotePortNumber=16630 milliSecondPacketSize=2 0
compressType=4(Media_Payload_G711Ulaw64k) RFC2 833PayloadType=0
qualifierOut=?. mylP: IpAddr.type:0 ipv4Addr:0x0a0102 0b(10,1.2.11)
I1,100,49,1.463*10.1.25 0.1Q1*CFB-HQ1
6-90 Troubleshooting Cisco Unified Communications (TVOICE) v8 0 201C Cisco Systems, Inc
ipAddr:0x0a01020b000000000000000000000000(10.1.2.11), Port=23996,
PartyID=16777231jl,100,49,1.465*10.1.2.11*SEP002497A90D32
UnicastBridgeControl::star_MediaExchangeStartTalking need to send
SMTAck.|1,100,49,1.465*10.1.2.11*SEP002497A90D32
[1.100.9.1] mylP: 65fa010a (10.1.2 50.101), security:
0.|1,100,49,1.465*10.1.2.11*SEP002497A90D32
ConferencelD: 16781219 MediaPartyld: 16777230, RemotelpAddr: b02010a
(10.1. 2 .11) RemoteRtpPortNumber: 5DBC msecPacketSize: 20
[1.100.9.1]|1,100,49,1.465*10.1.2.11*SEP002497A90D32
The bottom half of the trace output in the figure shows how the branch phone
(SEP0024C445556!) with the IP address 10,1.4.12 joins thc Mcet-Mc conference. The RTP
media setup procedure is like the ones that are shown for the previous two phonesthe only
exception is the codec. The branch phone uses G.729 instead of G.711.
2010 Cisco Systems, inc Voice Quality and Media Resources Issues
Meet-Me Conference Setup Completed
This figure shows thc Mect-Mc conference setup that is completed at the hardware conferenci.
bridge.
167812:J 167772 52
167S1223 16777250
16781223 167772=18
=?r.~foA' -j.
stype mode codec sport rport ripaddr
cont sendtecv y729 17 640 31736 10.1.1.12
:onf sendrecv g7llu 18390 31112 10.1.2.11
:ont sendrecv glllu 18996 30786 10.1.2.12
HQ-i#sr=aw dspfarnm dsp al 1
SLOT DSP VERSION STATUS CBNL USE TYPE RSC ID BRIDGE ID PKTS TXED PKTS RX
0 5 26.3 .I UP N/A FREE conf 1

,
1 -
V
.
1 1 . IE " 1
J 1
1
ft 1
0 7 26.3.1 UP N/A FREE conf 1
0 7 26.3 4 UP N/A FREE conf 1
Total du mbec o DSPFARM DEF c hannel (E) 6
The show seep connections command shows three conference legs, one to each phone. Ci.711
and G.729 codecs arc used for these legs.
The show dspfarm dsp nil command shows the allocation of digital signal processor (DSP)
resources. This conference bridge can host six DSP channels that equal to up to six conference
sessions. The three Meet-Me conference legs (it is a single conference session, though) have
allocated a single DSP. number 6, mixing the speech. The remaining DSPs arc available to
support additional conference sessions.
6-92 Troubleshooting Cisco Unrfied Communications (TVOICEl v8 0 ) 2010 Cisco Systems. Inc
Tracing an Unsuccessful Meet-Me Conference Setup
This figure shows an example of trace for an unsuccessful Meet-Me conference setup.
Tracing an Unsuccessful
Conference Setup
Stationlnit : (000 000!) Sof tKej-Bvent EcstF=,yF!vuTit= 16 (S'e'siHs.J linelnstanca-l
callReference.306767 96.|1,100,49,1,3335*10.1.2.12*SEP002 497A90D32
.. .truncated. ..
Digit analysis: match I pi." 2' . 1 qeo- -51155520 02- , en-"2 002" ,plv=."5",
psB.,HQ_Emergenci'_Pt:HO_Local_Pt! Internal _Pt",
TodFilteredPsa-"HQ Bmergency_Pt:EQ_I.ocal_Pt:Internal Pt",
:,- ".dec-"!") 1,100,49,1.2340*10.1.2 . 12 "SBP0O2 497*9 0D32
Digit analysis; analysis rasulta 1,100,49.1.2340"10.1.2.12"SEP002497A90D32
' PretranaformCallingPartysumber.2002
CalliogPartyNuiTLboc-2002
DialingPartltion-InternalPt
b::..;^ xi'.x i^pc! .'a; j.-31-ar ~,'.i yti~.=>/=. li!0
.. . truncated...
RouteBloctt Flag-So;. =.<=rhi ; Sat -.sen
Linecdpc (11 >: -dispat cbToAUDeviceo-, BlgName-CcProceedReq,
device.3EP002197A90D32 1.100.49.1.2340"10.1.2.12"SEP002497A90D32
HeetMeConSerenceHanager - awaitcal1 info_SsCallInfoRes - Conference coda
DN.(210D] is already in use. Clearing the MeetMe attempt for
party-i [306767961 . No. of current
instances? [1] '1.100.49,1.2340" 10.1.2. 12"3RP002 497A90D32
The trace starts with the phone softkey event the Meet-Me softkey button is pressed.
Then, the digit analysis stage shows the collected digits of 2100 and the digit analysis result is
RoutcThisPattern. This means that the dial plan does not have any issues, and the called
number is recognized as reachable.
At the end. thc trace output shows that the MeetMeConferenceManager has returned the error
message and reports that another conference has been set up at the Meet-Me pattern 2100.
Either another Meet-Me pattern must be used, if available, or thc current conference must
release the 2100 when it is completed.
Summary
Cisco Unified Communications Manager supports hardware and
software conference bridges and Ad Hoc and Meet-Me
conference types.
The common causes for conference bridge registration issues
include the Cisco IP Voice Media Streaming Application not
running, conference bridges being shut down or misconfigured,
or network connectivity issues.
The common issues of Ad Hoc conferencing include conference
participants who cannot be added, conference participants who
are suddenly dropped, or conferences that cannot be linked.
The common issues of Meet-Me conferencing include
conferences that cannot be set up or participants who cannot
join the conference
In this lesson, you ha\ e learned to explain thc common issues that are related to conferences
and identify the most likelv causes of these issues.
References
Cisco Unified Communications Manager System Guide, Release 8.0(2), Conference
Bridges at hup- uuu. eisco.coin en'I.'S partner'does/voice ip eomm'cuemadmin K 0 2
cems\s aiLVonfb.bunl
6-94 Troubleshooting Cisco Unitied Communicalions (TVOICE) v8 0 ?010 Cisco Systems, Inc
Lesson 4
Troubleshooting Transcoder
issues
Overview
Cisco Unified Communications Manager invokes a transcoder on behalf of endpoint devices
when the two devices are using different codecs and would normally be unable to
communicate. When inserted into a call, the transcoder converts the data streams between the
two disparate codecs to enable their mutual communication.
Objectives
transcoders and identify the most likely causes of these issues. This ability includes being able
to meet these objectives:
Reviewmajor scenarios and considerations when a transcoder is used, reviewthe codec
selection process for Cisco Unified Communications Manager and voice gateways, and
outline general issues that can be experienced with a transcoder
Describe the transcoder registration issues and explain how to troubleshoot them
Describe thc major reasons why a transcoder allocation might fail
Transcoder Review
6-96
This topic reviews major scenarios and considerations when a transcoder is used, reviews the
codec selection process for Cisco Unified Communications Manager and voice gateways, and
outlines general issues that can be experienced with a transcoder.
Transcoders are available only as a hardware media resource.
Hardware
Transcoder in
Cisco IOS Router
A transcoder takes the stream of one codec and converts it from one compression type to
another. For example, a transcoder could take a stream from a G.711 codec and transcode it in
real time to a G.729 stream. In addition, a transcoder provides Media Tennination Point (MTP)
capabilities and can enable supplementary services for 11.323 endpoints when required.
The Cisco Unified Communications Manager invokes a transcoder on behalf of endpoint
devices when the two devices use different voice codecs and would normally be unable to
communicate. When inserted into a call, the transcoder converts the data streams between the
two incompatible codecs. The transcoder remains invisible to the user and to the endpoints that
are involved in a call.
Troubleshooting Cisco Unified Communications (TVOICE) v8 0 '2010 Cisco Systems, Inc
Codec Configuration and Selection Review
*mm
This section reviews how to control the codec that is used for acall in Cisco Unified
Codec Configuration and Select!
Each "'; "'s configured with thecodec with thehighest permitted bandwidth
req uiremenls:
Within Ihe configured region,
Toward specific other regions (manually configured).
Toward all other regions (which ha/e notbeenmanually added).
Region is assignedtodevices viadevicepool.
Cisco IOS gateway:
. ",..-- --'i-.-,r=!M of c.y;ra (codec class)configured at incoming oroutgoing
VOIP dial peerisusedfor therespective direction ofthecall.
Actually used codec depends on the capabilities of the two devices:
. ? >..- :- -ihatis-s.-c-'i i ''V bo.i 1r.'to*. anddoes-id >>.-f-(>iH,Lin.lftidMi
i- ..- '-.of codec permitted inregionconfiguration.
- Ifdevices cannot agreeona codec, a transcodingdevice isinvoked.
The codec that will be used for acall depends on thc Cisco Unified Communications Manager
region configuration. Each deviee is assigned with aregion via the device pool configuration.
For each region, the administrator can configure the codec with the highest permitted
bandwidth requirement within aregion to other specifically listed regions and to all other
(unlisted) regions.
If aCisco IOS gateway is involved, one codec is set at the applicable dial peer, or the dial peer
is configured with alist of codecs by using the voice class codec command.
When acall is placed between two devices, the codec is determined based on the regions and
on the capabilities of the two devices. They will use the best codec that is supported by both
devices and that does not exceed the bandwidth requirements ofthe codec that is permitted tor
the regions that are involved in the call. If the two devices cannot agree on acodec (for
instance, ifregion configuration allows G.729 as the maximum codec but one device only
supports G.711), a transcoder is invoked, ifavailable.
Voice Quality and Media ResourcesIssues 6-97
Transcoder Decision Tree
The figure shows a decision tree for transcoders.
When acall is placed. Cisco Unified Communications Manager determines if atranscoder is
needed. In aease like this, no common codec is supported on both endpoints, or no common
codec is permitted, based on the region configuration.
Note
For example, in the second condition, both endpoints support G.711. However, based on the
regions, G.729 must be used. If only one device supports G.729, then-although the two
devices both support acommon codec (G.711 (-they cannot use that one because of the
codecs that arepermitted by theregion configuration.
It the need for atranscoder is identified, Cisco Unified Communications Manager tries to
allocate the appropriate media resource by using the Media Resource Group List (MRGI )of
the device that supports only the codec with higher bandwidth consumption. Ifno transcoder is
found, the caii tads. It atranscoder can be allocated, the call is split into two call legs- one from
each endpoint to thc transcoder.
Now, Cisco Unified Communications Manager checks ifacommon codec can be found at each
ot these call legs. If this is no. possible -for example because thc transcoder is in aregion dial
allows akm-bandwidth codec only, which is not supported by one endpoint then the call
tails.
If the region configuration allows acommon codec to be found at each call leg then Call
Admission Control (CAC) is performed. Ifone or both call legs are not admitted, thc call fails
More precisely, ano-bandwidth condition isencountered and automated alternate routine
(AAR) is used, if applicable.
Ifboth call legs to the transcoder are admitted, the call is established and the transcoder
transcodes between the two audio streams that arc generated by the endpoints.
J2010 Cisco Systems. Inc
taw
General Transcoder Issues
This section lists the general issues that are potentially experienced when using atranscoder.
General Transcoder Issues
General problems related totranscoders:
Calls between endpoints with different codec requirements
not possible.
Call setup fails when transcoder required.
General transcoder issues:
Transcoder cannot register dueto network issuesor
misconfiguration.
Transcoder registered but unavailable to calls due to running
out of resources or misconfiguration.
If calling does not work between regions with uncommon codecs, the problem could be directly
related to transcoder availability. The call setup might fail when the transcoder is required but
is unavailable.
The following lists the most common transcoder issues:
. The transcoder cannot register because of network issues or because of amisconfiguration
at the Cisco Unified Communications Manager or the Cisco IOS transcoder.
The transcoder could be successfully registered but is unavailable to support calls. The
transcoder can run out ofresources, orits association with adevice that requires
transcoding services could bemisconfigured.
Troubleshooting Transcoder Registration Issues
This topic describes transcoder registration issues and explains how to troubleshoot them.
6-100
3Q-lahow seep
SCCP Adnir, states UP
IPv4 Address: 10.1,250.JC1
Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 10.1.1. i. Port Number: 2000
Priority: N/A. Version: 7.0, Identifier- 1
Trustpoint N/A
Transcoding Oper state, ACTIVE - Cause Code: NONE
Actrve Call Manager: 10.1.1.1, Port Number; 2000
TCP Lir.k Status: CONNECTED, Profile Identifier- 1
Reported Max streams B, Keported Ma, OOS Streams:
Supported Coder: gTllulaw, Maximum Packetization Period: 30
Media Resources > Transcoder
til irt- 11 1 l.l lil
The figure shows how to venfy that atranscoder is registered with Cisco Unified
Commumcations Manager. The Cisco IOS show seep command displays that this transcoder
up and us operanonal state ,s ACTIVE. I, ,s also connected to the Cisco Unified
c3n'fa7rS ManagLT' ThC b0nm f'8Urc Sh0WS how l vcri* 'scoder Ration from
Usco Unified Communications Manager.
The typical configuration ofaCisco IOS transcoder is like this:
voice-card 0
dspfarm
i
interface LoopbackO
ip address 10.1.250.101 255.255.255.255
seep ccm lO.l.i.i identifier 1 version 7.0
seep
seep ccm group 1
associate profile 1 register TX-HQ1
dspfarm profile l transcode
codec gvilulaw
codec g"?Halaw
TroubleshootingCisco Unified Camm
unications (TVOICE) v8 =
2010 Cisco Systems
codec g729ar8
codec g729abr8
codec g729r8
maximum sessions 4
This panicular transcoder can transcode between G.711 and G.729 codecs.
)2010 Cisco Systems, Inc VoiceQualityand Media Resources Issues 6-101
HQ-l#show seep ccm group 1
Description: None
Binded Interface: None
Associated CCM Id: 1, Priority
Associated Profile: 1, Recistra
Registration Retries: 3, Regist
Keepalive Retries: 3, Keepalive Timeout: 30 sec
CCM Connect Retries: 3, CCM Connect Interval! 10 sec
Switchover Method: GRACEFUL, Switchback Method: GRACEFUL GUARD
Switchbac* Interval: 10 sec. Switchback Timeout: 7200 eec
Signaling DSCP value: es3, Audio DSCP value: ef
/
this CCM Group:
Name:|TX-HQl|
Media Resources > Transcoder
For a transcoder toregister properly, itsregistration name that isconfigured at the Cisco IOS
router must match the name that is configured at the Cisco Unified Communications Manager.
Olderdigital signal processor (DSPs) might use the namethat is built fromthedeviceMAC
address byusing MTPmac address notation. Thefigure shows thatthename TX-IIQ1 matches
as required. This is the most common error when configuring transcoders. Also, make sure that
the transcoder is not shut down at the Cisco IOS router.
For the trai^coder toregister properly, make sure that the Cisco II' Voice Media Streaming
Application service is running. Verify this bynavigating toCisco Unified Serviceability >
Tools > Control Center Feature Services. Incase the service is reported as not running, restart
the Cisco IP Voice Media Streaming Applicationservice.
6-102 TroubleshootingCisco Unified Communicalions (TVOICE) vB 0
Transcoder Registration Issues
This list summarizes the most common reasons why a transcoder registration fails.
Transcoder Registration Issues
The common causes for transcoder registration issues:
CiscoIPVoice Media StreamingApplication service not running.
Transcoder shutdown or its name does not match.
Type of transcoder does not match.
Network connectivity or DNS issues.
Access list in the path filtering SCCP.
These might bethe most common causes of failed registration:
Cisco IP Voice Media Streaming Application is notrunning properly. It has been
deactivated or is down duetosoftware error. Restart theservice inCisco Unified
Serviceability.
Thetranscoder is shut down or its name docsnot match theCisco Unified Communications
Manager configuration. The transcoder name can have up to 15 characters. Valid characters
comprise alphanumeric characters (a to z, Ato Z, 0to 9), as well as dot (.), dash (-), and
underscore (_). For Cisco MTP (WS-SVC-CMM) transcoders, thc system fills in this value,
which is based on theMAC address thatyouprovide. Make sure that noshutdown
command isapplied within the DSP farm profile for the transcoder. For most Cisco IOS
versions, however, theno shutdown command doesnot show when displaying thc
configuration (shutdown is displayed, though).
Cisco Unified Communications Manager supports several types of transcoding devices.
Choose the appropriate transcoder type:
Cisco Media Termination Point Hardware: This type supports the Cisco Catalyst
4000 WS-X4604-GWY and the CiscoCatalyst6000WS-6608-T1 or WS-6608-E1.
Cisco IOS Media Termination Point: This type supports the Cisco 2600XM,
Cisco 2691, Cisco 3725, Cisco 3745, Cisco 3660, Cisco3640,Cisco3620,Cisco
2600, and Cisco VG200 gateways (NM-IID).
Cisco IOS Enhanced Media Termination Point: This type supports Cisco
2600XM, Cisco2691, Cisco3660, Cisco3725, Cisco3745, and Cisco 3660Access
Routers (NM-HD andNM-HDV2).
CiscoMediaTermination Point (WS-SVC-CMM). Thistypesupports the
Catalyst 6500 Series and Cisco 7600 Series (WS-SVC-CMM).
The transcoder is usually located atclose proximity to thc Cisco Unitied Communications
Manager server, but itcan be placed wherever required. IP network connectivitv issues
could also be present, preventing successful registration. Ifnames are used instead ofIP
addresses, verify that Domain Name System (DNS) name resolution works as expected.
It atranscoder is behind afirewall or access list, make sure that the TCI1 port 2000 (Skinny
Client Control Protocol [SCCP] lis enabled for the communications between thc transcoder
and the Cisco Unified Communications Manager cluster.
Troubleshooting Transcoder Allocation
This topic describes the major causes atranscoder allocation failure.
Transcoder Allocation Issues
The common causes for transcoder allocation issues:
No transcoder registered at CiscoUnified Communications
Manager.
Transcoder registered but running out of resources.
MRGand MRGL misconfigured; transcoder unavailable to
endpoint.
Network connectivityissues to transcoder.
DSP farm and transcoder codec issues.
This section lists themost common causes fortranscoding allocation issues. Cisco Unified
Communications Manager can fail toallocate atranscoder resource, when required, for these
reasons:
No transcoder isregistered atthe Cisco Unified Communications Manager. Make sure that
the appropriate transcoder successfully registers before calls arc placed.
Transcoder is registered, but itmight be running out ofresources. Transcoders have afinite
number ofsessions that they can support, depending onwhich platform isselected to
implement the transcoder. Make sure that additional transcoding resources are deployed if
resources are frequently depleted.
Thc media resource might bemisconfigured when you use Media Resource Group (MRG)
and MRGL. Misconfiguration can make atranscoder unavailable toan endpoint when
needed. AnMRGL thatcontains MTP resources should beassociated witha device that
requires transcoding.
Atranscoder might seemtobe registered, but it isno longer accessible, and itcannot be
allocated. Keepalives are exchanged between the transcoder and Cisco Unified
Communications Manager with adefault 30-second period. Ifnetwork connectivity issues
appear in between two keepalives, this kind ofsituation can be experienced. In this case,
the transcoder will fail and will start renewing its registration.
Issues might also exist with a DSP farm that performs the transcoding function at the
transcoder platform. Arequired codec might not be part ofthe DSP farm profile and,
therefore, will be unavailable for any of the transcoding legs.
Voice Qualityand Media Resources Issues 6-105
Verify Transcoder Utilization
If problems with atranscoder allocation are experienced and thc transcoder is properly
registered with Cisco Unified Communications Manager, ensure that thc transcoder does not
run out of resources.
Ve t; i y
s-n'-=s<=;=w.
S-LS'-.vo.,,,;.
> '.....=,*
a ==-->.. .-=p...
*-C3 ""is-cc. Si*-ti
>Li.s.:--!-
e-l.1i. .?! 'o=-j
CKis,. "-..-^J-rlf
:.=. )= 11 *." tt *=!'
[i'lL-} =J.-fi
3-H, u-Jh. d .j,,,., ,.. .,
fV-,, I*,[=_-.
C' ===!-'.=--
!.=..= Vt.
(Ve i-ut'ilcj-
=** W|T)"L'=--!=:>
Transcoder performance can also beverified atthe transcoding
platform; command depends on platform type.
The figure shows the Cisco Unified Real-Time Monitoring Tool (RTMT) performance counters
for atranscoder device. You can select from the counters to display various variables like how
many resources arc available, how many resources are used currently, which transcoders run
out ofresources, how many resources could not be allocated because ofopening port errors,
and soon. The counters can bedisplayed perindividual transcoder.
The figure shows two variables for the transcoder named TX-I1QI that has acapacity offour
resources. One is currently being used and three are available for the next calls.
Resources ofthe Cisco IOS transcoder can also be verified by using the show dspfarm profile
command. The following command output shows that four resources have been configured at
the Cisco IOS Enhanced Media Resource (Cisco 2800 Series):
HQ-l#show dspfarm profile 1
Profile ID = 1, Service - TRANSCODING, Resource ID = 1
Profile Description .
Profile Admin State : UP
Application : SCCP Status : ASSOCIATED
Resource Provider : FLEX_DSPRM Status : UP
Codec Configuration
Codec :g7llulaw, Maximum Packetization Period : 30
Codec :g7ilalaw. Maximum Packetization Period : 30
Codec g729ar8, Maximum Packetization Period . 60
Codec :g729abr8, Maximum Packetization Period : 60
Codec : g729r8, Maximum Packetization Period : 60
Codec : g723r63, Maximum Packetization Period : 60
Transcoder Allocation Trace
The trace output ofasuccessful transcoder allocation is shown in this section
KHi
StationD: (0000003) DialedNumber .,-- - -, r ' . . , "
iiRf.rMe,.i7S3*ji. 1.100,49.LiesB-io.i.i.ia-BwioaiciiswSr nstance=1
StationD: (0000003) (1,100,9,36) Cal 1info call i0gPartyNalnB, =
<=allingParty=511S55200l cgpnVoicaMailbox, al terna teCa 11 IngParty-
=a lledParcyHame = ' * , - ... j .
, f, ' -' ' CiJpjlVD.C8Milbox=
or=.glnalcailedPaityNaD,e... originalCalledParty.12125550120
callType=2( ., linainEtanee=1 callRBferen(;e=1755649l veriilon.
85720013 1, 100, no, 12.1'*
. . .truncated. . .
SKDHSharedData: ,r .;."-. ,,.,-
dlOB-ddbaicOefele isActive=i Pid=(l,173,3)
found L100,49,1.1666*10.1.2,12*SEP0021C4454AD8
HediaBesourceHanager::waiting NrmAllocateXcnderReEoureeBeo - CI.1755649 1
Count-1 1,100.16,12.2*10 1.2SO .! 02 *Pc rt 23696 "556493,
HedisResouiceiiansger: waiting HrmAllocatsXcoderReaourceRe,, - crprtiw Tm
USING MR3L LIST 1.100,16.12.2*10.1,250.I02'POrt 23696 C,"W "
. . . truncated. .
MRM^ccdvartSc^StringToStdStrlng XCOI.E *rg|1,100,16, 12.2 "10.1.250.102"Port
KKH.iQ.tXcodcD.vic.Oiv^llrgl Dev,cetae.IX-H0i DevicaType, 112 QroupÔ
Counter^ Capability.0 HultiCaBt-0 MRQL=cat=6 535e-3793 -e959-2094
3300698dd333 1, 100, 16,12 ,2 *10.1.250.102'Part 23696
i Key*b4M2cbc-c8al-1999-
The calling Cisco Unified IP phone (SEP0024C4454AD8) with the IP address 10 12Pplaces
i call to the public switched telephone network fPSTN) through an 11.321 gateway that is
capable ol G.723 codeconly.
the beginning of the trace shows the calling Cisco IP phone (SEP0024C4454AD8) dialing
91.12^0120. All trace output showing digit analysis and path selection is truncated to focus
solely on a transcoder allocation.
The outbound H.323 gateway with IP address 10.1.250,102 is selected lor this call The
gateway has been registered and ready to take calls (isActive- 1).
The Media Resource Manager (MRM) starts searching for atranscoder to support this call (he
Pphone has bee), associated with the MRGL (its name is not show in the output) that points to
the MRG that is called XCODE_mrg. The MRG lists thc transcoder named TX-IIQI
Transcoder Allocation Trace (Cont.)
SMDMSharedData: ;findl
Idc73bl37dl isActivt
21013
,icB . Name.TX-HQl Ky.9l)6cl52d-e23b-o556-0cb2-
.(1.119.5) found|1,100.16,12.2'10.1.250.102"Port
spo
caced.
HediaTerîtiationîntControl (5) : ,getReBourCeSAllocated - d^c.b^.-tx-IKH
Ci.17556514 HesourceCount.l|l.100.16.12.2-10.1.250.102*Port 21013
ll.ai.Traii..tlonfoii.tConti<ill5l..BtMOr.Xll.Mt.d -- Logging "9ifi-H
Caps and P/XCod.r Mgi<-WCW<|l,lO0,18,ia.a-lO.l.aso.l0a Port 21013
Kedi^ôatiouPolntControllSl^getRaBOurcesAUocated -- Logging RegionB.BR
Caps and KTP/XCoder Regior-HO Cape I1,100, , 13 .2"10. 1.250.102 'Port 21013
IS 11 12 9 257 259 261 |1,100.16.12.2'10.1.250.102"Port 21013
MediaTe^nationPoiutControl'Sl^logCap.bilitieainTrace -- Devica Cape . 9
1,100, 16,12.2*10.1.250.102*Port 21013
MadiaTenninationPoidtConttol (5): .gatBasoutcaaAllocatd -- DevicsName-TJI-HQl
Ci-lT55t5H RaBourc.Allocated.il 1. 100,16,12.2" 10.1. 250.102'Port 21013
Media^rain.tionPoiatConti-olfSl^logR.-ourc.StatUBiiiTrace -- Device Naae-TX-
BQ1 ResourceAv.ilable.3 Be8oUIU..d-l | 1,100,16 ,12 . 2*10.1.250.102 Port 21013
KRHiiupdateXcodeCounter devNaaenTJI-HQl, countchangii-l
1,100,16,1 a.2"10.1.250.102*Port 21013
The transcoder is reported as found. The MRM starts with the allocation ofasingle resource at
the transcoder TX-HQ1. The call will be set up between devices that are assigned to separate
regions: RegionA-HQ (IP phone) and RegionB-BR (H.323 gateway). Thc transcoder is
capable of several codecs that are shown by the codes (Caps =4216 11 12 9257 259 261).
From this capability list, the codecs 4(G.711) and 9(G.723) are important for this call. The
transcoder has been allocated successfully. The bottom ofthe trace output shows that one
resource isused (for this call), and three resources are available after the allocation process at
TX-IIQ1.
Voice Quality andMedia ResourcesIssues 6-109
MediaTe
HediaTe
is'
tionPointCoDtrol (5) - MediaExchangeAgenaAEsoeiateReg ~
inserted AIFI45.0) 1 . 100, 211 , 45. 1 * *-
tiooPointControUS): :star HediaExchangeAgenaOpei.Logi.calCha.inel -
1, 100, 211,45.1* "'
KediaTeroinaticmPointControl (5): ;Etar MediaExchangeAgenaOpenLogi.calChannel -
- inserted Agenalastance-45 i 1 . 100, 211,45.1*='"'
HediaTerninationPointControl(5): :star StationOutputOpenReceiveChannel -
TCPPid, [1.100.9.311 mylP: 0x65fa010a (10.1.250.101) . .;;.. , =, -, ;,
1.100.211.45. I*'.*. ' ' "' cp"cktai"= compressionType: 4
. . . truncated. . ,
StationD; (0000003) SEP0024C4454AD8,
star HediaExcharigeAgenaOper.LogicalC'hannel pac*etSize =20 codec*4
ci-17556512 1.100,16,19.2*10.1.250.102'Port 21013
StationD: (0000003) OpenReceiveChannel t, .] -;<?. =::.. .=,., .
"-"- --' ' "i lliaecondPacl(etSize=20
coEpre6siooType=4(Media. Payload G711tIlaS4k) RPC2 e33PaylofldType==0
q.'-=alifierln =? sourcelpAddr.IpAddr. type;0
lpAddr:OxOa01a65QOOOOOOOOOOOOOOOOOOOOOOO(10.] .250,101). mylP. IpAddr type-0
ipv4Addr:Qx0a01020o(10.1.2.12) !1,100.16.19.2*10.1.250.102*Port 21013
Staucm.lt: ,0000005) OpenReceiveChannelAck statua-0, IpAddr, IpAddr. type :0
ipAddr:0<0aOlfa65000000000000000000000000 (10. 1.250.101) . Port=19514
1. 100, 4 9, 1,2667*10. 1,250. 101* TX-HQ 1
This section shou smedia setup between the IP phone and the transcoder and between the
H.323 gateway and the transcoder. Note also thc conference and party identifiers that can be
used to match with the Cisco IOS plalform. The upper part ofthe output shows that the Real-
Iime Transport Protocol (RTP) connection (receive side) is set up at the transcoder by usine
the compression type 4 (G.711 mu-law).
The bottom of the trace output shows the setup of the RTP connection between the IP phone
(10,1.2.12) and the transcoder (10.1.250.101) at the phone side. The port 195 14 is the RTP port
at thc IP phone side. Thc media setup coniinues at thc next figure.
6-110 Troubleshooting Cisco Unified Communications {TVOICEl v8 0
Transcoder Allocation Trace
Station
ipAddri
remoteP
conpres
qualli
1,100,
Station
ipAddr:
. . . tmn
HediaTe
CI-167S
. . .trim
MediaTe
TCPPid
(10.1.2
HediaTe
(10.1,2
1.100,
MediaTe
(0000003)
tartdediaTraosmiSBion ccsnf*rfin~'f'
. remoteIpAddrasb-IpAddr.type:0
OxO01fa6500 0000000000000000000000(10.1.250. 101)
ortNumt.er.19 5H milliSecondPaOtetSiie-20
sTvpe-4[Media Payload G711Ulaw6410 RPC2833PayloadType=.0
erOut-7. mylP: IpAddr.type: 0 ipv4*ddr: OxOa0102 0c(10.1. 2.12)
49,1.2 86 7'lO.1.25 0.10l'TJt-Hgl
Initi (0000003) OpenReceiveChannelAck StatuB.0, IpAddr-IpAddr.typB:0
OxOa0102OcOO000000000000000O000000(10.1.2.12), Fort-216B8,
;-:'-;<'. 1,100,4 9,1. 2868*10.1.2.12*SBP0024C4454ADB
cated...
ntioationpolntcontrol(51::atar HediaExchang*StartTalking -
1229, DSCPValue - 184i1,100.49,1.2868"10.1.2.12"SBP0024C4454ADa
cated... . ,
nDin9tionPointControl(5)::Star StationOutputStartHediaTransmission -
. (1.100.9.311 fflyIP; 0x65a010a
50 101) 1 100,49, 1.2868*10.1.2.12 *SEP0Q24C4454ADB
nninationPointControl(51::8tar StationOutputStartMediaTranamission -
... .....,,-;9 .,..,-^.!,-v;.:. !<'?-;->:, RemotelpAddr: 0x=O201Oa
."^rRemoteRtpPortNuaber: 21688 msecPacketSize: 20 coapreaBionType: 4
49,1 286 8*10.1.2. 12'SEP0024C44 54M1B
rninationPointCootroHS): utar HediaExchengeagenaOpenLogicalChannel -
:l:;;.i 1,100, IS, 19.6*10.1.250. 102*Port 21013
it ill
The media transmission at thc RTP channel has started. Port 21688 is the RTP port at thc
transcoder side for the leg toward the IP phone. The bottom of thc trace output shows the RTP
channel setup at the IP phone-to-transcoder leg in the opposite direction where the RTP channel
uses the compression type 4(G.711 mu-law). The leg between the IP phone and the transcoder
is complete with the establishment ofits RTP channels.
MediaTe
tion
.nationPointControl (5) : ifltar MediaExchangeAgenaOpenLogicalChannel
- - . . inserted
Agenalnstance=46 1, 100, 16,19 .6*10.1. 250. 302'Port 21013
MediaTernmationPointControKS): :star StaUonOutputOpenReceiveChannel -
TCPPid. [1.100.9.311 mylP: 0x65a010a (10.1,250.101) .;;(<.,,<,,,.-
'= DisecPacketSize: 30 compreasionTvue 1
1, 100,16,19 .6*10.1.250.102*Port 21013 "
MediaTernunationPointControl (5) : !Star NediaExchangestartTalking -
CI=16781229, DSCPValue . 1B4 1,100. 16.19.7 *10.1.250.102 *Port 21013
, - . truncated. . .
HediaTerninationPointControKS^star sta tionOutputsta. tMediaTranamiasion
TCPPid = [1.100.9.31) mylP: 0x65ta010a
110.1.250.101) 1,100,16,19.7*10,1.250.I02-Port 21013
MediaTerînationPointControl (5) t. :star sta tiOn0utpUtStartMadiaTra.nHmi3 aion -
,..","'.,";:,/' ----- '.- -. -:, ,.-_. i. :.--_>-,._ RemotelpAddr: 0*66aQ10a
(10...2SC.102) RenetaRtpPortMunbex, 18588 msecPacKetSUe: 30
compressionType: 9 1,100,16,IS.7'10.1.250.102'Port 21013
. . . truncated, . .
S"";i,*t- '0000005) CpenReceivechannelAc* st.tw-0, IpAddr,IpAddr. type, 0
ipAddr:0*OaOla=S5Oa01Q2O 0000000 00000 0000(10. 1.250. 101), Port=17292,
1.100,49,1.2670"10.1.250.101'TX-HQ1
This figure shows how the RTP channels for the transcoder-to-gateway leg are built. The upper
pan otthe trace output shows that thc compression type <-> (G.723 6.3 kb/s) is used.
Pon !? is the RTP port number at the 11.323 gateway (10.1.250.102), and port 1729^ is the
RTP port at the transcoder side for this transcoding leg. The RTP channels are built between the
gateway and the transcoder and placed into the media transmission state. Immediately the
transcoder starts transcoding between thc codecs Ci.711 (to and from the IP phone) and (i 1~>\
(to and from the gateway).
-112 Troubleshooting Cisco Unified Communications (TVOICE) v8.0
2010 Cisco Systems
"" Transcoder Allocation Debug
This section shows adebug of asuccessful transcoder allocation from the perspective of the
transcoder.
Transcoder Allocation Debin
HO-l#debijg accp meesagee
SCCP:rcvd OpenSeceiveChan
ooenReceiveChaanelHsg Into:
,.=,, - .. ,,,'!iij, '=!-. :,;lUl.ji) Jir'i' i'5 HIT "J'i'!
msec p)it aiie 20, compression type 4
qualifier in.ecvalue - 0, g723_bitrt. - 0, call_re - 17556514
stream pass through id - 16777216, rfe2833_pyload_type x 0
codec dynamic payload 0, codec node 0
Encryption info := algorithm id 0, =rey_len 0, salt.len 0
reguestedAddrType - 0, source.. ipaddr.ipAddrType - 0, source ip_addi -
10.1.2.12, source portnunber 4000
3CCP =sand opanReceiveChaiinelAck
SCCP:rcvd startHediaTransmisBlon / Relates to the IP phone
StartKediaTransmiss ionMsg Into:
msec pxtaiie 20, compresaiontype - 4
remote ipaddr 10.1.2.12, remote_port - 21683
qualifier out.precedence value - 184, qualif ierout.ssvalue - 0
qualifier out,m_f raoaa~p.sr_p.tt - 0, g723_bitrste 0, call ref - 17556514,
atream_pass_througti_id - 16777216 rfe2B33_pyload_typ . 0
codec dynamic payload 0, codec mode - 0
Encryption Info :: algorithmid 0, *ey_len Osaltlen 0
Kelatas to the IP pnone
The debug seep messages command generated this debug output, which shows thc SCCP
messages being received and sent at the transcoder. This figure shows the debug output that is
related to the IP phone leg. Note the same identifiers as seen in the trace output. Cisco Unified
Communications Manager requests thc transcoder to set up aleg (open RTP channel) by using
apacketization period of 20 ms and acompression type of 4(G.711 mu-law). The transcoder
acknowledges with OpenReceiveChannelAck.
The bottom ofthe debug output shows the start ofthe media transmission at this leg.
If problems with the DSP farm at the transcoder -for example, DSP cannot be allocatcd-you
can generate and inspect aDSP farm allocation debug. The debug dsp-resource-manager flex
dspfarm command shows the DSP allocation process at call setup and the DSP release process
at call disconnect:
HQ-lBdebug dsp-resource-manager flex dspfarm
Apr 12 14:01:28.212: flex_dsprm_dspfm_dsp_open;resld=0xl,serviceîd=2
mtp,2-xcode,4-conf}
Apr 12 14:01:28.212: flex_dsprm_dspfm_find_free_dsp_channel
Apr 12 14:01:28.212: flex_dsprm_dspfm_open_and_set_dspchnl
Apr 12 14:01:28.212: flex_dsprm_dspfm_ppen_and_set_dap_chnl: dsplO/4)
channel(Oi , credit ((
Apr 12 14:01:28 .212
dsp_channel(0/4/0)
flex_dsprm_dspm_find_free_dsp_channel: return
II-
Apr 12 14:02
Apr 12 14:02
credit=40
D4 045: lex_dsprm_dspfm_dsp_close: resld=0xl, service_id=2
34.045: flex_dsprm_dspfm_return_dsp_credit: dsp=0/4 channels
2Q10Cisco Systems. Inc.
Transcoder Ai
SCCP:rcvd OpenReceiveChannel Relates to the H.323 gate*
OpenHeceiveCbannelMsg Info:
pkt - 3C
pression type - 9
qualifier in.ecvalue = 0, g723 titrate . 0, call ref . 175565M
stream pass through id =, 16777216, rtc2=333 payload type - 0
codec dynsaic payload = 0, codec mode = 0
Encryption Info :: algorithm id 0, Key len 0. salt len 0
reguestedAddrType = 0, source ip addr.ipAddrType . 0, source ip addr -
10.1.250.102, soiree port number = 21013
SCCP:send 0[. enRec ei veChanne 1Ack
SCCP:rcvd StartMediaTransmiss:
StartMediaTransmissionMsg Infc
maec pkt
Rali
to the H.323 gateway
- 30, compression type = 9
remote ip addr = 10.1.250.Iu2. remote port iB58e
gualifier out .p recedence value =. m, qualifier out.ssv;
qualifier out.max frames per pkt = 0, S723 titrate = 0
stream pass through id = 16777216 rfc2B33 payload type .
codec dynamic payload = 0, codec mode = 0
Encryption Info :: algorithm id 0, key len Dsalt len 0
ue = 0
all ref = 17556514,
This tigure shows ihe debug output that is related to the H.323 gateway leg Cisco Unified
Communications Manager requests the transcoder to set up aleg (open RTP channel) in which
Ihe RTP channel can use the packetization period of 30 ms and the compression type 9(G.723)
Ihe transcoder acknowledges with the OpenReceiveChannelAck.
The bottom of the debug output shows the start of thc media transmission at this leg.
* Transcoder Allocated
This section shows how you can verify that the transcoder has been successfully allocated.
Transcoder Allocated
BO-ltsho
sess id
ccp connections
conn id stype node codec sport rport ripaddr
.code sendrecv g723br 17292 18588 10.1.250.102
xcode sandrecv g711u 19514 21668 10.1.2.1!
ot active eeesion(s) 1. and connectionls) 2
HQ-llshov dspfarm dsp all
SLOT DSP VBRSION STATUS CENI, USE
PKTSRIBD
0 1 26.3.4
TYPE RSC ID BRIDGE1D PKT3TXED
OP N/A FREE xcode
UP H/>V FREE xcode
UP S/tk FREE xcode
UP 1 USED xcode
UP 1 USED xcode
DSPFW*K DSP channel (a) 4
OxSF
0x5 E
596
8 95
This upper figure shows the transcoder legs that have been set up. The show seep connections
command shows the session and both legs. The first row ofthe command output shows the leg
tothe H.323 gateway, and the second row shows the leg tothc IP phone.
Note the identifiers. The transcoder has two connections (legs) but only asingle transcoding
session that is equal to atranscoding resource. You can also see the same identifiers as are in
thc trace and debug output.
Thc bottom figure shows the DSP farm at the transcoder. Asingle DSP number 5has been
performing the transcoding on the two transcoding legs. Three DSPs are free and unallocated.
Summary
This topic summarizes the key points that were discussed inthis lesson.
"UnHVldf'
Transcoders are hardware media resources performing
transcoding between voice codecs.
Thecommon causesfor transcoder registration issues
include the Cisco IP Voice Media Streaming Application not
running, the transcoder being shutdown or misconfigured, or
network connectivity issues
The common causes for transcoder allocation issues include
thetranscoder not being registered or running out of
resources, media resource misconfiguration, network
connectivity issues, or DSP farm issues.
In this lesson, you have learned to explain the common issues that are related to transcoders and
to identify the must likelycauses of theseissues.
References
For additional infonnation. refer to this resource:
Cisco Unified Communications Manager System Guide, Release 8.0(2), Transcoders at
http; wuu cisco.,-om en L'S pannur.docs'wice ip cnmni'cucrn/adiiiin'8 D2
cans-.-. :iti>ir:!iis.lnm]
6-116 Troubleshoolmg Cisco Unified Communications (TVOICE) v8 0
&2010 Cisco Systems Inc.
rV
Lesson 5
Troubleshooting Issues with
RSVP Agents
Overview
Resource Reservation Protocol (RSVP) specifies aresource-reservation, transport-level
protocol for reserving resources in IP networks. RSVP provides amethod to achieve Call
Admission Control (CAC) in addition to location-based CAC. Location-based CAC constitutes
apoint-to-point CAC mechanism that does not take into account topology changes or multitier
topologies, whereas RSVP does consider these topologies.
Many customers request afull-mesh network topology for their video conferencing and video
telephony environments to match their existing topology. RSVP can manage complex
topologies. Location-based CAC supports only hub-and-spoke or point-to-point topologies such
as simple Multiprotocol Label Switching (MPLS) any-to-any topologies.
Cisco Unified Communications Manager uses an RSVP agent, aCisco IOS Software-based
RSVP proxy with aSkinny Client Control Protocol (SCCP) interface to support RSVP. Cisco
Unified Communications Manager communicates with the RSVP agent through aset ofSCCP
messanes. The RSVP agent registers with Cisco Unified Communications Manager as either a
MediaTenninationPoint(MTP)or a transcoder device.
Objectives
RSVP agents and identify the most likely causes ofthese issues. This ability includes being
able to meet these objectives;
Review the CAC process that is based on RSVP, RSVP agent implementation, and the
characteristics ofthe three call legs and outline the major issues that are experienced with
RSVP agents
Describe the RSVP agent registration issues and explain how totroubleshoot them
Describe the major RSVP CAC issues that can be experienced at the call setup process and
explain howto troubleshoot them
Review the intercluster RSVP (known as SIP Preconditions) feature, describe the major
issues that can be experienced when using it, and explain how totroubleshoot the issues
RSVP CAC
This topic reviews CAC processes that are based on RSVP, RSVP agent implementation, and
characteristics of the three call legs, and outlines major issues that are experienced with RSVP
agents.
Based onCisco Unified Communications Manager locations
Allows RSVP to beenabled selectively between pairs of
locations or within a location
Uses RSVP agents
Devices (MTPs) through which the call must flow
RSVP used between RSVP agents
Topology-aware
Works well with all topologies (full mesh, partial mesh
hub and spoke)
Adapts to network changes, considers actual topology;
* Link failures
* Backup links
" Load-share paths
RSVP-enabled locations are based on Cisco Unified Communications Manager (standard)
locations. The main difference is that RSVP can he enabled selectively between pairs of
locations or within alocation. As endpoints such as Cisco IP phones do not support RSVP
the solution uses so-called RSVP agents.
An RSVP agent is adevice (an MTP. to be precise) through which the call must flow. RSVP is
then only used between the two RSVP agents, while the Real-Time Transport Protocol (RTP)
stream from IPphone to RSVP agent is not using RSVP.
The second and most important difference to standard locations is that by using RSVP this
CAC mechanism is topology aware. It works well with all topologies (full mesh, partial mesh
hub and spoke) and adapts to network changes by considering thc actual topology. Advantages
include these considerations:
Link failures: Ifone link in the IP network goes down and packets are routed at different
paths. RSVP is aware ofthc change, and considers the bandwidth that is now available at
the actually routed path.
Backup links: Ifbackup links arc added after link failures orifbandwidth on demand is
used to add dial on demand circuits RSVP again is fully aware ofthe currently used routum
path and the bandwidth available on each link along that path,
Load-share paths: Ifload-sharing is used, RSVP isaware ofthe overall bandwidth that is
provided by multiple load-shared links.
Using RSVP for CAC simply allows admitting ordenying calls that are based on actual
oversubscription always based on the currently available bandwidth and interfaces and not on
a logical configuration that is ignoring the physical topology.
Three Call Legs with RSVP-Enabled Locations
When using RSVP-enabled locations, the end-to-end call is split into three separate call legs.
Three Call Legs with RSVP-Enabled
Locations
Location A
-;;f=> t'.-R=VP4=
.'" Cisco Unified *
Communications
Manager
AV'VP
RSVP
RTP
Location B
Cull le;-! 3
Phone1 'oRSVPAg'-ci'
In the figure, Phone 1in location Aplaces acall to Phone 2that is configured with location B.
Cisco Unified Communications Manager location configuration specifies that RSVP must be
used for calls between these two locations.
Cisco Unified Communications Manager instructs the two involved RSVP agents (one in
location Aone in location B) to use RSVP to try to set up the call between them. Ifthe call is
admitted (that is, there is enough bandwidth available in the network path between these two
devices), the RSVP agents inform Cisco Unified Communications Manager that the RSVP call
leg is successfully setup.
Now Cisco Unified Communications Manager tells the phones to set up their call legs, each to
its respective RSVP agent. If the RSVP call setup between the two RSVP agents is denied,
Cisco Unified Communications Manager considers that the call failed CAC.
Note the three separate RTP streams: Phone 1talks to RSVP agent 1, RSVP agent 1talks to
RSVP agent 2, and RSVP agent 2talks toPhone 2.
RSVPCACis usedbetween the RSVPagentsonly.
Characteristics of Phone-to-RSVP Agent Call Legs
Jhis section describes the characteristics of the call legs between phones and RSVP agents.
Based on Cisco Unified Communications Manager locations.
Usually a phone anditsRSVP agentare inthesame location.
If phone and its RSVP agent are in different locations standard
location-based CAC is performed for this caii leg.
Aphone must use itsRSVP agent
The RSVP agent thatis usedbya certain phone should beas
close as possible lo the phone.
The RSVP agent lobe used by a certain phone isdetermined by
the media resourcegrouplist ofIhe phone.
The RSVP agent is an MTP.
Pass-through codec is supported;
Nochanges to RTP payload.
Allows secure RTP to be used.
! No DSPs required at router.
Standard locations algorithms apply to the call leg between an IP phone and its RSVP aeent
Usually, they will be in the same location. Ifthey are in different locations, standard locations-
based CAC is performed for this call leg (phone to RSVP agent) first. Only if there is enough
bandwidth available for the IP phones to reach their RSVP agents will thc two RSVP agents trv
to set uptheircall leg by using RSVP.
An RSVP agent registers with Cisco Unified Communications Manager as aspecial MTP
device. Cisco Unified Communications Manager uses the Media Resource Group List (MRGI )
of the IP phone to determine which RSVP agent is to be used by which IP phone The
association of aphone to its RSVP agent is not performed by searching an RSVP agent in the
same location as the phone. As mentioned earlier, the IP phone and its RSVP agent can be in
different locations Only MRGLs are used to identify thc RSVP agent that is to be used by an
IP phone.
The RSVP agent that is used by acertain IP phone or group of phones should be iogicallv
located as close as possible to the IP phones. Close proximity reduces the possibility of '
suboptimal paths (accessing RSVP agents over the IP WAN). It also ensures that thc call leq in
which no RSVP-based CAC is performed (phone- to-RSVP agent) is ashort network path
(idealK. LAN)onl>. '
Thc RSVP agent supports pass-through codec configuration that allows any codec to be used
including Secure Real-Time Transport Protocol (SRTP) where the RTP payload is encrypted
(The codec docs not have to be known or supported by the RSVP agent.) In this ease the MTP
at thc Cisco IOS router can be implemented in software only. Ilowever, ifmidcall features
require amedia resource, for example, atranscoder, then an additional transcoder is allocated
ft the RSV Pagent ,sprovided by adigital signal processor (USP), that same media resource
can be used tor transcoding
Agent-to-RSVP Agent Call Leg
This section describes the characteristics ofthe call leg between two RSVP agents.
Characteristics of an RSVP Agent-l<
Based on standard Cisco IOS RSVP.
IPnetwork betweenRSVPagents is RSVP-enabled.
Eachinterface is configured with maximum bandwidth tobe
reserved by RSVP.
>IfRSVP is not enabled on any hop inthe path, the
appropriate link isignored by the CAC algorithm.
IntServ or DiffServ model is used:
RSVP only forCAC (control plane)
Low latency queuing for QoS(dataplane)
Call is set up only aftersuccessful RSVP CAC.
The call leg between two RSVP agents uses standard RSVP as implemented in Cisco IOS
routers. The IP network between the RSVP agents is RSVP-enabled that is, each interface is
configured with the maximum bandwidth that can be used for RSVP calls. If there is not
enough bandwidth available end-to-end (between the two RSVP agents, in this case), RSVP
CAC denies the call.
IfRSVP is not enabled on any hop in the path, the appropriate link is ignored by the CAC
algorithm (that is, it isalways admitted onthis link).
Cisco Unified Communications Manager RSVP agent CAC uses the Integrated Services or
Differentiated Services (IntServ or DiffServ) model for the RSVP call leg. This means, that
RSVP is only used for CAC (the "control" plane) but not with RSVP-reservable queues for
providing quality of seniee (QoS) to the streams. Instead, standard low latency queuing (LLQ)
configuration is required to provision QoS for the voice stream (the "data" plane).
The end-to-end call- that is, the incorporation ofall three call legsis only established after
the RSVP call leg has been admitted. Otherwise, the call fails because ofaCAC denial (not
enough bandwidth).
RSVP Agent Operation
The decision tree shows the operation ofRSVP agents.
L. %:
^
\
During an attempted call setup between two endpoints. Cisco Unified Communications
Manager first determines the codec to be used for thecall.
Then, Cisco Unified Communications Manager determines thc location of the calling and the
called devices. The location configuration is checked to determine which RSVP policy to apply
between the two locations or within the location (if thc called and the calling devices are in the
same location).
If the policy is set to Use System Default, thc Default Intcrlocation RSVP Policy Cisco
CallManager service parameter is looked up. Otherwise, the policy that is configured at the
location isapplied. Here are several possible policies:
No Reservation: In this case, standard location-based CAC without RSVP agents is used
Standard location-based CAC verifies adequate available bandwidth (if limited) for calls
coming out ofand going into alocation. Calls within alocation are not subject to standard
location-based CAC. For calls between two locations, the available bandwidth is checked at
both locations.
Optional (Video Desired), Mandatory, or Mandatory (Video Desired): If thc policy is
set to Optional (Video Desired), Mandatory, or Mandatory (Video Desired) Cisco Unified
Commumcations Manager determines which RSVP agenis will be used by thc calling and
the called devices.
Then, the call is split into three call legs. The first is from the calling device to its RSVP agent
the second is between the two RSVP agents, and the third call leg is from the called deviee
RSVP agent to the called device.
Ifany of the following applies to the first or third call legs thc call legs that are between the
endpoints and their RSVP agent then thc call fails. More precisely, the call is not admitted
This means that it automated alternate routing (AAR) is enabled and applicable, AAR will be
used:
RSVP is enabled for the call leg: RSVP is to be used within one call leg of an RSVP-
enabled call. This configuration is not supported and is treated as acall that is not admitted.
. Standard location-based CAC is applicable and the call is not admitted: The device
and its RSVP agent are in different locations, and at least one of these locations does not
have enough available bandwidth to process the call leg.
Ifnone of these conditions apply, Cisco Unified Communications Manager instructs the two
RSVP agents to establish the middle call leg between them. If RSVP admits the call, the end-
to-end call is successful. Otherwise, the call fails with ano bandwidth condit.on, wh.ch means
that AAR is invoked if applicable.
General RSVP CAC Issues
This list specifies the most common issues that are experienced when RSVP CAC is used at
*H .m** & ti
General issues of RSVP CAC:
RSVP CAC blocks calls when bandwidth isavailable.
" RSVP CAC doesnotmake bandwidth reservations when
expected
" RSVPagents cannot register.
These are general RSVP CAC issues:
Calls are being blocked by RSVP CAC when bandwidth is still available. When the call is
blocked, the reorder tone is heard.
Bandwidth reservation is not made for an RSVP CAC-controlled call when it is expected.
RSVP agents that are available to callers are unable to register with Cisco Unified
Communications Manager: consequently, RSVP CAC-controlled calls are blocked.
Troubleshooting the RSVP Agent Registration
This topic describes the RSVP agent registration issues and explains how to troubleshoot them.
Verifying RSVP Agent Registration
Hg-l#ahow seep
SCCP Admin seati= UP
Gateway Local Interface: LoopbacliO
IPv4 Address; 10.1.350.101
Port Number: 2000
IP Precedence: s
User (tasked Codec lint: None
Call Manager: 10.1.1.1. Port (lumber; 2000
Priority; N/A, version: 7.0, Identifiei
Trustpolnt; N/A
MTP Op
ACti
TCP
Report
Suppor
Suppo
. . . cr
RSVP
TLS :
r State: ACTIVE - Causa Code: NONE
Call Manager. 10.1.1.1. Port Number: 2000
nk Status: CONNECTED, Profile Identifier; 1
Max Streams; 20, Reported Max OOS Streams: 0
ted Codec: g711ulaw. Maximum Pecfcetiiation Period: 30
ted Codec; pans-thru, Maximum FacKetination Period; N/ft
.nested...
ENABLED
ENABLED
The figure shows how to verify that an RSVP agent is registered with Csco Unified
Communications Manager. Use the show seep command to display the states ofan RSV1
agent. Thc output shows that the RSVP agent is up, and its operational state (as the MTP) is
ACTIVE. Also, the SCCP over TCP is connected to the Cisco Unified Communications
Manager, and RSVP is enabled.
The typical configuration of aCisco IOS RSVP agent looks like this:
voice-card 0
dspfarm
I
interface LoopbackO
ip address 10.1.250.101 255.255.255.255
interface Serial0/1/0
no ip address
encapsulation frame-relay
no keepalive
no fair-queue
interface Serial0/l/0.121 point-to-point
ip address 10.1.6.101 255.255.255.0
frame-relay interface-dlci 121
ip rsvp bandwidth 128
seep ccm 10.1.1.1 identifier 1 version 7.
seep
I
seep ccm group 1
associate profile 1 register RSVP-HQ1
codec g711ulaw
codec pass-through
rsvp
maximum sessions software 10
-126 Troubleshooting Cisco Unified Communications (TVOICE) v8.0 2qio Cisc
Verifying RSVP Agent Registration
(Cont.)
Troubleshoot registration issues byusing debugseep messages
command, debug seep events command, orCisco CallManager
trace.
H0-l*show seep ccm group 1
Description; None
Binded Interface: Nona
Associated CCM Id: 1, Priority in this CCMGroup; 1
Associated Profile: 1. Registration Hi= |BS^-HQl|
Registration Retries: 3. Registration Timeout: 10 sec
Keepalive Retries; S. Keepalive Timeout: 30 sec
CCM Connect Retries: 3, CCM Connect Interval; 10 sec
Switchover Method: GBACBFUL, Switchback Method: GRACEFUL. GUARD
Switchback Interval: 10 aec, Switchback Timeout: 7200 aec
Signaling DSCP value: cs3, Audio DSCP valuejef
Media Resources > MTP
j^m.
To register properly, the registration name of an RSVP agent that is configured in Cisco IOS
Software must match the name that isconfigured at the Cisco Unified Communications
Manager. The figure shows that the name RSVP-HQ1 matches as required. Thc RSVP agent
registration name mismatch is acommon error when configuring RSVP agents.
Also, make sure that the RSVP agent profile is not shut down at the Cisco IOS router:
HQ-l#show dspfarm profile 1
profile ID = 1, Service - MTP, Resource ID = l
Profile Description :
Profile Admin State ; OP
Application : SCCP Status : ASSOCIATED
Resource Provider : NONE Status . NONE
Hardware Configured Resources
Hardware Available Resources
Software Resources : 10
Codec Configuration
Codec : g711ulaw, Maximum Packetization Period : 30
Codec : pass-through, Maximum Packetization Period : 0
RSVP : ENABLED
If the RSVP agent fails to register, use the debug seep messages command to display the
exchange of messages between the agent and Cisco Unified Commun.cations Manager. The
following debug output shows the successful registration:
HQ-lttdabug seep messages
0
0
Apr 15 07:45:44.782: SCCP:send RegisterMessage
protocol_ver=0xll0 08100
Apr 15 07:45:44.782: SCCP.send IpPortMessage
Apr 15 07:45:44.826: SCCP:rcvd RegisterAckMessage
Apr 15 07:45:44.826: SCCP:keepaalrve intervals, agreed seep ver=17
Apr 15 07:45:44.826: SCCP:send MediaResourceNotification
Apr 15 07:45:44.826: SCCP.send VersionReqiMessage
Apr 15 07:45:44.826: SCCP:revd Capabilr UesReqMessage
Apr 15 07:45:44.826: SCCP:send CapabilitiesResMessage
Apr 15 07:45:44,946: SCCP:revd VersionMessage
Apr 15 07:45:44.946: VersionMsg Info:
ve r gion:
Another way to troubleshoot registration problems is to use the debug seep events command
winch displays more details about the registration events:
.-.Q-^ndebug accp events
KQ lij
Apr HO^tlllt't-/. Syr-xaPP-l;lk-all-^rvices: create sym xapp_eve_q
seep setup_appl_service: Appl initialized with appl=4962D7K4 type-1
applrs-.ate=J, tnSg_har.d:e^41028A60, oper_state^0, sLati^47CE3E0C '
Zl 'I J:;^1r31I: sy"-xaPPJ-nsfGr_3UpporLed_ codec: supported codec cr.t 5
,; ;: -; '.*-;^-J-': sc-P^y^set_Secnre_param:: appl profile I rpm
dev_.fc_m._vie 0app_ device mode OLrp status for prof id 1 is 0
syin_xapp_new_rpT-_profilc info
*-!L15 C;:48:f:2--i2^ =->yn-,.xapp_new_rpm_pl-ofile_irlfo; max streams 20 p.-3f -q '
c.,aec_.._ ,, 1SS iJ( mspc 0( rsvp enabled| Lls e,lablpd " - "
Ar' i"r ,,'.''*V'--'?^: s=CIJaPP-Procoss-aYS_eveiiCs: sys_eve i, x:ated eve 0
-he rrMoll-''*'' SCCp-get -CCm 3rouP_îng_asSoc_prof; prof_id 1 found m
lPapp'jy.e3;r5yrevrr'PP-F:OCeSS--3y!i-eVen:S: Pr-Id ^ 3SSOC Co ccm^roup
J'",',3 <"'S-.S2.iZl; sccp_bri:]g_iJp_appl service: Trigger FSM for prof -r> _
llr~- iV-l-V* tOZ prf id lr sccP-în9_up_appl_Bervice "
to ccm o^îo^^^s^p^ri:0^-^"^^1-' ^
ddr uîXîo^r^^ Tryi119 CCM wich
tKe'oJ'^n?^^'2" SCCp-appl îce_start_tiêr: Start timer -vpe 1,;L,
^l1! -:4G:52'-21; -c"P_^P_^cket_cormect. Trying tcp soc connect to-
appJ_typc i, PrcE_id 1, lo ipaddr :o.l,i l
vS-a5"2r;4Sr'2"'21:-SCCP-TC-CCm-lntf-V1"f--id; Cc:m Bcc*> local êrface
vtt.a-^pp. type is ,, appl profile 1
toraddrCioTi52i'321r 5CCp-^P-open-an^s-Pt-ion: Socket 0opened and Pinded
to addr 10.1.1.1 - for appl_type 1, prof id 1
sccd1^-9-^ sccP-socket-conn"t_to_ccm :: connoting to port 2000
S-S ! " SC COnn t0 10-1-1-1 POrt'JOOO in prcgressfor
app-_.yp.= ., eule 1, sc--_td 0 app] 4962D7F4
Aor ^ ^:i-:""3f1: '^"^.appl, service, stop timer; Stop 4962U84C timer
Soc_id O^'state 1 SCCP prOCeSS-socket-conne-------result: appl type 1, evo ,,
^Vi 7i'^ :52':-&1:.sccF-Setap_aPpl_to_tcp_conn_atate: soc connected to
10.1.1.,, tor pior_ia 1, appl __ type 1
Apr is C?:48:52.33i: scc^_register_with_(-;onnecte<S ccm
Se'Lr^ôL'9" SCCP-app:-S startler: Start timer type 2w,th
acnl1t5.-r7:4S:52,"B1: S^p-re3lfitGr-with-Connected_ com :Sending regis msg for
appi_t/pt i , state ; J a ^ =-
arLp5lC":4S:S7''H; 'C"p-yet-aPPI-aerv-bV-Pr"f-_^^ Prof id 1found in the CCX
fipi 15 0/:4S:--2..-2-: SCCp_appl_servi ce_sL op_tirr.e r:Stop 49S2D84C timet
ItTlT^S^^-. seep set_swb_pbm_andj-.tart._t imer: prof_id 1, appl_type 1,
curr swb_bm C, ccm_pn 1, temp_swb_bm 0, new swb_bm 0, ccjnontyjojry 0,
Apr^Ôî-52.^25: sccp_xapp_associate_appl_service: XAPP type 1, prof assoc
result 0, prof id 1, appl_type 1
Apr 15 07:48:5^.425: SccP_appl_service_Start_timer: Start timer type 3 with
time out 30000ms , . _ .
Apr Is 07:48:52.425: seep send_capabilitieS_rSp_msg_vl: codec_rec->codec = 4
Apr 15 07:48:52.425: sccp_send_capabilities_rsp^sg_vl: mSg_cap->paYload_capS
Apr IS 07-48:52.425: seep send_capabilities.rspjnsgj/l: codec_rec->codec = 25B
Apr 15 07:48:52.425: sccp_Bend_capabilitieB_rsp_mSg_vl: mSg_cap->Payload_capS
Apfl5 07.4B:52.425: seep Send_capabil itieS_rSp_mSg_vl: codeo^reo---codec =257
Apr 15 07:48:52.425: scCp_send_capabilities_rsp_mSg_vl: mSg_cap->Payload_caps
Aprils 07:48:52.425: Sccp_send_caPabilities_rSp_mSg_vl: codec_rec->codec - 259
Apr 15 07:4B:52.425: sccP_send_Capabilities_rSp_msg_vl. msg_cap->payload_caps
ApflS 07.48:52.425: seep send_capabilities_rsp_mSg_vl. codec_rec->codec_= 261
Apr 15 07:48:52.425: sccp_send_capabi1itieS_rsp_mSgvl: msg_cap->payload_caPs
= 261
Another way to identify potential causes for RSVP agent (MTP) registration issues is to
generate asystem diagnostic interface (SDI) trace for Cisco CallManager service.
Voice Quality and Media Resources issues 6-129
Troubleshooting the RSVP CAC Operation
This topic describes thc major RSVP CAC issues that can be experienced at the call setup
process and explains how to troubleshoot them.
Phones are associated with their RSVP agents (MRGL).
RSVP signaling istriggered by anRSVP agent.
PATH explores IP routed path, RESV installs reservation
RSVP CAC is performed at every router hop enabled for RSVP,
Crsco Unified
Ccrrmunicaliona ]
Manager
The figure describes the operation of RSVP CAC for acall from headquarters to the branch
(BR) sites. For correct RSVP CAC operation, the following must be configured:
Calling phone that is associated with its locat ion-specific RSVP agent (HQ) throimh
MRGL and Media Resource Group (MRG).
Called phone that is associated with its location-specific RSVP agent (BR) rhrotmh MRGI
and MRG.
Regions must define the codecs that are used within and between sites.
RSVP agents must be registered to Cisco Unified Communications Manager, and they must
haveRSVPenabled on their WANinterfaces.
An RSVP state must exist in the WAN routers and adapt to ever changing IP network
conditions.
Depending on the codec that i-, selected, Cisco Unified Communications Manager assumes that
each call stream consumes the following amount ofbandwidth that must be reserved in the
WAN:
G.711 calls use 80 kb's.
G.722 calls use 80 kbs.
G.723 calls use 24 kbs.
G.728 calls use 26.66 kbs.
G.729 calls use 24 kb s.
Global System for Mobile Communications (GSM) calls use 29 kb/s.
6-130 Troubleshooting Cisco Unified Communications (TVOICE) u8 0
Wideband calls use 272 kb/s.
Internet Low Bitratc Codec (iLBC) calls use 24kb/s.
Advanced Audio Codec (AAQ calls use thc value that the video line specifies.
RSVP is usable with manv QoS control services, which, in turn, are intended to work with
more than one reservation' setup mechanism. The role of RSVP consists merely of distnbut.ng
QoS control data across the network and of passing this data to the traffic control module on
each host along the path. Because of the logical separation between QoS control scrv.ces and
the distribution ofQoS control information, the RSVP specifications define only the RSVP
QoS control objects, but the RSVP code treats the contents of these objects as opaque data.
RSVP objects are enclosed in RSVP messages, which are sent hop-by-hop between RSVP-
capable routers as raw IP datagrams with protocol number 46.
RSVP standards describe how to use acertain subset of the RSVP objects with the Controlled-
Load and Guaranteed QoS control services. These objects are encapsulated in the two
fundamental RSVPmessage types, Pathand Resv.
Path messages travel downstream from the RSVP sender (calling RSVP agent) to the set of
receivers following the same route as the voice packets. The Path messages store path state in
each node along the path. The path state currently consists of the unicast IP address of the
previous hop node. This information is used to route Resv messages in the reverse (upstream)
direction. Ifthere are any non-RSVP-capable routers in the path, the RSVP signaling is
transparently passed through such router.
Resv messages are sent upstream by receivers who want to establish areservation for one or
more senders Thc Resv messages follow exactly the reverse route ofthe voice packets, using
the path state that is created by the Path messages. The fundamental objects that are contained
in Resv messages belong to the FLOWSPEC, FILTER_SPEC, and STYLE classes. The Resv
message coming in from adownstream node is passed to the admission and policy control
modules. If they both grant the service request, the reservation state is set up in the node by
passing the F1LTER_SPEC object to the packet classifier and by configuring the packet
scheduler with the FLOWSPEC data. Note that this process does not necessarily produce anew
reservation. Resv messages with apotentially modified FLOWSPEC are forwarded upstream if
the reservation request is not already covered by previous requests from other receivers of the
same flow.
)2010 Cisco Systems. Inc.
RSVP CAC Operation Issues
This list presents the issues and their possible causes when using RSVP for CAC.
6-132
- Problem: Acall has been blocked, reorderlone heard:
If a calling phone shows not enough bandwidth, considerthe
following causes:
Too many reservations exist, RSVP bandwidth 15 exhausted.
IP phones arenot associated with RSVP agents, RSVP is not
configured or ismisconfigured onRSVP agents.
Current IP routed path goes via network segments where
RSVP bandwidth configured is insufficient for thiscall.
' RSVP per-flow bandwidth needs tobeset to 16kb/s more
than actual codecwill really require (initial reservation reauest
only). ^
Network connectivity to anagent or between agents islost
Problem: Acall is admitted, but no reservation was made for the call.
RSVP CAC not enabled.
Default "no reservation" policy applies.
The first ,ssue is the most common ,ssue that can occur. When RSVP is implemented acall is
blocked, and the reorder tone isheard at the calling party.
Consider several possible causes based on whether or not the "Not Enough Bandwidth-
message is d.splayed at the callmg phone. If the phone shows this message when reorder tone is
heard, consider the following:
RSVP CAC is an extension of location-based CAC at Cisco Unified Communications Manager
Depending on acodec that is configured at Regions, Cisco Unified Communications Manager
decreases the available bandwidth to and from the location by the amount that is required by
the codec, per each call that is being established. Ifthe location bandwidth is exhausted all
calls to and from this location are being blocked. AAR could be configured in addition lo
rerouting these calls over PSTN.
Ifyou are certain that the active calls do not exhaust the location total bandwidth
resynchronize the location bandwidth. When alink to alocation experiences blockage it can
result from bandwidth leakage that has reduced the usable bandwidth for thc location You can
resynchronize the bandwidth allotment to the maximum setting for the location without
restarting the Cisco Unified Communications Manager server. Choose the location that you
want ,0 resynehromze, display its Location Configuration page, and click Resync Bandwidth
to resynehromze the bandwidth for the chosen location. If calls are using the bandwidth for this
location vihen the bandwidth is resynchrom/ed, the bandwidth might be oversubscribed until
all calls that are using thebandwidth for thislocation disconnect.
Too many concurrent reservations at the location can exhaust the RSVP bandwidth Thc RSVP
bandwidth is configured separately from the locations bandwidth at the Cisco IOS router
interfaces. This RSVP bandwidth might be set too low, or it might be depleted by existing calls
ôu can either view the currently reserved bandwidth by using the Cisco Unified Real-Time
Monitoring Tool (R IMT) or Mew i, on the Cisco IOS router that includes Cisco [OS RSVP
agents. The show ip rsvp interface command shows how much bandwidth is allocated and
Troubleshooting Cisco Unifed Communications (TVOICE) v8 0
how much is allowed in total per Cisco IOS router interface. The following command output
shows 24 kb/s of currently allocated bandwidth and 128 kb/s of total bandw.dth for all RSVP
reservations:
HQ-l#show ip rsvp interface detail
Seu/1/0.121.
RSVP: Enabled
Interface State: Up
Bandwidth:
Curr allocated: 24K bite/sec
Max. allowed (total) 128K bitS/fseb
Max. allowed (per tlow) : 128Kbita/sec
Max. allowed for LSP tunnels using sub-pools: 0 bits/sec
Set aside by policy (total): 0 bits/sec
Admission Control:
Header Compression methods supported:
rtp (36 bytes-saved), udp (20 bytes-saved)
Traffic Control:
RSVP Data Packet Classification is ON via CEF callbacks
Signalling:
DSCP value used in RSVP msgs: 0x3F
Number of refresh intervals to enforce blockade state: 4
Authentication: disabled
Key chain: <none>
Type: md5
Window size: 1
Challenge: disabled
Hello Extension:
State: Disabled
Acall would be also blocked ifRSVP is not configured on an RSVP agent or ifitis
misconfigured in terms of total RSVP bandwidth, or per-flow bandwidth. The following
configuration shows thc RSVP-enabled interface with 128 kb/s of total RSVP bandwtdth and
40 kb/sof bandwidth thatanindividual flow is allowed to reserve:
interface SerialO/1/0
ip rsvp bandwidth 128 40
RSVP reservation follows the path that is selected dynamically by an IP routing protocol. If
link failures occur in the network, an IP routing protocol can route the reservations through
network segments in which the RSVP bandwidth that is configured is insufficient to support the
call. This is only atransient situation, and when the links are restored, the IP routing protocol
will put the path back.
To make an initial RSVP reservation for acall, Cisco Unified Communications Manager adds
16 kb/s to the bandwidth ofthe selected codec for the expected signaling traffic. This applies to
the initial reservation request. During the RSVP signaling process, the bandw.dth is adjusted to
the amount that is equal to the codec requirements. For instance, ifG.729 is selected for acall,
the initial reservation request asks for 40 kb/s, but shortly thereafter, the bandwidth is
readjusted to 24 kb/s. If you configure RSVP bandwidth that allows 24 kb/s per flow, the CAC
process will return afailure-even when the final reservation is installed for 24 kb/s. RSVP
per-flow bandwidth must be set to the bandwidth of the expected codec plus 16 kb/s for the
RSVP CAC to work properly.
Voice Quality and MediaResources Issues 6-133
2010CiscoSyslems, Inc *
If network connectivity to an RSVP agent is lost, or if the network connectivity between two
agents is broken, the call will be blocked with a reorder tone as well.
Another problem could potentially occur with RSVP CAC. Acall has been admitted but no
reservation ,s made tor the call that isusing RSVP. The possible causes could be that either
RS\ PCAC is not enabled at Locations, or the default RSVP policy applies. To configure the
clusterwide RSVP policy, configure the Default Interlocation RSVP Policy service parameter
in the Cisco CallManager sen ice. You can set this service parameter to the following values:
No Reservation: No RSVP reservations are made between any two locations.
Optional (Video Desired): Acall can proceed as abest-effort, audio-only call iffailure to
obtain reservations tor both audio and video streams occurs. The RSVP agent continues to
attempt RSV Preservation tor audio and informs Cisco Unified Communications Manager
it the reservation succeeds.
Mandator) :Cisco Unified Communications Manager does not ring the terminating device
until the RS\ Preservation succeeds for the audio stream and, ifthe call is avideo call for
the video stream as well.
Mandatory (Video Desired): Avideo call can proceed as an audio-only call if a
reservation tor the audio stream succeeds but areservation for the video stream does not
succeed.
-134 Troubleshooting Cisco Unified Communications (TVOICE) v8.0
Verify RSVP CAC Performance
This figure shows how to verify the RSVP CAC performance by using Cisco Unified RTMT.
In its performance section, several performance variables have been defined.
Verify RSVP CAC Performance
dene
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-3
The figure shows that an RSVP agent that is associated with the location branch can reserve up
to 128 kb/s oftotal bandwidth. Currently, there isan active single reservation.
For troubleshooting purposes, the OutOfResources variable is very useful because it shows how
many RSVP agents have failed to make reservations because of insufficient bandwidth. Similar
inspections can be made ifRSVP TotalCallsFailed is displayed.
RSVPCAC Debug
6-136
This section describes the debug output that was taken during the RSVP CAC
HQ-l#debug ip revp resv ~~~~~"~~"""~""
*SVPi 1O.1.J5O.102 lM->m.l.250.101 .1H1010.0.0.01: ,t.rt requesting ,0
kbps FF reservation on Seri.10/1/0 .121, neighbor 10.1.6.102
RSVP: 8eBBiQD 10. 1.550.102 19281 [0.0.0.0] : Heceived Kesv meSBaae from
10.1.6.102 Ion SerialO/1/0,J21)
HSVP-RESV: Admitting nEv reservation: 166CE1F0
RSVP-RESV; reservation was installed, 466CE1F0
. . .truncated. . .
KSVP: session 10.1.250.102 1928110.0.0.0]; Received Resv meaaage rotn
10.1.6.102 (on SerialO/l/o,121} W
RSVP; 10.1,250.101 19330-^10.1.250.102 1928410.0.0,0]- Resv chanoed-
RSVP: 10.1.250.101 19330-.10.1.250.102 19284[0. 0. 0. 01:
process reservation change: Resv change requires triggering of Resv upstrea,
ESVP-RESV
KSVP-RESV
on change: 466CB1P0
installed: 46 6CB1F0
RSVP: io. 1.250.102 19 284 -=10 . 1.250 .101 193 3010 .0.0. 01 : start revesting 24
Kbps FP reservation on Serialo/1/0.121, neighbor 10.1.6 102
RSVP: 10.1.250.102 19281->10, 1.250.101 19330IO.0.0.O]: Resv refr
config: 30000 curl: 30000 xmit: 3000D
RSVP: 10,1.250.102 19284->10.1.250.101 19330[0.0.0.0) sending R
to 10.1.6.102
sh (n ),
process.
Because RSVP Resv messages arc installing the reservations, the debug ip rsvp resv command
is used io generate this debug output.
Acaller that is associated with an RSVP agent 10.1.250.101/10.1,6.101 (originating) has placed
acall to the party that is associated with an RSVP agent 10.1.250.102/10.1 6102 (tcrminatim.)
by using the G.729 codec.
Note that the debug output stans with thc terminating RSVP agent that initially requested 40
kbs (24 kb sotG.729 f 16 kb's). This reservation has been admitted and installed.
Then, another Resv message was received that brings in the FLOWSPHC objeci. FLOWSPEC
objects describe the desired resource reservation parameters ofthe receiver.
Now the reservation has been modified to match with the G.729 bandwidth requirements (24
KD S).
While the tigure shows the successful RSVP reservation, the following output .hows a
reservation error because ofinsufficient per-flow bandwidth. Note that the initial bandu idth
requested is 96kbs for thissingle G.711 call,
HQ-l#debug ip rsvp resv
iesaa0^!--'2"; "SV?: SeSE10n "1-25-ll-"B58{0.0.0.0]: Received Resv
menage i.tJ 1''.j.j.1 len lecciver host)
Apr 15 09:22:25.256: RSVP :10.I.2->0 .102_] 654 6-,10.1.250 .101 16858[0 0 I* 0j-
bjccesd.ully parsed Resv message from 127.0.0.1 (on lerPivor has-'
Apr IS 09:22:25.256: RSVP-MSG: 10.1 .250 .102_16546
>1O.i.250.1G1_16858 [0.0.0.C] : :io matching path state for Resv
Apr 15 09:22 =25.264: RSVP: session 10.1.250,101_16858[0.0.0.0j: Rs.ce-.ved Resv
message :ium -27.0.:*!.; ion receiver host}
Apr lb 09:22:25.268: RSVP: 3C.1.2bD.102_16546->10.1.2b0.101_16858[0 0 C nj .
buccessfu_ly parsed Resv -.cssage from 127.0.0.1 (on receiver host)
Apr li 09:22 :2b .268: RSVP RESV: Admitting new reservation: 466CklJSS
Apr 1- 09:22:25,29: KSVP-ÊSV: Locally created reservation No
auîssicr.'-.raff :c centre; needed
Troubleshooting Cisco Unified Communications (TVOICE) vS.O
Apr 15 09-22-25.268: RSVP: 10.1.250.102_1S546->10.1.250.101_1G85B[0.0.0.0]:
tart requiting 96 *bp* reservation on SerialO/1/0.121, neighbor
ADr115'09222-25.26B: RSVP: 10.1.250.102_16546->10.1.250.101_168BB[0.0.0.0].
start requesting 96 kbps FF reservation on SerialO/1/0.121, neighbor
Apriib'o9:22:25.272: RSVP: session 10.1.250.102_16546[0.0.0.0!. Received ReSV
messaqe from 10.1.6,102 (on SerialO/1/0.121)
J! tl 09-22-25 2^2: RSVP: 10.1.250.101_1GB5B->10.1.250.102_1654G [0.0.0.0] :
Successfully parsed Resv message from 10.1.6.102 (on Serial 0/1/0.121)
Apr 15 09:22.25.272. RSVP: 10.1.250.101_16856->10.1.250.102,16546 [0.0.0.0] :
reservation not found--new one _ ,rrr-nmr.
ADr 15 09-22-25 272: RSVP-RESV: Admitting new reservation: 466CE1F0
Apr 15 09^22:25.276: RSVP: 10.1.250.101_1685B->10.1.250.102_16546 [0.0.0.0] :
building error spec object with err-node addr: 10.1.6.101
Apr 15 09:22:25.276: RSVP: 10.1.250.101JL6858->10.1.250.102_16546[0,0.0.0]:
Sanding ResvSrror messag* to 10.1.6.102 ,cc^,cn
Apr 15 09:22:25.276; RSVP-RESV: reservation was not installed: 46bLfc.ii: u
Anr 15 09-22-25 280: RSVP-RESV: Deleting reservation: 466CE1F0
Apr 15 09-22-25.280. RSVP: 10.1.250.101_16858->10-1.250.102_16546[0.0.0.01 :
Exoirinq SerialO/1/0,121 RESV state, reason: Traffic control error
Apr 15 09 22-25 280: RSVP: 10.1.250.101 16858->10.1.250.102_16546[0.0.0.0]:
Expiring SerialO/l/0.121 RESV state, reason: Traffic control error (17:16858)
Apr 15 09:22:25.280: RSVP: 10.1.250.101_16858->10.1.250.102_16546[0.0.0.0]:
Sending PathTear aesaage to 10.1.250.102
Apr 15 09:22:25.284: RSVP: 10.1.250.102__16546->10.1.250.101^16858[0.0.0.0]:
Refresh RESV, req=466D33A0 [cleanup timer is not awake]
To troubleshoot an interaction ofRSVP PATH and Resv messages, use the debug iprsvp
signalling command. This debug output shows all events that are related to both RSVP
messages. The following debug output shows asuccessful reservation:
H0-l#debug ip rsvp signalling
Apr 15 08 26 38.783: RSVP: 10.1.250.101_190B2->10.1.250.102_19192 [0.0.0.0] :
Received Path message from 127.0.0.1 (on sender host)
Apr 15 08-26-38 783: RSVP: new path message passed parsing, continue...
Apr 15 08:26:38.787: RSVP: Triggering outgoing Path due to incoming Path
change cr new Path
ADr 1= 08-26-3B 787: RSVP: Triggering outgoing Path refresh
Apr 15 08:26:38.787: RSVP: session 10.1.250.101_19082[0.0.0.0]: Received Resv
mpssaae from 127.0.0.1 (on receiver host)
Apr Tos.26-3B.791: RSVP: 1G.1.2SQ.102_19192->10.1.250.101_190S2 10 .0.0.0] :
Successfully parsed Resv message from 127.0.0.1 (on receiver host)
Apr 15 0B:26:38.791: RSVP-MSG: 10.1.250.102_19192-
>10 1 250 101 19082 10.0.0.0]: no matching path state for Resv
Apr'l5 08-26-1=3.791: RSVP: 10 .1.250.101_19082->lO.1.250.i02_19192 [0 .0.0.0] :
Path refresh, Event: rmsg not enabled or ack revd, State: trigger to normal
Apr 15 08-26:38.791: RSVP: 1Q.1.250.101,190B2->10.1.250.102_1919210.0.0.0]:
Path refresh (msec), config: 30000 curr: 30000 xmit: 30000
Apr 15 08:26:33.791: RSVP: Triggering outgoing Path due to incoming Path
change or new Path
ADr 15 oe-26-38.791: RSVP: Triggering outgoing Path refresh
Apr l' 08-26-38.795: RSVP: 10.1.250.10i_19082->10.1.250.102_I9192[0.0.0.01 :
Path refresh, Event: rmsg not enabled or ack revd, State: trigger to normal
Apr 15 08-26-38 795: RSVP: 10.1.250.101_19082->10.1.250.102_19192[0.0.0.0]:
Path refresh (msec), config: 30000 curr: 30000 xmit: 30000
Apr 15 08:26:38.795: RSVP: 10.1.250.101_19082->10.1.250.102_19192 [0 .0.0.0] :
Sendinq Path message to 10.1.250.102
Apr 15 03:26:38.795: RSVP: 10.1.250.101_19082 ->10.1.250.102_19192[0.0.0.0] :
buildinq hop object with sre addr: 10.1.6.101
Apr 15 08-26:38.799: RSVP: 10.1.250.102,19192-,10.1.250.101,19082[0.0.0.0] :
Received Path message from 10.1-6.102 (on SerialO/1/0-121)
Aor 15 ob-26-38 799: RSVP: new-path, message passed parsing, continue...
Apr 15 08^26:38.799: RSVP: session 10.1.250.101_19082 [0.0.0.0] :Received Resv
message from 127.0.0.1 (on receiver host)
Apr l! 08-26-38.B03: RSVP: 10.1.250.102_19192->10.1.250.101_19082[0.0.0.0] :
Successfully parsed Resv message from 127.0.0.1 (on receiver host)
Apr 15 08:26:38.B03: RSVP-RESV: Admitting new reservation: 466CE35B
)2010 C,sco Systems, Inc v-ce 0uali*y and Media Resoljrces lssues 6'137
6-138
Apr 15 08:26:38,803: RSVP-RESV: Locally created reservation No
aamissicn/traff;c control needed
Apr 15 08:26:38.803: RSVP: 10.1.250,102_19192 ,10.1,250,101 1908?[0 0 0 01-
10 l6rS6 94 kbP8 FF reservatiort Serial0/l/0,i2l,"neighbor '
Apr 15 08:26:38.803: RSVP: IC.1.250.102_19192- >10.1 .250 .101 1908210 00 0]-
loTfiôT S" PS ^ reSerVation otl SerlalO/l/0.121,"neighbor '
Apr 15 08:26:38.807: RSVP: session 10.1.250.102 19192[0.0.0.0] -Received Resv
message from 10.1.6.102 (on Serialo/l/o.121) " "eceived Resv
Apr 1, CS:26:3S.B07: RSVP: 1C.1 .250.!01_19082 ,10.1.250.102 19192[0 C C 0'-
..^Cfcss-ily parsed Resv message from 10.1.6.102 (on SerialO/1/o 1211
Apr 1, 0^26:36.807: RSVP: 10.1.250.101 19082 >10.1 ,250.102 19192J0.0 0C]-
ie3eiVd;.]or. not :cjna- new one " u.u.M.
Apr 15 03:26:38.8C7: rsvp-rhsv: Admitting new reservation: 466CE1FC
Apr 1= ,,:,6:38.811: RSVP-RESV: reservation WaS installed: 466CE1F0
pr .5 =,8:20^8.811: RSVP: 10.1. 250. 102_19192->10.1.250. 101 19082[0 C 0 0^-
Re_resr. k^.,, req .466D33A0 (cleanup timer is not awake] " ' ' "
Re"vie-^h"8'ell: RSVP: 10-1-250.102..l9192->10. 1.250. 101 19082 [C .0.0.0-
, V,:etlL'sh' t-vent: rmsg not enabled or ack rovd. State: --"rigger to -;
*ll 1S ;B = =3fl-eil= RSVP: 10.1.250.102__l9192-ilo.1.250.101 19082 [0 0 0"0T:
6j -t*resr- 'rsec:, config: 300C0 curr. 30000 xmil : "JOOOO
Sendi^0-126138,81" RSV?: 10-1-250-102-^2->10.1.250.101 19092 10 .0 .0.Dj -
Sending Kesv message to 10,1.6.102 "
Apr 15 08:26:38 811: RSVP: 10.1.250.102_19192->10.1.250.10t 19082[0.0.0 C<-
ouiia.ng nop object wit:: sre addr: 10.1.6.101
melaînm:42"9 ^ ' ^"^ J*X""*]0i-1^82 [0 .0.0.0;:Received Resv
message Liom _2;.iJ.v,i lon teceiver host!
Apr lb 08:26:40.843: RSVP: 10.1.250.102 19192 ->10 ,1.250.101 190S2[G C C 01-
S^cess.u.ly parsed Resv message from 127.0.0.1 (on receiver host)
X. ~,:!;1:4-r^;^SVP: 1C,*1-250.1C2_19192- =10.1.2b0.101 19082 [0.C,0.0::
Kt,v nangea: i-LQwiPEC,
Api 15 08:^6:40.843: HSVP: 10.1.250.102 19192 ->10.1.250.101 19082[0 0 0 0'-
ADC-?-r^r4-r^-C^rge: RSSV Chan9e ^^."iggering'of Resv upstream
Apr ,5 .,8:26:40.841: RSVP-RESV: accept reservation_char,ge: 466CF358
Apr .5 nfa:26:4C.843: RSVP-RESV: locally created reservation No
aamissior., t:aftio control needed
Apr 15 08:26:40.843: RSVP: i0.1.250.102_19192 ->10.1.250.101 19082[0 00 0]-
, r-/eêst-r-9 24 *bps FF reservation on Serial 0/1/0 .121, ""neighbor
Xl't C';i6,:4='8": RSV?: 1C,'1-2bC-i02-1"92->10.1.250.101 19082(0.0.0.0"
Re. refer. Rt-W, req-466D33AC (cleanup timer is not awake]
pr .5 08:26:40.8*7: RSVP: 10.1 .250 .102. 19192 ->10 .1.250.101 19082[0.0 0 0]-
Ar,=e'ce^".fV^: f9 n0t nabled r aCk rCVd' StaLR: L-*39" to normal
"P" " -6:^6:^.847: RSVP: 10 ,1.25 0.102. 19192 ->l 0.1.250 .101 19082[0.0 0 Di
kes.' -eêsn ..msec, config: 30000 curt : 30000 xmit- 30000 "
Apr 15 08:26:40.847: RSVP: 10.1,250,102_19192 ->10.1.250.101 1908210 0 2 11
bending Resv message to 10.1.6.102 ~ '
Apr 15 08:26:40.84- RSVP: 10.1 .250.102_19192-.10. 1.250.101 19082(0 0 0 0-
td-ldmg nop ociect with sre addr: 10 1 6 101 " ' '"
Apr IS 06:26:40.887: RSVP: session 10.1.250.102 19192(0.0.0 0]- Received Resv
message from 10.1.6.102 (on SerialO/l/0 121) received Resv
Apr 15 08:26:40.897: RSVP: 10.1.250.101_19082 >10 .1.?50.102 19192[0 0 0 0'-
Successful parsea Resv message from 10.1.6.102 {on SerialO/1/0 12^ ' '
Apr 15 08:26:40.887: RSVP: 10.1.250.101 19082 ->10.1.250.102 1919210*0 ' 01
Resv changed: FI.OWSPEC, ' ' ' ''
Apr 15 08:26:40.887: RSVP: 10,1.250.101 19082 ->10 .1.250 .102 19192[0 C0 01-
Aprjr\8e;6-"-T88"-CKvpCRE^SV Cha09e reqUltSS "189-ri-ng-of Resv'upstrean
Apr 15 id.*6.,0.68.-: RSJP-HESV: accept_reaervat ion_change : 466CE1F0
Apr is .8:26:40.887: RSVP-RESV: reservation was installed: 46SC&1F0
If RSVP reservation fails, you can verify ifRSVP-enabled routers can exchange RSVP
signaling. Use the show ip rsvp neighbor command to display the known neighbors with
RSVP CAC enabled.
HQ-l#show ip rsvp neighbor
Neighbor Encapsulation Time since msg rcvd/sent
10.1.6.102 Raw IP 00:00:00 00:00:03
10.1.250.102 - never 00:00:05
Voice Qualily andMedia Resources Issues 6-139
RSVP Reservation Installed
This figure show, the RSVP reservation that has beett successfully installed for acall that uses
the ti.729 codec.
H0-1#S!.(
stalled detail
RSVP= s.rl.lo/1/O.ui aa th. f0ll9 ln.t.u.d reservation
SSVP Seseruation. Destination is 10.1. 250.IC 2. Source is 10 1 250 i
Protoscl is UDP, Destination por, is i9284, Source port ia 1933Q
Traffic Control ID handle: 18000406
Created- 09,17:55 CET Thu Apr 15 2010
Admitted flowspei;
2SK b!ts/set band"ld"h: liK "--"/mc. xl-u- burst: 120 bytes. Pea
Mm Policed Unit: 60 bytes. Max Pkt Size: 60 bytes
Resource provider for this flow: None
Conversation supports 1 reservations [OxlD000401]
Data given reserved aervice, 7702 packets (462120 bytes)
Data given beat-effort service, 2 packets (120 bytesl
Reserved traffic classified for 156 seconds
Long-te average borate (bits/aec): 23569 reserved, 6 bset-effo
Policy- INSTALL. Policy aouice(s). Default
The reservation extsts between the two RSVP agents: 10.1.250.101 (originating) and
10.1.2M).1<)2 (terminating). Thc reservation has been installed at the local interface
SerialO I 0.121 of theagent.
This command output also shows traffic statistics for the current reservation.
You can also get compact, brief output that shows the same reservation bv using the show in
rsvp installed command: '
HQ-lushow ip rsvp installed
xSVP: Sei-ialG: 1 ' Z 121
BP ^ From
24K 10.1.250.102 10.1.250.101
Both commands show areservation in the full-duplex format. To see reservations for asingle
call ,n opposite directions separately, use the show ip rsvp reservation command It shows
that essennally two reservations of 24 kb/s were made in opposite directions. This command is
useful menvironments ,n which asymmetric reservations cm be made (music on hold IMOHI
streaming). l J
HC-lffshow ip rsvp reservation
!? n ., -"rOTr Prc DPoir- Sport Next iop l/F F, Set- 6>
;'"'' "x i'J-1 ^lc; ">U? 19082 19192 none no-ie ff - on -^w
10-:.2,0.10, 10.1..5C.101 UBF 191,2 i,082 10.1..,102 SeOA/0 . FF S 2^K
6-140 Troubleshooting Cisco Unified Communications (TVOICE) v8 i
Protoc DPcrt Sport
UDP 19284 19330
VRF
Troubleshooting Intercluster RSVP with SIP
Preconditions
This topic reviews thc intercluster RSVP (known as Session Initiation Protocol [SIP]
Preconditions) feature, describes major issues that can be experienced when using it, and
explains how to troubleshoot the issues.
Intercluster RSVP CAC with
Preconditions Review
For SIP calls going out ofthe cluster. RSVP can beused end-to-end
between different domains.
In case ofCisco Unified Communications Manager, RSVP agentofthe
phone is used (nottheone ofthettunk).
Cisco IOS gateways and Cisco Unified Communications Manager Express
also support SIP Preconditions.
SIP Trunk
SCCP
RTP
- - RSVP
flnalru) or Digital Voice
Use end-to-end RSVPwhenall of these conditions are met:
Bothends of a SIP trunksupportSIPPreconditions
The IP phone and the SIP trunk are indifferent locations
RSVPis enabledbetween thesetwo locations
This means that only the RSVP agent that is associated with the IP phone is invoked; there is
no second local RSVP involved. The RSVP agent ofthe phone now uses RSVP-based CAC to
the other end of the SIP trunk.
Ifthe other end isanother Cisco Unified Communications Manager cluster, then the same
happens at that far end: only one RSVP agent is invoked. Ifthe other end is aCisco IOS router,
then that router (either Cisco Unified Communications Manager Express or Cisco IOS SIP
gateway) terminated RSVP atthe far end.
With SIP Preconditions, RSVP isnow virtually end-to-end -it spans the two call-routing
domains and is not limited to the local cluster.
Note Because of proprietary extensions, the SIP Preconditions for RSVP-enabled CAC is
currently supported only between Cisco Unified Communications Manager, Cisco Unified
Communications Manager Express, and Cisco IOS SIP gateways. Third-party SIP devices
are currently not supported. ^
2010 Cisco Syslems, Inc.
The intercluster RSVP CAC with SIP Preconditions is built on the capabilities of intracluster
RSVP CAt ot Cisco Unitied Communications Manager, Thc feature is based on RFC 33P
Integration of Resource Management andSIP. In addition to Cisco Unitied Communications
Manager, it supports audio RSVP reservations via SIP trunks to Cisco RSVP-enabled Cisco
IOS products such as Cisco Unified Communications Manager Express and SIP time-division
multiplexing (TDM) gateway.
The intercluster RSVP CAC is signaled by using SIP Preconditions. While RFC K1iallows
tor other types of precondition signaling, Cisco Unified Communications Manager currently
implements RSVP. J
The intercluster RSVP CAC with SIP Preconditions is configured by using the following
elements:
The SIP trunk is configured between the two Cisco Unified Communications Manager
clusters. The feature is enabled at the SIP Profile that is associated with the SIP trunk by
setting the RSVP over SIP field to E2E and by enabling SIP Prack method to be sent on the
SIP trunk. The RSVP Over SIP field chooses thc method that Cisco Unified
Communications Manager uses toconfigure RSVP over SIP trunks:
Local RSVP: In a local configuration, RSVP occurs within each cluster between
the endpoint and thc local SIP trunk, but not on the WAN link between the clusters.
E2E: In an end-to-end (E2E) configuration, RSVP occurs on the entire path between
the endpoints. including within the local cluster and over thc WAN,
The SIP RellXX Options field configures SIP RcllXX, which determines whether all SIP
provisional responses (other than 100 Trying messages) are sent reliably to the remote SIP
endpoint. Valid values are thefollowing:
Disabled: Disables SIP RellXX.
Send PRACK if IXXcontains Session Description Protocol (SDP):
Acknowledges a 1XX message with PRACK, only ifthe 1XX message contains
SDP.
Send PRACK for all 1XX messages: Acknowledges all 1XX messages with
PRACK.
The SIP trunk is associated with alocation that is enabled for RSVP and aregion that sets up a
codec that is used for intercluster RSVP calls. The calling side region determines the codec that
is used on the call. In the figure, the location and the region named Intercluster is used.
IP phones are associated with an RSVP-enabled location that is separate from the SIP trunk
location Acall in such asituation triggers thc RSVP reservation process. Originating and
terminating IP phones must be associated with their local RSVP agents, like those for
intracluster RSVP CAC.
For RSVP CAC to work, IP WAN must be enabled for RSVP- thus, enabled to set up a
bandwidth pool for RSVP reservations.
Cisco Unified Communications Manager allows failed calls to fall back to alocal RSVP. Mark
the SIP Profile Fall Back to aLocal RSVP check box ifyou want to allow failed end-to-end
RSVP calls to fall back to the local RSVP to establish the call. Ifthis box is unchecked, end-to-
end RSVP calls that cannot establish and end-to-end connections will fail. Also, for fallback to
local RSVP to function properly, the SIP trunk must have an MRGL that is configured w.th
access to aseparate RSVP agent from the one that is used for the phones.
Cisco Unified Communications Manager only accepts calls from the SIP device whose IP
address matches the destination address ofthe configured SIP trunk. In addition, the port on
which the SIP message arrives must match thc one that is configured on the SIP trunk. After
Cisco Unified Communications Manager accepts thc call, Cisco Unified Communications
Manager uses the configuration for this setting to determine whether the call should be rerouted
to another trunk. The field Reroute Incoming Request to New Trunk Based On within the SIP
profile controls this behavior, and you can choose the method that Cisco Unified
Communications Manager uses to identify the SIP trunk where the call is rerouted:
Never: Ifthe SIP trunk matches thc IP address ofthe originating deviee, choose this
option which equals thc default setting. Cisco Unified Communications Manager, which
identifies the trunk by using the source IP address of the incoming packet and thc signaling
port number, does not route the call to adifferent (new) SIP trunk. The call occurs on the
SIP trunk on which the call arrived.
Contact Info Header: Ifthe SIP trunk uses aSIP proxy, choose this option. Cisco Unified
Communications Manager parses the contact header in the incoming request. It uses the IP
address or domain name and signaling port number that are specified in the header to
reroute the call to the SIP trunk that uses the IP address and port. Ifno SIP trunk is
identified, the call occurs on the trunk on which the call arrived.
Call-Info Header with purpose=x-ciSco-origlP: Ifthe SIP trunk uses aCisco Unified
Customer Voice Portal (CVP) oraBack-to-Baek User Agent (B2BUA), choose this option.
When the incoming request is received, Cisco Unified Communications Manager parses the
Call-Info header, looks for the parameter (purpose=x-cisco-origIP), and uses the IP address
or domain name and the signaling port number that is specified in the header to reroute the
call to the SIP trunk that uses the IP address and port. Ifthe parameter does not exist in thc
header or no SIP trunk isidentified, the call occurs on the SIP trunk on which the call
arrived.
,2010 Cisco Systems, Inc. Voice Qualit* and Media ReS0rCeS ^^ ^
Issues of Intercluster RSVP with SIP Preconditions
This section lists the issues that are experienced with intercluster RSVP.
6-144
t't, n ^.^^B^4^ U- -Urf,
IT 2 }
Problem: Acall is blocked; a reorder tone is heard.
If a calling phone shows not enough bandwidth, consider the
following causes-
Bandwidth configured at anyofthe locations affected
might be exhausted
Too many reservations existbetween clusters' RSVP
bandwidth exhausted
IP phones are not associated with RSVP agents.
RSVP Is not configured orIs misconfigured on RSVP
agents. All RSVPconfiguration issues of intercluster
RSVPwould apply here as well.
*Network connectivity toRSVP agent or between clusters lost.
Possible dial plan or digit manipulation issues at either
cluster.
lortroubleshooting intercluster RSVP CAC calls, you can use the same mechanisms and
troubleshooting tools as for an intracluster RSVP CAC. The only fundamental difference
between these two ,s aSIP trunk that is enabled for RSVP CAC functions. Also, the issues that
are experienced with intercluster RSVP CAC are like those ofintracluster RSVP CAC.
The most common issue is acall being blocked and areorder tone that is heard by the caller If
thc calling phone shows the message "Not Enough Bandwidth," consider these most likclv
causes:
For atypical RSVP CAC-controlled intercluster caii, at least three locations must be
configured at Cisco Unified Communications Manager: the originating phone location the
originating SIP mink location, and the terminating SIP trunk location. Acall will ifthe
bandwidth that is configured atany ofthese three locations isexhausted,
RSVP is supported by alimited number ofbandwidth resources in the IP WAN Iftoo
many reservations exist in the IP WAN and thc RSVP bandwidth is exhausted, the call will
tail due to insufficient bandwidth.
Both IP phones that participate in an intercluster RSVP CAC-controlled call must be
associated with their local RSVP agents. Ifone ofthe phones does not have aproper
association, the call will fail.
RSVP must be enabled and properly provisioned on RSVP agents for acall setup to
succeed, as mentioned in the previous topic. Essentially, all RSVP configuration issues that
were mentioned in the intercluster RSVP CAC section would apply here as well, because
the same control and data plane is used to carry on the call.
In addition to the causes that are related to bandwidth resources, acall will also fail ifnetwork
connectivity ro alocal RSVP agent, or between the agents, or between participating clusters is
lost. Make sure that the network connectivity is maintained.
Other causes that are not directly related to RSVP CAC, but possibly can affect the success of a
call setup are Cisco Unified Communications Manager dial plan issues that include the issues
of partitions and CSS, or digit manipulation issues at either participating cluster.
Tracing Successful Call Setup with SIP Preconditions
This section discusses how to use trace for asuccessful intercluster call setup that uses SIP
Preconditions.
Tracing Successful Call Setup with SIP
Preconditions
Stationlnit = 10000001) SoftKeyBvent s . tt JTi-j-E ,<.=: i r*'<!=-'. 1.1' linalnstanca.O
callReference-O. 1.100,49.1.1127'10.1.2.ll-SEP0024C4454ftD8
Digit^nll'.is! nmtCh(pi--2"= fqcn-"5115552001". cO--2001".plv--5-,
osb.-HQ Emergency Pt:HQ Intl Pf.aQ LD_Pt:HQ_I=ocal_Pt =Internal_Pt" .
VodFilt^radPBB.-HQ Emerg.m.y Pt:HQ_IDtl Pt:HQ U,.M,B.Lol.Pt.IMll.l.tf
;,-.;;.:.;; .dac-lM l, 100,49. 1.1135"10 .1. 2.ll'SEFO02 4C4454AD8
Fi-ci .-<="!' . '- =;MatchCapabilitiea -- '.! p=s -',-=, capACOuntrl2,
capBCOunt.24 '**
v* ..,!;., .-.,;(=> ;.1jc.yr::Miting_HiinAllocata!lSVPBesoijreeBe<3 - RflvpPolicy-3 .
CI-30738107, Resource RagueBtaa-l CapCountA-12, EapCout.tB.24, "iW'K,
r.gionB.IntM-c: luster, SuppresBioo-4, d.viceCapbiHty-272, TryPwaThru-l,
IBEMCCD evice-0 I1,100,49,1.1135-10.1.2.11-SBP0024C4454M8
MEM: .convertScmStringToStdString H3VP-H0 Kg
1,100, 49, 1.113 5*10 .1.2. irSEP0024C4454ADB
HBdi^Bo"=eCdpC19)=:...--.-:.=' - Narce-RSVP-HQl type- SiaTRPHuat-l
eight. 0 1,100,49,1.113S'10.1.2. ll*SBP002 4e4*54M>8
.di^^rCeCdpc(9);;aendRBvpftllOcatBRe(jueBtI0Device RavpRsBouros-RSVP -HQ1
1,100,49,1.113 5*10.1.!.ll*SEP0O!4C44 54ADS
v^'- =. .i-r--=,- -=-.(-' . !:-,=-.i.rcl (!) [ :logRafloureeStatuainTraca -- Davico
Name.R3VP-HQl RaBourcaAvailble-9
-=-=.;,:- --.:-,' i. ,'1.100,49. 1.1135*in.l.2.11"3EP0Q24C44 54M>9
The beginning of the trace output shows the NewCall softkey event, and the digit analysis
output shows that thc called party number 8022001 has been dialed.
Then regions set the codec to G.722 (shown as kbps-64, codec type is not mentioned).
Media Resource Manager (MRM), which is based on the RSVP requirement that is set in the
locations, shows the MRG RSVP-HQ_mrg as the one used for the RSVP agent selection. The
RSVP agent name isRSVP-HQ1 for this intercluster call.
The successful attempt to allocate the RSVP agent resource is shown at the end ofthis trace
output.
Voice Quality and Media ResourcesIssues 6-U5
6-146
//SIP/SIPTcp/wait SdlSPI Signal:
5063 index 1
ige to 10.2.1.1 on port
sip:200iei0.I. 1.1: 50=50 SIP/2.0
Via: SIP/2.0/TCP 10.1.1.1:50S0;branch=i9hG4bK134b93593f
,.,.;'; ' ' >'t9=ae40d4d.Sde5-428-b79-2614e6ea9d44-
JO'J o10 5
. . -truncated. . .
Supported: X-rri3co-srtp-allbac*
Supported: Seolccation
Call-in fos taip:10.1. :. 1: 50tSO>;met hod= "NOTIFi; Event-telephone
eventjCuration=500"
Bequire: precondition
Cisto-Guid: 066S4S15S1-3209777095- 0000000003-0016843018
=CiscoSystemsCCH-SIP 2000 2 IN IP! 10 1 1 1
s=SIP Call
S=IN IP4 10. 1,250.101.
t=0 0
U1=. audio 16760 RTP/AVP 9 0 S 116 IS 101
===dea:qc=j mandatori' c2=< sendrpcv
artpmap: 9 Q722/BO0O
a=pt;ice :20
This figure -.hows Cisco Unitied Communications Manager formulating an INV1TF message
and sending it to the remote cluster. This INVITE message indicates the requirement of SIP
Preconditions to the remote cluster.
The precondition tag can be included meither the Require or Supported header. When the
RSVP policy is set to Mandatory, the precondition tag is included in the Require header (as
shown m.his figure). When thc RSVP policy is set to Optional, the precondition tag is included
in the Supported header. When the RSVP policy is set to Mandatory (video desired) the
precondition tag is included in the Require header. When the RSVP policy is set to None thc
precondition tag is not included in cither header.
Note
The called parly number was modified through direct inward dialing (DID) and the site code
802 was removed. To the calling-party number on the other side, the site code 801 was
added to present the correct calling number at the remote cluster.
The body ofthe INVITl: message (SDP) shows the two attributes that arc related to the SIP
Preconditions:
curnqos indicates thc current QoStype and direction.
des:qos indicates the desired QoS type and direction.
Thc originating SIP user agent (UA) (Cisco Unified Communications Manager) sends the
RSVP agent address (10.1.250.101) in the body of the initial INVITE message.
>2010Cisco Systems. Inc.
*f*~
Tracing Successful Caii Setup with SIP
Preconditions (Cont.)
//SIP/SIPTcp/wsit SdlP.eadP.spi '"='
5060 index 1 with 329 bytes:
SiP/2.0 .' ii -;..;
Via= SIP/2. 0/TCP 10.1. l.l:5060|brsnch.i9h<*J4bK134b93E93r
//SlP/SiPTcp/-it 9dlReadRsP: rn3 *I? TCP message from 10.2.1.1 on port
5O60 Index 1 with Hi! bytes:
sip/2.0 .:: S'.s-.--r- 'f"-"'
Via- BIP/2 0/TCP 10.1. l.l:5060ibrancti-i9hG4bK134b93S93r
Fron: <sip: BOlIOOielO. 1.1.1=;tag>ae40d64fl- 5de5-4t2d-bTf9 -2=>lte6ea9d44 -
To =3"iL20010.2.1.1>(tag.72b06df-6=d3-4b89-92el-de847500b57a-3073a257
. . . truncated. ..
Support ads Geolocation
Require; precondi tion
p-Asserted- Identity : sip:2 00181O.2 .1.1
. . . truncated. . ,
C=1S IP4 10.2.250.101
t-o o
m.audio 18576 RTP/AVP 9 101
iii= tcp message from 10.2.1.1 on port
a.curnqos e2e none
adesiqos mandatory e2 e
sendracv
a-conf:qos e2e recv
a-rtpmap:9 G722/80O0
a=.ptinw:20
The remote cluster has responded to the INVITE with 100 Trying and 183 Session Progress,
where the acceptance of SIP Preconditions has been declared (the same attributes are repeated).
The terminating SIP UA (the remote Cisco Unified Communications Manager) remrns the
RSVP agent address (10.2.250.101) in the body ofthe 183 Session Progress message.
//SIP/SIPTrp/wait SdlSPrsignal: - - , TCP
5060 index 1
sip:20C1810.2.1.1:5060:transport,tcp SIP/2 0
Via. SIP/2.0/TCP 10.1. 1.1= 5060,branch=iShl3bK14Sd7eBd73
'"" '**-P= 901200 11.10. 1.1 .la,tag.ae40d64a-5deS-4f 28-b79 -2614e-SGi
To: *sip:200 110.2.1.1>,taa.72fb0 6df-6Cd3-4b89-92el.aea47500bS7d-
. . =. truncated. . .
//aiP/SIPTcp/wait SdlKeadRsp: ... ..., . tcp meBBage from 10 2
-.DSD iDds, 1 with 350 bytes:
SIP/2.0 .
Via: SIP/2.0/TCP 10.1.1.1:5060,branCh=r9hG4bK145d7eSd73
307381<fllP:801;:OClelD':l'I'1>itagii'e,Od64d"5de5"4f28"b79"2614e6ea
To: -rmp,3001810. 2. l.l>i tag=72b0 6d.6cd3-4b89-92el-de847500b57d-
Date: Thu, 20 May 2010 15:05:13 GMT
Call-ID: 27d7c300-bf514fc7-3-101010aei0.1 1 1
CSeq: 102 PFACK
age to 10. 2.1.1
L9d44
30731
1.1c
9d44-
30736
257
n port
This tigure shou sthe call-originating Cisco Unified Communications Manager sending thc
PRACK message to the remote cluster.
The remote cluster responds \sith 200 OK.
148 Troubleshooting Cisco Unified Communications (TVOICE) v8 0
m*
Tracing Successful Call Setup
Preconditions (Cont.)
//SIP/SIPTcp/wait_SdlSPISignal: '.Uraa-ug S.1F TCP
5060 index 1
'i-'.i-i- sip:2001810.2.1.1:5060itransport-t=:p SIP/2.0
Via- SIP/2 0/TCP 10.1.1.1:5060jbranch.i9hG4bK157df77b23
From: <3ip: 8012001810.1.1.1>> teg-<40d64d-5de5-4f2i3-b79-2614e6ea9d44-
To"!Bipi2001910.2.1.1>Jtag-72b06df-6c:d3-4bB9-92el-de847S00b57d-30738257
...truncated...
CSeq, 103 UPDATE
Contact: <=Bip: 8012001910.1.1.1:5060, transport-tcp>
Require: precondition
Session-ExpirnB: laOOjrefresber-uan
...truncated...
c.IN IP4 10.1,250.101
. t-0 0
m.audio 16760 RTP/AVP 9 101
a-curr:qos e2e send
a=dea:q h mandatory e2e aendrecv
.rtpmap:9 G722/B000
a.ptime 20
usage to 10.2.1.1 on port
The reservation results are exchanged by using UPDATE (shown here) and OK messages (next
figure).
This update shows the results from thc perspective of the call-originating Cisco Unitied
6-150
//SIP/SIPTcp/wait SdlReadHep: , ;. TCP
5060 iodei 1 with 1025 bytes,
SIP/2.0
Via: SIP/2 0/TCP 10.1.1.1:5060;branch=*9hG4bK157df77b23
tram: *sip =*0l200ieio.l.l.l:>,tag=ae40d64d-5de5-4m-b7f9-2614eea9d44-
3 0 "?3 S 10 5
To: <sip:2001*10.2.1 .1=,tag=7 2f bO 6df -6cd3 -4b89 -92el -deO-i 7500 b57d-3073B257
Supported: Geolocation
Call-info, .umix-sisco-rcotecoipLrpose^.c, sco-preconditions
Session-Expires: 1800;reresher.su as
C=IN IP* 10.2.250 .101
t=o o
=u=audlo 18576 RTP/AVP 9 101
: qos sndi
a-rtpmap:9 G722/B0OO
a=ptlme:20
//SIP/SIPTcp/wait SdlReadPsp: .. ., TCP
5060 index 1 with 6S8 byteg:
SIP/2.0
i: SIP/2.0/TCP 10.1.1.1:5060,branch=z9hG4bK134b93593f
>m: <sip,eoi20Ol>l0.l.i,i>,tag=:ae4Odi=ld-5de5-4f28-b7f9-2614t
saage fi 10.2.1. 1
age 1
The 200 OK message completes the SIP Preconditions exchange, and the rest ofthe intercluster
call setup continues as with any other call. The call setup continues with 180 Rineing. which is
received trom the remote cluster asshown inthe figure.
During the preconditions exchange, the media is negotiated by the RSVP component of the
Cisco Unitied Communications Manager, The media componenr initiates the renegotiation of
the media after the call is connected to ensure proper capabilities on each call leg.
Tracing Unsuccessful Call Setup with SIP Preconditions
This section discusses trace for an unsuccessful intercluster call setup that uses SIP
Preconditions on one side ofthe SIP trunk only. The trace output is presented from the
perspective ofthe left Cisco Unified Communications Manager.
Tracing Unsuccessful Cal
SIP Preconditions
Managei Cluster 1
//SIP/SIPTcp/waitSdlSPISignel:
5060 index 3
-,.-= =,ip:2001910.2.1.1:5060 SIP/2.0
Via- SIP/2 0/TCP 10.1.1.1 =5060,brnch.=r:9hG4bK26227dd72d
FroB: elp: 8012001910. 1.1. l,,t.g.ae40d64d-5d.5-428-b7f9-2614e69d44-
30738113
Tc: <Bip:2001810.2,l.l>
...truncatd...
Call-Info: <eip: 10 .1.1.1!5060>iMthod--HOTIFY,Event-tlephono-
event:Duration.500-
Ciuco-Guid; 3388778624-32097 B1663-0000000005-001684301B
JITT-
TCP
Cisco Unitied Communications
Manager Clusler 2
.SIP Preconditions
Not Configured
sage to 10.2.1.1 on port
The figure shows ascenario in which the left Cisco Unified Communications Manager cluster
has the SIP Preconditions required (configured) but the right Cisco Unified Communications
Manager cluster has the SIP Preconditions completely disabled (including the fallback option).
The call-originating Cisco Unified Communications Manager sends the INVITE with the SIP
Preconditions requirement tothe remote, right cluster.
6-152
//SIP/SIPTcp/wait SdlReodRsp; .- -, .,- TCp
5060 index 3 with 329 bytes,
sip/2.0 . :_ . -
Via: StP/2.0/TCP 10.1.1,1:5060,br0nch,Z9hG4bK26227dd72d
Proa: <ip:80iaODl10.1.1.l>,t.9-B,404.1-51J5-4f2B-b7f9-aS14,SM9d*4-
30739113
To: <sip:200110.2.1.1>
Date: Thu, 20 May 2010 16:21:50 GMT
Call-IB: c9feaflBO-bf51619e-5-101010a=310.1.1 1
CEeq: 101 INVITE
Allow-Events: presence
Content-Length: 0
...truncated...
//SIP/SIPTcp/wait SdlReadBep: ,.., ..TCP meSHage Erom 10 , 1 1 Qn
5060 index 3with 356 bytes: 10.2.1.1 on
SIP/2.0 5=33 Precondition failed
Via: SIP/2.0/TCP 10.1.1 l:5060,branch.z9hG4bK26227dd72d
Froa: <fllp:8012OC13I0. l.I.l>, tag,ae4 0d64d-5de5 -428-b7 f9-2614e6ea9d44 -
3 07 3 B113
To: <sip:2001910.2.1.1>,tag=603303070
Date: Thu, 20 May 2010 16:21,50 GMT
Call-ID: c9ca88O-b51619e-5-101010a91O.l,l 1
CSeq: 101 INVITE
A How- Events : nresensR
Beage
10.2.1.1 on port
The remote cluster responds with a100 Trying and then immediately with a580 Precond.tion
Failed message. At this stage, the calling user hears areorder tone (or annunciator).
Tracing Call Setup with Fallback to Local RSVP
This section shows atrace for acall setup that falls back tothe local RSVP.
Tracing Caii Setup with Fallback lo the
Manager Cluster
RSVP over SIP = E2E
Send PRACK Enabled
Fallback to Local RSVP Enabled
SIP Preconditions
Nol Configured
//SIP/SIPTcp/wait_SdlSPISitjnal: Oytjoii
port 5060 index 6
-'.""- sip,300110.1.250.102:5060 SIP/2.0
Via' SIP/S 0/TCP 10.1.1.1:5O60ibrsnch.z9taG4b*C302e56045b
rroo: <Hip:B012001*10.1.1.1>.ta9.aa40d64d-5d85-4i28-b7f9-2614e6ea9d44-
30738153
To: flip:3001910.1.250.102
...truncated...
...truncated...
a.curr:qoB e2e none
a.desiqos mandatory e2e sendrecv
a-rtpmap:9 G722/8000 ^
TCP massage to 10.1.250.102 on
This figure shows ascenario in which the left Cisco Unified Communications Manager cluster
has the SIP Preconditions required (configured), but the right Cisco Unified Communications
Manager Express is configured only for local RSVP (no SIP Preconditions enabled). In
addition, the Cisco Unified Communications Manager cluster enables fallback to the local
RSVP.
The figure shows twoRSVP agents:
HQ1 associates with thecalling IPphone.
BR1associates with the SIP trunk (it is needed for the proper operation ofthe fallback to
the local RSVP). In this particular example, BR1 has two roles, itserves as the RSVP
agent, and it acts as Cisco Unified Communications Manager Express. However, it could
be that these roles are separated and implemented on two physically separate platforms.
The Cisco Unified Communications Manager sends thc INVITE with the SIP Preconditions
requirement to the remote Cisco Unified Communications Manager Express.
//SIP/SIPTcp/wait SdlEeadRsp: ;>
port 5060 index 6 with 424 bytes:
SIP/2.0 =120 Bad Extension
Via: SIP/2. 0/TCP 10.1.1. 1:5060, bra nch. ! 9hG4 bK30 2eL56 045b
Fro*: =sipi8 012O0 1ilO.l.l.l>jbag=.ae4OdS4d-5de5-428-b7f9-2S14e(ea9d44-
To: <Bip:300 1=*10 . 1.2 50.102> =tag=ieA96C-13B
. . . truncated". . .
Allov-Events: telephone-event
Unsupported : prcoondi tion
Server: Cisco -SIP Gateway/ IOS- 12.x
io'r&T^ SdlSP1SlS -- ' "' --- to 10.L250.102
<* - Bip:300110. 1.250. 102:5060 SIP/2,0
Via: SIP/2.0/TCP 10.1. 1.1: 506 0; branchy 9hG4blGl 5672a037
FroD: <sips =501200 Iei0.1.l.l>,tag=ae40d64d-5de5.42g-b79-214eea9d44-
3073B153
To: sip :3001IS10. 1.250. 102
. SIP precondition r.ct requested h6re, local RSVP EsllbacK in action
Call-Info: csip iio. 1.1.1:5060?,method=: 'NOTIFY; Event =telephone-
event, Duration=500"
CiBCO-Guid: 0289132704-3209789942-0000000013-0016843018
Session-Expires = 1BO0
TCP meBflage from 10.1.250.102
The Cisco Unified Communications Manager Express responds with 420 Bad Extension
indicating that the SIP Preconditions is not supported there.
IheCisco Unified Communications Manager, because fallback to the local RSVP is enabled
sends are-INVITE that does not include any SIP Preconditions requirements to the Cisco
Unified Commumcations Manager Express. From this stage on, the call setup proceeds as
2010Ctsco Systems. Inc
Summary
Summary
Cisco Unified Communications Manager allows RSVP CAC to
beenabled selectively between pairs oflocations orwithin a
location. The general issues ofRSVP CAC areblocking calls
gratuitously or not making bandwidth reservations.
RSVP agents must beregistered for RSVP CAC towork. If
registration issues occur, use Cisco IOS debug seeporCisco
Unified Communications Manager trace to find a cause.
. When a call is blocked by RSVP CAC, consider these most
common causes: bandwidth isexhausted at locations or the IP
WAN, IP phones arenot associated with RSVP agents or agents
are misconfigured, or network connectivity toan agentor
between agents has been lost.
- SIPPreconditions allows for intercluster RSVP CAC. Theissues
possiblyexperienced are similar to those of intracluster RSVP
CAC.
In this lesson, you have learned to explain the common issues that are related to RSVP agents
and toidentify the most likely causes of these issues.
References
Cisco IOS Quality of Service Solutions Configuration Guide, Release 15.0, Configuring
RSVP at http: "\\ vvw.eisco.com/en/US/docs/ios/qos/configuration/gLtide/
coiitig_ri\p_psl0591_TSD_Products_Configuration_Ciuide_Chapter.html
Cisco IOS andNX-OS Software, Configuring SIPRSVP Features at
http: wwu .cisco com en. US/docs/ios/voicc/sip/configuration/guide/sip^cg-rsvp.html
Cisco Unified Communications Manager Administration Guide, Release 8.0(2), Location
Configuration at . .
https: www.cisco.eom'en US.'docs.'voice_ip_conim'eucm/admin/8_0_2/ccmc1g'
b021ocat.html
&2010 Cisco Systems, Inc
Lesson 6
Troubleshooting Voice Quality
Issues
Overview
Isolating the source of voice quality issues is one of the most difficult problems that you can
face when troubleshooting aCisco Unified Communications system. Voice quality can be
subjective and data networks are not sensitive to the same impairments or limitations to which
voice over data networks are sensitive. This lesson concentrates on maintaining the quality of
the voice traffic that crosses the data network and focuses on how to troubleshoot vo.ce quality
issues.
Objectives
Upon completing this lesson, you will be able to explain common voice quality issues and
identify the most likely causes of these issues. This ability includes being able to meet these
objectives:
Describe the voice quality issues from which aCisco Unified Communications system can
potentially suffer and explain the negative effects of lack of bandwidth, long end-to-end
delay, long jitter, and packet loss onVoIP communications
Describe the QoS traffic requirements for voice traffic and define voice QoS policy and the
major stepsto createit
Describe how to identify voice quality issues and the tools to use when troubleshooting
VoIP quality problems
Describe how Layer 2switching issues can cause quality problcms and diagnose and
troubleshoot these problems
Describe what gateway issues can result in VoIP quality problems and diagnose and
resolve these issues
Resolve VoIP quality issues when given symptoms and information that is gathered about
the problem
Voice Quality Issues in Cisco Unified
Communications Systems
6-158
This topic describes the voice quality issues from which aCisco Unified Communications
system can potentially suffer and explains the negative effects of lack of bandwidth long end-
to-end delay, long jitter, and packet loss on VoIP communications.
The following major problems may occur:
Converging voice, video, and data onthe same infrastructure
might cause several issues
*Voice and video are sensitive to latency
Voice and video are sensitive to jitter, which is variation in delay
3Voice andvideo are sensitive to drops
*Guaranteed bandwidth required for voice and video
The converged network infrastructure requires QoS
Here arc thc five main problcms that face converged enterprise networks:
Bandwidthcapacity
Delay issues
Variabledelay
Variation indelay (also called jitter)
Packet loss
Large graphic files, multimedia, and increasing demand for voice and video will all potentially
compete lor bandwidth over data networks and can cause capacity problems.
Delay is the time that it takes tor apacket to reach thc receiving endpoint after being
transmitted from thc sending endpoint. This time is called the end-to-end delay and it consists
ottwo components: fixed network delay and variable network delay. Jitter is the delta or
difference, in the total end-to-end delay values oftwo voice packets in the voice flow.'
Two types of fixed delay arc serialization and propagation delays. Serialization is the process of
placing bits on thc circuit. The higher the circuit speed, the less time that it takes to place the
bits on the circuit. Therefore, the higher the speed of thc link, the less serialization delay that is
incurred. Apropagation delay is the time that it takes frames to transit the physical media.
Aprocessing delay is atype of variable delay. Aprocessing delay is thc time that is required by
anetwork device to look up the route, change the header, and complete other switching tasks.
In some cases, the packet must also be manipulated. For example, the encapsulation type or the
hop count must be changed. Each of these steps can contribute to thc processing delay.
Another tvpe of variable delay is aqueuing delay. This is the time that apacket spends in a
queue or buffer before being processed. Packets can be queued by routers or swttches on an
ingress interface, an egress interface, or both. Aqueuing delay can be significant if arate
change occurs or ifmany interfaces are aggregated into asingle uplink.
Congestion either in the WAN or on aLAN interface is the typical cause of packet loss, which
results in speech dropouts or astutter effect ifthe playout side tries to accommodate by
repeating previous packets. WAN loss is usually caused by tail drops that occur because of
congestion on the WAN link. LAN packet loss can be caused by congestion on an tnterface in
whichthe buffersfill up because of a spike in traffic.
Lack of Bandwidth
6-160
This tigure explains how alack of bandwidth can adversely affect voice quality in anetwork.
.ack of tlarui width
- If QoS is deployed properly a limited bandwidth may not be
noticeable for a voice or video calls
>Otherwise, issues can be experienced:
- Call setup delay, not hearing dial tone for several seconds
Choppy voice, latency andjitter
Bandwidth
Bandwidth
,= min (10 Mb/s, 256 kb/s, 512 kb/s, 100 Mb/s) =256 kb/s
,,- bandwidth ,,/ flows
n
Vou must consider bandwidth on the entire communication path between thc source and thc
destination. This example illustrates an empty network with four hops between aserver and a
client. Each hop ,s using different media with adifferent bandwidth. The maximum available
bandwidth is equal to the bandwidth of the slowest link. So, although the workstation has IO
Mb sof bandwidth, packets that flow between these devices must cross the slow-speed WAN
link at 256 kb s. '
It is rare that only asingle communication flow is present on acomputer network at agiven
time. In reality, multiple communication flows are competing for thc same bandwidth The
calculation ol the available bandwidth is much more complex when multiple flows are
traversing the network. The calculation ofthe available bandwidth in the figure is arough
approximation.
The usual packet loss occurs when routers run out of buffer space for aparticular interface
(output queue).
Routers might also drop packets for other (less common) reasons:
Input queue drop: The main CPU is congested and cannot process packets (the input
queue is full).
Ignore: Thc router ran out of buffer space.
Overrun: The CPU is congested and cannot assign afree buffer to thc new packet,
Frame errors (cyclic redundancy check |CRC], runt, giant): The hardware detected an
error in a frame.
WAN links: frame slips or discards by thecarrier.
Troubleshooting Cisco Unified Communications (TVOICE) v8 D
Packet loss is usuallv aresult of congestion on an interface. Most applications that use TCP
experience aslowdown because of the TCP adjusting to the network resources (dropped TCP
segments cause TCP sessions to reduce their window sizes). Other applications do not use TC P
and cannot managedrops (fragileflows).
Proper network design when using drop-sensitive applications, such as VoIP and video, dictates
thc use of packet loss prevention mechanisms. Ilere are various approaches that you can take to
prevent drops of sensitive applications:
Increase link capacity to ease or prevent congestion. Guarantee enough bandwidth and
increase buffer space to accommodate bursts offragile applications. The recommended
queuing mechanism islow latency queuing (LLQ).
Prevent congestion by dropping packets from other nondrop-sensitive or less-critical
applications before congestion occurs. You can use weighted random early detection
(WRED) tostart dropping these packets before congestion occurs.
Confirm that proper trunk sizing and quality of service (QoS) settings are configured.
Confirm that proper switch QoS settings are configured.
On frame Relay circuits, use traffic shaping to minimize the loss ofvoice and video
packets inthe Frame Relay network.
Implement traffic policing to prevent aclass of traffic from monopolizing an interface.
2010 Cisco Syslems, Inc. Voice Quality and Media Resources Issues 6-161
End-to-End Delay
6-162
This figure explains how end-to-end delay can adversely affect QoS in anetwork.
Delay below 150 ms one way is normally acceptable
Longer delay can cause the following issues:
- Participants in the call will step on each other
Echo is very annoying
=P1 +Q1 +P2 +Q2 +P3 +Q3 +P4 =Xms~~l
You must consider delay over the entire communication path, end to end. Therefore the total
end-to-end delay is the sum total of all delay that is experienced over acommunication path
between asender and receiver. The figure illustrates the impact that anetwork has on the end-
to-end delay, fcaeh hop in the network adds to thc overall delay because ofthese factors:
Propagation delay is caused by thc speed of light that is traveling in the media (for
example, the speed oflight that is traveling in fiber optics or copper media).
Serialization delay is the time that it takes to clock all of the bits in apacket onto the wire.
This fixed value is a function of the link bandwidth.
Processing and queuing delays within arouter are caused by awide variety of
conditions
People generally ignore propagation delay, but itcan be significant (about 40 ms coast to coast
over fiber optics ping is one way to measure the round-trip lime (RTT) ofIP packets in a
network).
Adelay below 150 ms is generally acceptable for networks that transport real-time traffic, and
it should be the target ofanetwork design phase.
Jitter
This figure describes how jitter can adversely affect voice and video quality.
Jitter
Traffic Flow Direction
No Jitter Present
Jitter Present
Jitter is defined as a variation inthe delay of received packets
Occurs due to network congestion, improperqueuing,
configuration errors orsimply multiplexing various traffic types
Jiner is defined as avariation in thc delay ofreceived packets. At the sending side, packets are
sent in acontinuous stream with the packets spaced evenly apart. Because ofnetwork
congestion, improper queuing, or configuration errors, this steady stream can become lumpy, or
the delay between each packet can vary instead ofremaining constant.
The easiest and best place to start looking for jitter is at the router interfaces, because you have
direct control over this portion ofthe circuit. Another alternative to see the jitter is to use
statistics ofthe Cisco Unified IP phone. Both methods will be demonstrated later in this lesson.
You should use LLQ. Queuing is generally not acause ofjitter, because the variations in delay
that are created by it are relatively small. However, ifVoIP packets are not queued properly
and there is data on thc samecircuit,jitter canresult.
On slow links (less than 768 kb/s), you should also deploy link fragmentation and interleaving
mechanisms, which help tomaintain jitter within acceptable limits.
The digital signal processors (DSPs) inside the router can make up for some jitter, but they can
be overcome by excessive jiner, which results in poor voice quality. Jitter occurs when apacket
becomes queued or delayed somewhere in the circuit, where there was no delay or queuing for
other packets. This behavior causes avariation in latency, Both router misconfiguration and
misconfiguration by either the carrier or provider of the permanent virtual circuit (PVC) can
cause jitter,
Echo
Echo is one of the more difficult voice quality problems to troubleshoot, and it is one of the
most commonly encountered problems.
- >& jj m$ L*.
Echo isa function ofthe echo delay andtheloudness ofthe echo.
Packet segmentation will introduce delay which is typically around
30ms ineach direction. This can result inecho that was
previously indistinguishable to the human earnow being
perceivable.
Echo Path Delay (ms)
Echo is present in almost any Cisco Unified Communications system. IP telephony in
particular, makes echo problems more apparent than they might be in anonpaeket network.
Before identifying echo in any Cisco Unified Communications system, you first must know the
possible causes and the impact that echo has on an IP telephony network Gathering key
information regarding the type and kind ofecho is critical to isolate, reduce, or eliminate echo.
Echo is the sound of your own voice reverberating in the telephone receiver while you are
talking When tuned properly, echo is not aproblem in thc conversation; however ifthe echo
interval exceeds approximately 25 ms. it can be distracting to the speaker. In the traditional
telephony network, echo isgenerally caused by an impedance mismatch when thc four-wire
network isconverted to thc two-wire local loop.
IP phone-to-IP phone echo is rare; however, when speaker-phones are used close to one another
echo is commonly heard. This echo occurs because ofacoustic feedback through the phone
speaker and is heard when the speakerphone volume is set at 75 percent or above.
The most common echo is talker echo, which IP phone users hear. You can eliminate most
issues relative to echo at the gateway. 11.323 gateways have Cisco IOS Software commands
that can assist in reducing or eliminating echo. Media Gateway Control Protocol (M(iCP)
gateways have options within Cisco IInified Communications Manager to assist in reducin, or
eliminating echo. Cisco Catalyst 6500 Series Switches and Cisco Catalyst 660K Voice
Gateway-1 Ior E1 ports have no options for port tuning on thc switch; thc tuning is all done
from the Cisco Unified Communications Manager web pages.
For echo to be aproblem, all ofthe following conditions must exist:
An analog leakage path between analog transmit (Tx) and receive (Rx) paths
Sufficient delay in echo return for echo to be perceived as annoying
Sufficient echo amplitude to be perceived as annoying
6-164 Troubleshooting Cisco Unified Communicalions (TVOICEj v8.'
QoS Requirements and QoS Policy
This topic describes the QoS traffic requirements for voice traffic and defines voice QoS policy
andthemajor steps to create it.
Voice QoS Traffic Requirements
Latency < 150 ms*
Jitter < 30 ms*
Loss<1%*
17-106 kb/s guaranteed
priority bandwidth
per call
150 b/s (+Layer 2
overhead) guaranteed
bandwidth for voice-
control traffic per call
I "One-way requirements
1Smooth
Benign
Drop-Sensitive
Delay-Sensitive
UDP Priority
Voice traffic has extremely stringent QoS requirements. Voice traffic usually generates a
smooth demand on bandwidth and has a minimal impact on other traffic as long as you manage
it.
While voice packets are typically small (60 to 120 bytes [B]). they cannot tolerate delay or
drops The result of delays and drops is poor and often unacceptable voice quality. Because
drops cannot be tolerated, User Datagram Protocol (UDP) is used to package voice packets,
because thcTCPretransmit capabilities have no value.
Voice packets can tolerate no more than a150-ms delay (one-way requirement) and no more
than a1percent packet loss (it might be even less if aloss-sensitive codec is used like G.723,
for instance).
Atypical voice call requires from 17 to 106 kb/s (depends on the codec used) of guaranteed
priority bandwidth plus an additional 150 b/s per call (if SCCP is used) for voice control traffic.
To determine the bandwidth requirement per each VoIP call, several factors must be
considered. These factors include the codec that isused, the sampling rate, and the overhead.
Use this formula tocalculate bandwidth requirements persingle call:
Bandwidth per call =(voice payload +L3 Oil +L2 OH) *packets per second *8bits/byte
This formula illustrates how to calculate the required bandwidth per single G.711 call when the
call is transported over Ethernet with IEEE 801 .Q VLAN framing at 50 p/s (20-ms
packetization period):
160 bytes isthe G.711 payload that iscollected over 20ms.
40 bytes is the Layer 3+ overhead (IP, UDP, Real-Time Transport Protocol [RTP] headers).
32 bytes isthe Layer 2overhead for 801 .Q Ethernet.
50 p s is the sampling rate.
8bits 'byte is used to convert byte values into bits for the final bits per second result The
result is 93 kb's.
Use the same approach when you calculate thc required bandwidth for other codecs and Lava
2 technologies.
Multiplying these bandwidth requirements times the maximum number ofcalls that are
expected during the busiest time indicates the overall bandwidth that is required for voice
traffic,
6-166 Troubleshooting Cisco Unified Communicalions (TVOICE] v8 0
2010Cisco Systems Inc
Typical QoS Policy Definition
After the majority of network traffic has been identified and measured, use the business
requirements to define traffic classes.
Typical QoS Policy Definition
,'Oico Bearer
IP Routing
Network Mgmt.
CallSignaling |
Interactive Video |
Streaming Video |
Mission-Crit. Data ]
1Transactional Data|
Bulk Data
Scavenger
| EF
| CS6
] CS2
| CS3
| AF41
| CS4
|AF31
| AF21
| AF11
h
|CS1
"
' No bandwidth guarantee
Because of its stringent QoS requirements, voice traffic will usually exist in aclass by itself.
Many enterprises have several hundreds of data applications that run over their networks.
Adequate bandwidth must be provisioned for this class as awhole to manage the sheer volume
of applications that default to it. It is recommended that at least 25 percent of the bandwidth of
a WAN link bandwidth be reserved for thedefault best-effort class.
It is recommended toallocate about 30 to33 percent ofthe bandwidth ofaWAN link
bandwidth to priority queuing. This is especially important on low- and medium-speed links
(less than Tl or El). This limit for the sum of alt low-latency queues is simply abest-practice
design recommendation; it isnot a mandate.
Thc rest ofthe bandwidth ofalink can be given to rate-based queues and their associated traffic
types.
QoS Policy Implementation Options
The two most comprehensive QoS architectures arc Integrated Services (RFC 1633 f19941) ant
DifferentiatedServices (RFC"' 2474 [19981).
itBeB&*,* =A.<M,=*gL=B.^L.
tl Kl *!,,
Classification Each class-oriented QoSmechanism has to
support some type of classification.
Marking: Used to mark packets based onclassification,
metering, or both.
;- Congestion management: Each interface must have a
queuing mechanism to prioritize transmission of packets.
- Congestion avoidance: Used todrop packets early toavoid
congestion later in the network.
- Policing and shaping: Usedto enforcea rate limit based on
the metering (excess traffic is eitherdropped, marked, or
delayed].
*Link efficiency: Used to improve bandwidth efficiency through
compression, link fragmentation, and interleaving.
The differentiated services (DiffServ) model is aclassify-and-mark QoS model that does not
provide end-to-end pcr-fiow QoS. DiffServ is classified as a"coarse grained" QoS model It is
more scalable than IntServ, DiffServ was explicitly developed to be an IP QoS model that can
act as amore scalable alternative to IntServ. While DiffServ provides solutions to scalability
limitations with IntServ, DiffServ has aset of limitations as well. DiffServ is the predominate
IPQoS model that isdeployed today.
The DiffServ architecture is based on asimple model in which traffic that is entering anetwork
is classified and possibly conditioned at the boundaries ofthe network. The traffic class is then
identified with adifferentiated service code point (DSCP) or bit marking in the IP header.
DSCP -.allies are used to mark packets to select apcr-hop behavior (PUB). Within thc core of
the network, packets arc forwarded according to the PIII3 that is associated with the DSCP. The
PUB is defined a* an externally observable forwarding behavior that is applied at aDiffServ -
compliant node toa collection of packets with the same DSCP value.
One of the primary principles of DiffServ is that you should mark packets as close to the edge
ot the network as possible. It is often adifficult and time-consuming task to determine the
traffic class for adata packet. You should classify the data as few times as possible. By
marking the traffic at the network edge, core network devices and other devices along the
forwarding path will be able to quickly determine thc proper class of service (CoS) to apply to a
given traffic flow.
DiffServ is not designed to eliminate delay or delay variation; it is designed to make delay and
delay variation predictable.
2010 Cisco Systems,
The main categories of DiffServ tools that are used to implement QoS in anetwork arc as
follows:
Classification and marking: The identifying and splitting oftraffic into different classes
and the marking of traffic according to behavior and business policies. The typical QoS
tool in this category is the Cisco Modular QoS CLI (MQC), especially its class-map
configuration. Class-based marking is used for marking the traffic.
Congestion management: The prioritization, protection, and isolation of traffic that is
based on markings. LLQ is the recommended mechanism for congestion management in
converged networks.
Congestion avoidance: Discards specific packets that are based on markings to avoid
network congestion. The intelligent dropping mechanism that belongs to this category is
WRED.
Policing and shaping: Traffic conditioning mechanisms that police traffic by dropping
misbehaving traffic to maintain network integrity. These mechanisms also shape traffic to
control bursts by queuing traffic. Either class-based policing or class-based traffic shaping
can be used to condition the traffic.
Link efficiency: One type oflink efficiency technology is packet header compression,
which improves thc bandwidth efficiency ofalink. Another technology is link
fragmentation and interleaving (LFI), which can decrease the jitter of voice transmission by
reducing voice packet delay. Typical LFI mechanisms that can be used on Cisco IOS
platforms are Multilink PPP (MLP) with interleaving and the Frame Relay Fragmentation
Implementation Agreement (FRF.12).
Identifying and Isolating Voice Quality Problems
This topic describes how to identify voice quality issues and the tools to use when
troubleshooting VoIP quality problems.
Thefollowing helps in the process of identification:
Understand the codecand bandwidth neededfor the codec
Understand the network topology and WAN technologies
"Use queuing techniques that allow voiceand videoto be
identified and prioritized
LFI and cRTP should be used on slow WAN links when voice
and video traffic shares the WAN links withdata
*Look for thefollowing when troubleshoofing:
Interface drops
Buffer drops
Policy-map drops
Interface congestion
Alter VoIP calls are properly established, the next step is to verify that thc voice quality is
good. Consider the following guidelines as you attempt to achieve good voice quality:'
Know how much bandwidth aVoIP call consumes with each coder-decoder (codec),
including these headers: Layer 2, IP, UDP, and compressed RTP (cRTP).
Understand the characteristics of the IP network over which the calls travel. For example,
the bandwidth ofaFrame Relay network that is operating at the committed infonnation rate
(CTR) is \ery different from the bandwidth ofaFrame Relay network that is operating
above CIR. lor Frame Relay networks that operate above CIR, packets could be dropped or
queued inthe Frame Relay cloud. Ensure that delay and jitter are controlled and eliminated
as much as possible. One-way transmit delay should not exceed 150 ms (per G. 114
recommendation).
Ubc aqueuing technique that allows VoIP traffic to be identified and prioritized.
When transmitting VoIP over low-speed links, consider using Layer 2packet fragmentation
techniques such as MLP with LFI on point-to-point links or FRF, 12 on Frame Relay links.
Fragmentation of larger data packets allows less jitter and delay when transmitting VoIP traffic,
because the VoIP packets can be interleaved ontothe link.
With VoIP, uhen you are troubleshooting QoS issues, in particular, look for dropped packets
and network bottlenecks that can cause delay and jitter. Specifically, look for the following:
Interface drops
Buffer drops
Policy map drops
interface congestion
Link congestion
You should examine each interface in the path ofthe VoIP and eliminate drops and congestion.
Also, reduce round-trip delay as much as possible. Pings between the VoIP endpoints give an
indication ofthe round-trip delay ofalink. The round-trip delay should not exceed 300 ms,
whenever possible. Ifthc delay must exceed this value, try to ensure that this delay is constant,
sothat you do not introduce jitter or variable delay.
2010 Cisco Systems. Inc Voice Quality and Media Resoles Issues 6-171
Ask an end user the following questions:
* Has it ever worked?
- Did any error messagesappearor were any error messaqes
heard? y
' What time did the problem occur?
a Are problems limited to the site or subnet?
Have there beenrecentchanges that you are aware of?
- Have you tried the same destination from another phone, and
if so, what were the results?
' Do calls to other destinations work?
What device were you trying to use?
Has this happened before?
Voice quality problems can be difficult to resolve. The problems can be intermittent, and
quality evaluation can he subjective.
In aCisco Unified Communications system network, one ofthe following types ofusers
typically discovers and reports problems:
External users trying to reach employees within your company
Internal users calling employees in other company locations or public switched telephone
network (PSTN) destinations and performing basic actions such as call transfers and dialing
into conferences
Collect sufficient information from these users lo allow you to isolate the problem. Detailed
accurate information will make this task easier. Common questions toask the end user when
reporting a problem shouldinclude the following:
Has it ever worked'.'
Did any error messages appear or did you hear any error messages?
What time did the problemoccur?
Are problems specific to the site or subnet?
Ha\e there been recent changes of which you areaware?
Have you tried ihe same destination from another phone, and, ifso, what were thc results?
Do calls to other destinations work?
Which devicewere you trying to use?
Has this happened before?
Identifying and Isolating Voice Quality
Problems (Cont.)
Isolate potential issues to most likely causes
Networkdiagrams can be useful
Call flowdiagram can be useful
Action plan
Take the mostlikely cause ofthe problem andtake
correctiveaction, beingcareful to change onlyone
variable at a time
Analyze the results
Test the results to see ifthe problemhas been resolved.
If not, undo any changes made, and repeat forthe next
most likely cause of the problem.
- If you have exhausted all potential causes, contact
Cisco TAC.
After additional information has been collected from the end users, IPphones, switches,
routers, and Cisco Unitied Communications Manager servers, begin tonarrow down the source
of the issue to a list of the most probable causes.
To help narrow down the potential issues, dothe following:
Use anetwork diagram to understand the network and to verify connectivity.
Use acall flow diagram tounderstand where VoIP traffic should flow.
Have an understanding ofthc devices that arc involved inthe problem.
The use of phone, switch, gateway, and Cisco Unified Communications Manager tools can help
tofurther define and narrow the list ofpossible issues that could be causing the problem.
Some of thc more commonly used toolsinclude thefollowing:
IP phone tools
Status button to display RTP statistics
Quality Report Tool
Switch tools
show commands
Console messages
Router tools
show commands
debug commands
Console messages
Cisco Unified Communications Manager tools
Cisco Unified Real-Time Monitoring Tool (RTMT)
Trace files
System logs
Quality Report Tool (QRT) reports
Alarms
Call Management Records (CMR)
6-174 Troubleshooting Cisco Unified Communications (TVOICEl v80
Cisco IP Phone RTP Statistics
The figure shows now to display RTP call statistics.
Cisco IP Phone RTP Statistic;
During avoice eail. you can view the call statistics information from the Cisco IP phone. This
can show properties ofthe call like codec, packets sent, packets received, packets lost, delay,
jitter, and so on.
Ifweb access is enabled, you can obtain this infonnation by browsing to the IP address ofthe
Cisco IP phone.
Start troubleshooting voice quality problems with the real-time data that is available during the
call from the Cisco IP phone. To access the call statistic information from the phone, rapidly
press the i button twice (or on some other models, the question mark [?] button). Here is some
of the information that displays:
Sent and received packet counts (because the IP phone opened the RTP stream)
Send and receive stream type (for example, the codec that isbeing used)
Average and maximum jitter (because the IPphone started the stream)
Numberof receivepacketsthat are lost in transit
Number of receive packets that arediscarded after being received
IfCMR is enabled, you can also retrieve the call statistics information after the call has ended
by using the Cisco Unified Communications Manager Call Detail Record (CDR) Analysis and
Reporting (CAR) tool. To retrieve the call statistics information, choose System >Service
Parameters. Choose the Cisco CallManager service for a server and change the setting for Call
Diagnostics Enabled to Enabled Only When CDR Enabled Is True or Enabled Regardless of
CDR-Enabled Flag (the default is False). Repeat these steps for all servers in the cluster. CMR
records will nowbe generated for every call.
Note Enable CDR records ina similar fashion. Depending on the selection forthe Call
Diagnostics option, you can enable CDR to generate CMR records.
VoiceQuality and MediaResources Issues 6-175
Troubleshooting Layer 2 Quality Problems
This topic describes how Layer 2switching issues can cause quality problems and how to
diagnose andtroubleshoot these problems.
3-176
>OU
In LANs, general lack ofbandwidth istypically not an issue.
Buffer congestion is an issue.
Buffer congestion occurs when there isa ratechange or if
manyinterfaces are aggregated to a single uplink.
Make sure that voice is mapped toan expedite queue.
100 Mb/s
100 Mb/s
100 Mb/s
Direction of Traffic Flow
Direction of Most of Traffic
With physical connectivity established, another logical troubleshooting target is Layer 2of the
Open Systems Interconnection (OSI) reference model. This topic examines Layer 2bottlenecks
and what you can do to overcome these network limitations.
In a campus environment, with latency-sensitive applications, buffers, not bandwidth, are the
issue. Buffers can fill instantaneously. When this occurs, the router or switch can drop packets
when the packet attempts to enter the interface buffer. For applications such as voice, this
results indegradation of voice quality.
Another possible source of problems occurs when there are speed mismatches. Where aGigabit
Ethernet segment flows into an Ethernet segment, oversubscription is likely. Also, consider a
situation in which you have a server and multiple clients that are connected toa Fast Ethernet
port onthe same switch. With this configuration, clients could saturate the Fast Ethernet link to
the server ifthey all send traffic simultaneously for an extended period.
2010 Cisco Systems inc
mm
Layer 2Voice Quality Considerations
This section explains what to consider on Cisco Catalyst switch regarding voice quality.
Layer 2 Voice Quality Considerations
To avoid voice quality issues at Layer2 checkthe
following on Catalyst switch:
If egress and ingress queueing for voice ensureslow latency
If voiceis mapped toqueue threshold minimizing drops
If outgoing voice is mapped tocorrect outgoing marker
- If rate limiting is used, make sure its impact onvoice is
minimal
Poor voice quality due to layer 2is less likely than WAN congestion issues, however, the
checks for the LAN switch per recommended settings should also be performed as part of
troubleshooting procedure. To minimize voice quality issues due to incorrect Layer 2
configurations, check the following onCisco Catalyst switch:
Voice traffic must beserviced with priority. Verify, if egress and ingress queueing that
services voice is set up toguarantee lowlatency.
During traffic congestion, Layer 2switch queues start to fill up. When this happens, the
switch can either letthe queues overflow orcan drop packets toprevent the queues from
overflowing. Switches can use WRED to intelligently drop lower-priority traffic, keeping
higher-priority traffic, like voice, in the queue. Verify, ifvoice traffic is mapped to such
queue threshold that is minimizing drops for voice packets.
When voice traffic leaves the switch, other devices down the stream should be able toapply
the same rules tomaximize quality ofvoice. Verify, ifoutgoing voice traffic ismapped to
correct outgoing marker according torecommendations.
Ifthe switch implements port-based rate limiting, verify ifits impact on voice traffic is
minimal, hence the voice traffic gets dropped minimally.
VoiceQuality and Media Resources Issues 6-1T7
Queuing and Scheduling on Cisco Catalyst 3750 Series
Switches
For the demonstrations inthis lesson, the Catalyst 3750 Scries Switch is used,
lor other switches, check out what isappropriate.
txamplo of Queuing a
Ingress Queue Options: 1P1Q3T
Egress Queue Options- 1P3Q3T
4Q3T
The Catalyst 375(1 Series Switch has nvo ingress and four egress queues per port.
Both the ingress and egress queues are serviced by shaped round robin (SRR), which controls
the rate ai which packets arc sent. On the ingress queues, SRR sends packets to the stack ring.
On the egress queues, SRR sends packets tothe egress port.
You can configure SRR on egress queues for sharing or for shaping. However, for ingress
queues, sharing is the default mode, and it is thc only mode supported.
In shaped mode, the egress queues are guaranteed apercentage of thc bandwidth, and they are
rate-limited to that amount.
In shared mode, the queues share the bandwidth among them according to the configured
weights. The bandwidth is guaranteed at this level but not limited to it.
The Catalyst 3750 Series Switch supports two configurable ingress queues, which are serviced
by SRRinsharedmodeonly:
Normal: User traffic that is considered to be normal priority. You can configure three
different weighted tail drop (WTD) thresholds to differentiate among the flows.
Expedite: High-priority user traffic such as DiffServ EF or voice traffic. You can configure
the bandwidth that is required for this traffic asa percentage of the total stack traffic. The
expedite queue has guaranteed bandwidth.
There are two configurable options for the ingress queues:
1P1Q3T: One queue isexpedite (low latency) and one queue isnormal with the three WTD
thresholds. This is the recommended optionfor voice traffic.
6-178 Troubleshooting CiscoUnified Communications (TVOiCE] v80
mm
*m
m 2Q3T; Two queues are normal with the three WTD thresholds. This is the default on Cisco
Catalyst3750SeriesSwitch.
Each port supports four egress queues, one of which (queue 1) can be configured as the egress
expedite queue. If thc expedite queue is enabled, SRR services it until it is empty before
servicingthe other threequeues.
Therearc two configurable optionsfor the egressqueues:
1P3Q3T: One queue is expedite and three queues are normal with the three WTD
thresholds. This is therecommended option forvoice traffic.
4Q3T: Four queues are normal with the three WTD thresholds. This is the default on Cisco
Catalyst3750SeriesSwitch.
The commands to verify ifCisco Catalyst 3750 Series Switch uses the recommended settings,
are explained later.
,2010 Cisco Systems, Inc. Voice Quality and Media Resources Issues 6-179
WTD on Catalyst 3570 Series Switches
Both the ingress and egress queues use an enhanced version of the tail-drop congestic
avoidance mechanism called WTD.
6-11
** &a.t*dt&ttsiX?ii nidO^Ji
hJSkka +t tsW, J*t
Ingress and egressqueues useweighted tail drop, anenhanced
versionof the taildrop congestion avoidance mechanism
Manages Ihe queue lengths and provides a drop precedence for
different traffic types based on thresholds
Usesthe frame's assigned QoSlabel tosubject ittodifferent
thresholds
CoS 6-7
CoS 4-5
60%
1000 Packets'
600 Packets
-l ' P;K*kL-jis
-3 Thresholds
WTD is implemented on queues to manage ihe queue lengths and to provide drop precedences
tor different traffic classifications.
As aframe is enqueued to aparticular queue, WTD uses the assigned QoS label ofthe frame to
subject it to different thresholds. If the threshold is exceeded for that QoS label (the space
available in the destination queue is less than the size of the frame), the switch drops the frame.
The figure shows an example ofWTD that is operating on aqueue whose size is 1000 frames
Three drop percentages are configured: 40 percent (400 frames), 60 percent (600 frames) and
100 percent (1000 frames). These percentages mean that up to 400 frames can be queued at the
40 percent threshold, up to 600 frames at the 60 percent threshold, and up to 1000 frames at the
100 percent threshold.
In this example, CoS values 6and 7have agreater importance than the other CoS values and
they are assigned to the 100 percent drop threshold (queue-full state). CoS values 4and -i
(voice traffic) are assigned to the 60 percent threshold, and CoS values 0to 3are assigned to
the 40 percent threshold.
Suppose that the queue is already filled wiih 600 frames, and a new frame arrives. Itcontains
CoS values 4and 5and is subjected lo the 60 percent threshold. Ifthis frame is added tothe
queue, ihe threshold v. ill be exceeded, so the switch drops it.
Note
Ingress and Egress Congestion Management allows configuring WTD with three
programmable buffer thresholds
*-
m^
Verify Ingress Classification
Classification is the process ofdistinguishing one kind oftraffic from another by examining the
fields in thc packet.
Verify Ingress Classification
Svitchlshow mlB goa map cos-dflcp
-.- -==,-=* nap:
0Dfli 0 12 3 *KH% S 7
dscp: 0 a 16 24 32*yi4B 56
Switchtsho- qos map dacp-mutation
L=,-s -;:-..: mutstion map'
Default E3CP Mutation Mp =
dl : d2 0 1 2 3 4
00 01 02 03 04 OS |H01 08 0*
10 11 12 13 14 15 |j|17 IB IS
20 21 22 23 24 25 BR-" 28 29
30 31 32_J3_34_35j||''7 38 39
|47 48 49
*57 58 59
60 61 62 63
Before voice traffic can proceed tocorrect expedite queue and threshold, the voice traffic has to
be recognized by the switch through classification.
Cisco QoS Solution Reference Network Design (SRND) determines the following CoS and
DSCP values for voice:
Voice Bearer CoS = 5 and DSCP - 46 (EF).
Voice Control CoS = 3and DSCP =24(CS3), while older voice control traffic marker that
was used at earlier stages is 26(AF31).
Classification isenabled only if QoS isglobally enabled onthe switch. By default, QoS is
globally disabled, so noclassification occurs.
During classification, thc switch performs alookup and assigns aQoS label to the packet.
The QoS label isbased on the DSCP orthe CoS value in the packet and decides the queuing
and scheduling actions toperform on the packet. Thc label ismapped according tothe trust
setting and the packet type.
CoS-DSCP Mapping
Either the switch can pass through the existing marking orit can translate the original value to
another value.
The first thing to consider is whether the QoS function isenabled on the switch. By default,
QoS is disabled on the Catalyst 3750 Series Switch, and the switch simply passes through
existing Layer 3markings. Such setup would not be able to minimize the impact ofcongestion
to voice quality.
VoiceQuality and Media Resources Issues 6-181
1he switch associates each frame with an internal DSCP value as the frame enters aport. If the
switch is instructed to trust CoS on aport, then the switch will consult thc CoS DSCP map to
determine the appropriate value. Typically, thc resulting DSCP value will be written to the ToS
byte in the IP header. As the frame exits, the switch consults the CoS DSCP map. This map can
determine the CoS value that is writien into the dotlq or Inter-Switch Link (ISI.) header as well
as the output queuing.
When aCisco phone is connected to aCisco Catalyst switch, it is common to trust the CoS
value (5 for voice) and let the internal DSCP value be calculated based on the CoS DSCP map
To maintain the original DSCP -.alue of 46. the default CoS DSCP map can he adjusted as
follows:
375CIconfig)8nus qos map cos-dscp 0 8 16 24 32 46 48 56
DSCP-DSCP Mapping
When you trust DSCP, you actually translate the external DSCP value toan miemal DSCP
value by using the default DSCP mutation map.
The outside numbers come from the original DSCP value, and the numbers in the table are the
resulting internal DSCP values, which are usually written to thc packet header. As you can see.
the default mutation map makes no change. In thc example, outside DSCP 46 maps to internal
DSCP 46.
You can create custom maps and apply them toports thai connect to domains with different
marking policies.
Troubleshooting Cisco Unified Communications (TVOICE] vB 0 2010 Cisco Systems Inc
Verify Egress Classification
The following figure shows how to check for egress classification.
Verify Egress Classification
SnitchlBhow
;oo 01 01
,02 02 02
!03 03 03
04 04 04
105 06 OS
06 06 OS 06 06 06 07 07 07 07
07 07 07 07
Similar to inbound mapping, outbound mapping at egress of the switch helps to distribute
correct mapping information further down the stream and thus supports the selection ot proper
voice quality mechanism on other devices. The figure shows that the outbound CoS DSCP
mapping is correct; Voice Bearer Traffic has 46 mapped to 5and Control Traffic has 24 (CS3)
mapped to 3. Also, 26 (AF31) maps to 3, as might be required.
Verify Mapping of Traffic to Egress Queues
To mamtain voice quality, ,t ,s very imponant ,o map voice traffic into an egress priority queue
that guarantees low atencv. Thc mapping is performed at the switch input interfaces, and acan
be based on CoS or DSCP markers. The figure shows how to verify the mapping of voice
traffic type (both bearer and signaling) into an appropriate queue on Cisco Catafyst 3750 Series
Switch.
'Wiiv ma\
i.tcn*3h=w mis qos
Dscp outpjtq tbr
a*->iJ$nLr,.
4
d 01 02 oi 02 oi o: oi 63'Bi 02-01 aa-W 02 01 02-01 02-01
03 t ft. 03 04-01 9>.b? 04 oi 03-02 04-01
.*X _ DlÔ, dl-M 4,.^ M.B ot^ 01.01 0401 04_ol
ncsteS '
outputq th
sh^ld
shoid: 3-i 3-1 3-1 3-3 3-; t.3 3_2 3_2
Switch#3hcw t.l3 qos 1
OigabitEthernet1/0/4
Egress Priority Queue
Siiaped queue weights
Shared queue weights
The port bandwidth li
The port is mapped to
gil.'O/* queue
i3 5 62 :
: 100 (Opt
il Bsndwidth=100.0)
Verify the mapping of voice traffic to egress queues by using ihe slum mis qos map command
The command output that is shown highlights voice bearer (DSCP 46) and voice control
(DSCP2-4| mappings only.
The voice traffic (DSCP 46) has been mapped to the queue numbcr I(the only queue that can
be set to pnon.y| and thc WTD threshold 3(notation 01-03) that guarantees for voice to be
dropped last. Similarly, voice s,gnal,ng (DSCP 24) has been mapped to the recommended
queue number 2 andiheWTD threshold 3 (notation 02-03).
You should also use the show mis qos interface interface-id queueing command to verify that
the priority queuing isenabled at egress.
Swi-ichftshow mis qos interface gil/0/4 queue
Gi gabir.Et nerr.eL1 /C/4
Egress Priority Queue : enabled
Shaped queue weights (absolute! . 3 0 0 0
Shared queue weigr.es : 33 5 .2 1
The port bandwid-h limit : 100 (Operational Bandwidih.-lOO 0=
;h.e pert ls mapped !_=.; qse:. i
6-184 Troubleshooting Cisco Unified Communications [TVOICE) v8.0
I2G10 Cisco Systems, Inc
IM
Monitor Packet Drops
Packet drops significantly influence quality of voice. They occur because of congestion or they
can also becaused by traffic policing toa preeonfigured rate.
lonitor Packet Drops
Packets drops due tocongestion ortraffic policing ontheswitch
S-itch#3ho- policy-map interface Fastetherneto/4
FastEthernetO/4
Service-policy input: Policy-CiacoPhone
Clase-map: VoIP RTP (match-all!
68298 packets, 32=17834 bytes
5 minute offered, rata 128C20 bps, drop rate 32456 bps
Match; :r '"~P *' ii'
Clasa-map: VoIP-Control (match-all)
8723 packets. 147382 bytes
5 minute offered rate 32656 bpu, drop rate 16741 bps
Hatch: :_r. >:'! .'=> ';-: *--* :'-i"
Class-map; class-default (match-any)
8972345 packets, 121417C3S bytes
5 minute olfere
j rate 524 061 bps
Hatch: any
0 pa -kets, 0 bytes
5 ml lute rate 0 bps
Look for interface, policy-map, or queue statistics that shows dropped packets.
The figure shows packet drops due to traffic policing, but similar output would be shown in
case of queue congestion.
class-map match-all VoIP-RTP
match ip dscp ef
class-map match all VoIP-Control
match ip dscp cs3 af3l
policy-map Policy-CiscoPhone
class VoIP-RTP
set dscp ef
police 320000 SO0O exceed-action policed-dscp-transmit
class VoIP-Control
set dscp cs3
police 32OG0 6000 exceed-action policed-dscp-transmit
i
interface FastEthernetO/4
switchport mode access
switchport voice vlan 312
srr-queue bandwidth share 10 10 60 20
priority-queue out
mis qos trust device cisco-phone
mis qos trust cos
spannir.g-tree portfast
service-policy input Policy-CiscoPhone
Troubleshooting Voice Quality Issues on a
Gateway
This topic describes gateway issues that can result in VoIP quality problems and how
diagnose and resolve them.
6-K
Things to consider when troubleshooting voice quality
issues on a gateway:
-- Duplexand speed mismatches to the switch
1 Bandwidth issues across slow WAN links
- LFI and cRTP required on slow WANlinks
* LLQshould be used outbound
- Incorrect configuration for theWAN type
An SLA that isdesigned for voice and/or video should be
purchased
" Carrier network issues
When voice and \ideo quality suffers, thc gateway is the likely cause. This topic discusses the
most common sources ofquality issues that involve agateway and suggests how to diagnose
and troubleshoot them.
Typically, in an enterprise network, troubleshooting Layer 3congestion focuses on the WAN
edge. Ihe bandwidth that is available to routers that are sitting at this WAN boundary might be
insufficient to transport packers in atimely fashion. If you arc experiences voice quality
issues, and you are concerned that thc router might be causing these issues"you should check
the following:
Duplex and speed mismatches: Duplex mismatching is acommon cause ofpacket loss.
Mismatching occurs when one side ofthe connection is set up for full duplex and the other
is set up tor half duplex. Look at the link between ihe two endpoints that are experiencing
thc packet loss, and check the speed and duplex settings for each side.
Bandwidth: Ifinsufficient bandwidth exists for arouter to send out traffic on the wire, the
router attempts to buffer packets until sufficient bandwidth docs exist. However, ifthe'
buffer fills, the router discards newly arriving packets. Cisco calls this occurrence tail drop.
LFI: On circuits with speeds that are slower than 768 kb/s, large data packets could cause
excessive latency for voice packets. Choose an appropriate LFI tool (for example, MLP,
FRF. 12. or the Voice over Frame Relay Implementation Agreement [FRF. 11]Annex C) for
the media. Then, configure afragment size that will result in aserialization delay of
approximately I(1 to20 ms. As arule, you can take thc link speed in bits per second and
divide it by 800 to obtain the fragment si/e. In bytes (B), that would yield aserialization
delay of 10 ms. For example, ifthe link speed were 64,000 b/s, the fragment size that
would exit theinterface in 10ms is 80B(64,000 / 800= 80B).
LLQ: LLQ has the ability to distinguish between multiple traffic types (up to 64) and to
reserve an amount ofbandwidth for each ofthose traffic classifications. One or more ofthe
classes can have their traffic sent to apriority queue. Therefore, ifyou are troubleshooting
asituation in which the network isnot giving voice traffic sufficient bandwidth, you can
use LLQ to classify voice traffic (perhaps based on an IP precedence value) and give a
specified amount ofpriority bandwidth to the voice traffic.
Interface configuration: The technology on which the WAN link is based will usually
dictate unique configuration parameters to properly enable it for real-time traffic like voice
and video. For example, when you use Frame Relay to carry voice traffic, you should
configure it to do very little bursting and no excess bursting.
Service level agreement (SLA): The circuit that is used to transport your voice and video
traffic across acarrier cloud should have an SLA that is acceptable for the nature ofreal
time voice and video.
Carrier network: The carrier network can be the source ofvoice and video quality issues.
For example, aFrame Relay network will discard packets when congestion is experienced
in thc carrier network.
Monitor Interface Load and Congestion
Ihis section shows how to monitor load and congestion on an interface.
HQ-l#9hov interface Serial 3/1/0 '
SerialO/1/0 iB up, iine protocol 1B up
Hardware is GT96K Serial
Description: to PSTN
MTU 150C bytes, BW 256 Kbit/sec, t,Y 20000 usee,
reliability 255/255, txlcad 243/255, rxload 178/255
input queue: 0/75/0/0 [. iWmax/drops/flushes) , Total output dropB o
Queueing strategy: Claes-baaed queueing
Output queue: 697/1000/963 (sue/max total/drops)
5 minute input rate 196788 bits/aec, 206 packeta/sec
5 minute output rate 236966 bits/sec, 315 packets/sec
1537319 packets input, 119712709 bytes. 0 no butter
Received 0 troadcaats, 0 runts. 0 giants. 0 throttles
0 input mora, 0 CHC , 0 frame, 0 overrun, 0 ignored, 0 abort
1606m packets output, 160838721 bytes, 0 underruna
0 output errors, 0 collision, e interface resets
0 unknown protocol drops
0 output buffer failures, 0 output bue
3 wappe d
DCB=up BSR=.up DTB.up RTS =Up CTS .up
Use the command shem interface interface to display the statistics about aparticular interface
\ou should select interfaces in the outbound direction, not inbound, because congestion and
resulting packet drops are more likely to occur on an outbound interface.
Look through the output to find the transmission load (txload). It is the fraction with the
meaning: the closer it gets to 1(255 255), thc higher transmission load is put on the link.
In thc output, look for complete queues and packet drops. All of them would indicate potential
congestion on an interface.
6-188 Troubleshooting Cisco Unified Communications (TVOICE] v8 0
Monitor Congestion at the Policy Map
The figure shows how to verify the queuing mechanisms that are used and how to explore if
any packet drops are present.
ionitor Congestion at Policy
HQ-ltsnow policy-map inteifac
. . truncated. . .
erial 0/1/0.101
Class-map: voice (match-all)
2073 packets, 183124 bytes
5 minute ottered rate 9B701 bps, drop rate 14763 bps
natch: ip dscp ef 1461
Priority: 106 kbps. burst bytes Z650. b/ exceed drops
cated...
Class-nap: signaling (match-any!
5023! packets, 2692896 bytes
5 minute offered rate 17254 bps, drop rate 8 51 bps
Hatch: ip dscp cs3 (2*)
50232 packets, 2692896 bytes
5 mioute rate 17254 bps
Queueing
tjueue limit 6* packets
Ij.Ki= >;V -.:./total drops/no-butler drops! !,:>/1567/239
(pkts output/bytes output) 50232/2692896
bandwidth 24 kbps
When LLQ isconfigured to check for policy map drops, use the show policy-map interface
interface command and look for packets that match the priority class. This command shows
how much traffic ismatching the priority class and how much bandwidth isused.
LLQ is afeature that provides astrict priority queue to class-based weighted fair queuing
(CBWFQ). LLQ enables asingle strict priority queue within CBWFQ at the class level. With
LLQ, delay-sensitive data (in the priority queue) is dequeued and sent first. In aVoIP network
with an LLQ implementation, voice traffic is placed in the strict priority queue.
The priority queue is policed to ensure that the fair queues arc not starved of bandwidth. When
you configure the prioriry queue, you specify in kilobits per second the maximum amount of
bandwidth that isavailable tothe priority queue. When the interface iscongested, the priority
queue is serviced until the load reaches the configured kilobits per second value in the priority
statement. Any excess traffic is dropped to avoid the problem with the Cisco legacy priority-
group feature, which would starve the lower-priority queues.
Using default or older queuing mechanisms can introduce packet loss or jitter into the RTP
stream. To verify that LLQ isconfigured properly on an interface, use thc command show
policy-map interface interface-name and look for priority.
In the figure, the class oftraffic that is called voice is configured with "Strict Priority'1 and is
allowed to use up to 106 kb/s ofthe configured bandwidth when congestion occurs. Also,
notice that there have been drops for this class oftraffic. Ifdrops are seen in this queue, one of
two problems exists. Either nonvoicc traffic is being sent to the priority queue, or there is too
much voice traffic. To prevent oversubscription ofthe priority bandwidth by too many
simultaneous calls, it is recommended thatyou usea Call Admission Control (CAC)
mechanism.
The figure also shows the signaling traffic class that is queued by using CBWFQ and the drops.
. 2010 Cisco Systems. Inc
Verify cRTP and LFI
This figure shows how to verify compressed Real-Time Transport Protocol and LFI for slow
links.
=M4j==^-=>=U.
(Continuesfrom previousfigure)
lass-map: voice (match-all)
5981462 packets, 419122340 bytes
5 minute offered rate 93010 bps, drop rate 45 bps
Match: ip dscp ef Hi)
Priority: 106 kbps, burst bytes 2650, b/v exceed drops: 32
header i F rtp
UDP/RTP (compression on, Cisco, RTP)
Ser=t: 1020 total, 979 compressed,
41957 bytes saved, 17983 bytes sent
rate 5000 bpa
Hg-i#3h w Irame-relay frags ent
Interface dlci frag-type size
Se0/l/0 101 101 end to-end 320
SeO/l/C 102
101 end to- end 610
in-frag out-frag dropped-frag
In the context of WAN links, there are three main groups of link speeds. These link speeds and
their respective design implications arc summarized here:
Slowlink is thc linkwitha speedof 768kb/s or lower:
Deployment of interactive video generally is not recommended onthese links
because ofserialization implications. These links require LFI tobe enabled if VoIP
is to be deployed over them.
cRTP is recommended (with a watchful eyeonplatform utilization).
Three- to five-class traffic models are recommended.
Medium link is the linkwiththe speedbetween 768kb/s and Tl or El:
VoIP or interactive video can be assigned to the low-latency queue (usually, there is
not enough bandwidth to do both and stili keep the low-tateney queue provisioned at
less than 2,3 percent alternatively, interactive video can be placed in aCBWFQ
queue).
LFI is not required.
cR"IP is optional.
Three- to five-class traffic models are recommended.
High speed link is thc link with the speed that is higher than Tl/Eil;
LFI is not required.
cRTP generally is not recommended(because the cost of increasedCPU levels
typically offsets thc benefits of the amount of bandwidth that issaved).
Five- to 11-class traffic models are recommended.
cRTP is amechanism that is recommended on slow-speed links or PVCs. cRTP can be enabled
as part of the MQC policy map by using the compress header ip rtp command. Ensure that
cRTP is configured on both ends of aFrame Relay PVC or aphysical link. The upper figure
shows that the cRTP is enabled and functional. This output was taken by using the show
policy-map interfacecommand.
Also, the LFI mechanism is very important on slow links to maintain low jitter. To meet thc
standard serialization delay goal ofno more than 10 to 15 ms to ensure low delay and jitter for
voice packets, you should configure afragment size of about 80 Bper every 64 kb/s of the
clocking rate for the interface.
LFI is aLayer 2technique, in which all Layer 2frames are broken into small, equal-sized
fragments and transmitted over alink in an interleaved fashion. LFI reduces delay and jitter by
expediting transfer ofsmaller frames through the hardware Tx ring.
There are two widely used LFI mechanisms in Cisco IOS Software as follows:
Multilink PPP with Interleaving (MLPPP): By far the most common form ofLFI, itworks
over PPP links.
FRF.12: Usedwith Frame Relay virtual circuits.
This is the formula to calculate fragment size for any link speed and the target serialization
delay:
(Link speed in bits per second / 8bits per byte) *target delay in seconds
Verify ifthe routers use correct fragment sizes by using the show frame-relay fragment
command for Frame Relay PVCs.
In addition, the ability to carry VoIP traffic over PVC requires that the PVC has very little burst
ability and no excess burst. You should configure the Frame Relay PVCs that are used for VoIP
with the following:
MinimumCIR should be equal to the CIR.
Committed burst (Be) is 1/100 of CIR.
Excess burst (Be) is set to zero.
Fragment data packets size should be set to interleave with voice packets.
ApplyLLQto the interfaces.
To view the active PPP interface, use the show interface multilink interface-number
command.
HQ-l#show interfaces multilink: 1
Multiiinkl is up, line protocol is up
Hardware is multilink group interface
Internet address is 172.22.130.1/30
MTU 1500 bytes, BW 64 Kbit, DLY 100000 usee,
reliability 255/255, txload 27/255, rxload 1/255
Encapsulation PFP, loopback not set
Keepalive set (10 sec)
DTR is pulsed for 2 seconds on reset
LCP Open, multilink Open
Open: IPCP
Last input 00:00:03, output never, output hang never
Last clearing of "show interface" counters 6d00h
input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0
Queueina strategy: weighted fair
Output queue: 0/1000/64/0/2441 (size/max total/threshold/drops/Lnterleaves)
Conversations 0/7/16 (active/max active/max total)
Reserved Conversations 0/0 (allocated/max allocated)
)2010 Cisco Systems. Inc. Voice Quality and Media Resources Issues 6-191
b minute input rate C bits/sec, 0 packets/sec
5 -mute output rate 7000 bits/sec, 6 packets/sec
The debug ppp multilink fragments command is avaluable troubleshooting tool when you
need to monitor MLP LFI operations. This command outputs the result of every fragmentation
operation, indicating whether the packets are fragmented into correct-sized fragments. You
should use this command with extreme caution in aproduction environment because ofthe
amount of output that it generates.
HQ ltfdebug ppp multilink fragments
Multi Lir.k Lrdyiier.ts debugging is en
Mar 1- 20:03 -OS.995: Se0/0 MLP FS: I seq C0004264 size 70
Mar 17 20:03:09.015: Se0/0 MLP-FS: I Seg 80004265 size 160
Mar 1/ 20:03:09.035: SeC/0 MLP-KS: I seq 4266 size 160
Mar -7 2i!:C3 :09.075: SeC/C MLP-FS: 1 seq 4267 size 160
Mai l"1 20:03 :C9.=79: 5e0'0 MLP-FS: I seq 40004268 size 54
Mar 17 20:03:09.091: SeC/0 MLF-FS: I soq C0004269 size 70
Mai :-=' 20:03:09.099: SeO/0 MLP-FS: I seq C000426A s_ze 70
Mar 17 2:.03: 09 10,: M-j1 MLP: Packet interleaved from queue 24
Mar 1, ,::0J:C9.107: SeO/3 MLr-FS: I seq C000426B size 70
Mar 17 J2-. 23:C9.i;9 : SeC=''0 MLP FS: I seq C000426C size 70
Mai I".' 20:03:09.12;: Mul MLP: Packet interleaved trom queue 24
Mar \i 20 :03 :C9 .131 : Mul MLP: Packet interleaved trom queue 24
Mar ,= 20:03:09. 135: SeC/C MLP FS: I seq C000426D size 70
Mar 17 20:03-09.155: Se0/0 MLP FS: I seq C000426E size 70
Sample Troubleshooting Scenarios
This topic describes how toresolve VoIP quality issues when given symptoms and information
that is gathered about the problem.
Delayed Voice
4
The problem:
AnIPphone user is reporting that longdelay is present inthe
voice path to the IP phone.
Possible causes:
- Consider all individual segments of end-to-end delay
(fixed and variable)
Maximum recommended end-to-end delay (one way) should
be less than 150 ms
SerializationDelay Propagation
/\ or Carrier Delay
Handling and
Queuing Delay
\
Dejitter Buffer
Delay
The problem: ACisco IP phone user atheadquarters isreporting that a long delay is present in
the voice path to the Cisco IP phone.
Use thefollowing suggestions togather information tohelp troubleshoot thevoice delay issue:
Configure ping packets tohave the same QoS asRTP andthen use pingor tracert in
Microsoft Windows. At the Cisco IOS CLI, use traceroute or ping commands with the
record option to test the latency of thepath. Example result: 500ms of delay is noticed
across the WAN link.
You can also measure delay(alsojitter and packet loss)by usingCisco IOSService
Assurance Agent (SAA) and Round-Trip Time Monitor (RTTMON) features at Cisco
routers. Onestrength of using SAA as the testing mechanism is thata voice call canbe
simulated. Check the following link to see how it can be used:
htlp. uu vv.cisco.com. en US'tech-'tk869,'tk76l>,lechnulogies .white paperOul S6at)0S01bLa
le.shtmi
Use CMR toverify that excessive delay that occurred and toquantify thedelay. Example
result: Delayof approximately 570 ms is occurring on the calls.
UseQRTrecordsto displayhigh and lowwatermarks for delay. Example result: The high
watermark is 585 ms, and thc low watermark is 565 ms.
Use the showcommands on the routers to verify that preferential treatment (LLQ) is being
applied toRTP traffic. Example result: TheQoS configuration uses LLQ for voice bearer;
no drops are reported.
)2010Cisco Systems, Inc
Narrow the scope ofthc problem by using the information that was gathered, and, potentially a
network diagram oracall flow diagram (or both). When dealing with delays, consider all
individual segments of end-to-end delay (fixed and variable delays). ITU-T recommendations
suggest that an end-to-end delayshouldbe less than 150ms for voiceand interactive video.
6-194 Troubleshooting Cisco Unified Communicalions (TVOICE) v80 2010Cisco Systems. Inc
Delay Budget Calculation
This section describes how to calculate delay budget.
Hjndl'ngard Q-jeuinq
Dei" (Va* .J
/L
A
128 kb/s
V
J T ; "25 -' pî bvte
in 20 ms Collecting 160 Bytes
G"2-i 'iirî.er 'I'' tT::,te-L
in 20 ms Collecting 20 Bytes
If G.729 Used over Frame Relay.
20 * 2 (cRTP) + 8 (FRF.12) = ';0 b^~
128 kb/s
V
Serialization Delay (Fixed)
Serialization delay =
Packet size in bytes x 8 bits
Link speed in b/s
-JO b/tesx 8 bits per byte)/128 kb/s) = 1.875 ms
Delay budget =2 ";-; +20rns+1.875 ms +0.005*800 +20ms +1 875ms +0.005*1200
+
Delay budget - 118.75 ms,which is less than150ms recommended
The goal ofdelay budget calculation is toverify that the one-way end-to-end delay is less than
the recommended 150ms. The overall end-to-end delay consists of these individual delay
segments:
Coder delay isfixed delay. Itdepends on the codec that isused for the call. Here are typical
examples:
G.711: For each0.125ms, a singlesamplebyte is collectedfromDSP
G.729: For each 10ms, a 10-bytes-long codeword is collected fromDSP
G.723: For each 30 ms, a 30-bytes-longcodeword is collected fromDSP
Managing and queuing delay isvariable, nondeterministic delay. This delay cannot be
precisely calculated, because it depends onthe strength ofthe router platform that is used
andits current utilization. Typically, values between 10and30ms areconsidered (10for
high-end. 20 for midrange, 30 for low-endrouterplatforms).
Serialization delay is fixed delay. It determines how long it takes to puteach bit of a frame
ontoa line(transmit). It canbecalculated from packet sizeandlinkspeedbyusingthe
formula that is shown in the figure.
Propagation delay is fixed delay, provided you know thc distance ofeach link that is used.
Generally, 5ms for each kilometer oflink distance isconsidered. Often, however, carrier
infrastructure hides the link details. In such cases, carrier network propagation values must
be obtained from service provider (carrier).
Dejitter buffer delay isalso variable delay, which cannot becalculated precisely. Dejitter
buffer is elastic andit changes its sizeautomatically, basedon thejitterthat is observed in
the network. Generally, values between 35and 50ms areconsidered (35 for networks with
low and 50 for networks with moderate jitter).
Asummary of all these segments along a voice pathinonedirection makes up thedelay
budget, as shown in the figure.
Synthetic Voice
This section describes how to troubleshoot the issue of synthetic voice.
."syiiUitjUC VOI
The problem:
An IPphone useris reporting that poor voice quality is heard at the
voice call. The end user was trained to use the QRT tool.
Possible causes:
Single packet drops orjitter longer than thedejitter buffer, which causes
the DSP to performa predetive insertion.
Carrier issues
Incorrect or missing queuing mechanism onthe egress ofthe WAN link
Awiretess IP phone e being used and due to the shared media-based
natureof wireless. coUsicns are occurring due to segment saturation
* Insufficient bandwidth across the WAN link
- LFI is not set correctly for the WANlink
The problem: ACisco IP phone user atheadquarters is reporting that poor voice quality is
being experienced when calling the Cisco IPphone at thc branch. Theenduseris trained to use
QRT.
Narrow the scope ofthe problem by using the infonnation that was gathered, and, potentially a
network diagramor a call flowdiagram (or both).
There are several possible causes of the poor voice quality:
Single packet drops orjiner that is longer than the dejitter buffer cause the DSP to perform
predictive insertions. Thc next section shows how toverify and set the dejitter buffer.
Carrier issues exist.
An incorrect or missing queuing mechanism isonthe egress of the WAN link. Use the
shou policy-map interface command to verity that thc egress queuing mechanism is set.
Awireless IP phone is being used, and because of the shared-media-basednature of
wireless, collisions arcoccurring because of segment saturation.
There is insufficient bandwidth across the WAN link. Use theshowinterfacecommand to
check the load of the link.
LFI issetincorrectly for the WAN link. Depending onthe I.FI technology, use either
Frame Relav or MLPPP with interleaving commands, as explained earlier.
You can use the earlier suggested questions togather information to help troubleshoot the
synthetic \oiee quality issue. Inaddition, usethefollowing tools:
Use QRT records to display high and low watermarks for delay. Example result: The high
watermark is 179 ms. and the low watermark is 73 ms.
5-196 TroubleshootingCisco Unified Communications (TVOICE) v8 0
Use showcommands on the routersto verifythat preferential treatment is beingappliedto
RTP traffic. Result: The QoSconfiguration appearsto be set to use LLQ, and thereare no
drops in the priority queue.
Use the Cisco IOS CLI and the Cisco Unified Communications Manager web
administrative sitetoverify thcWAN linkanddetermine thcbandwidth andcodec being
used. Example result: The WAN link is a Frame Relay 64-kb/s PVC toa remote site. The
codec that is being used is G.729.
Verify the LFI settings ontheWAN link. Example result: Thecommand frame-relay
fragment appears in the configuration.
Call thccarrier anddetermine if jitter is occurring inthecarrier network andif it is within
the SLA.
12010CiscoSystems, Inc. VoiceQuality and Media Resources Issues 6-197
Identifying Issues with Jitter
Jitter isthe difference between the expected and the actual arrival time ofa voice packet. The
difference is duetoa delay in thenetworking path.
Gatewaytshow caii active voice (entered repeatedly)
amp -
GapFillHithSilence=350 ms
^ ~.
GspFiUWithIneerpolation=0 ms 1
c
HiWaterPlayoutDelay=350 ms
LoWaterPlayoutDelay=25 ms
ReceiveDelay=29 ms
LostPackets=0
Earlypackets=0
1
SapFillWi t|-.Silence =:040 ms
GapFillWithlnterpolation-O ms
GapFillWitiiRedijndancy.O ms
HiWaterPlsyoutDelay=40 mg
LoWaterPXayoutDelay.28 ma
ReceiveDelayOS ms
LoatPacixet5 = 0
EarlyPac}xetB=.0
LacePacfcets^SS
Received packets can beheld ina dejitter buffer that controls the variations inpacket arrival
before sending it to thc codec. The amount of time that a packet isheld inthe dejitter buffer is
determined by its size; this, in turn, determines thc delay. Dejitter buffer size can be configured
by issuing the playout-dela> command.
Ageneral idea ofjitter in the network canbedetermined by repeatedly issuing theshow call
acthe \oicecommand while a call is inprogress. Ideally, these parameters should stay
relatively steady. If they do, that isanindication of smooth packet flow. However, ifjitteris
present, there arcsharp, short-term spikes such as those shown in thetwosamples in thc figure.
Theincrementing number of latepackets in thesample output reveals a degree ofjitter. Thc
silence insertion that is indicated byan increase in thcGapFiltWithSilenee value manifests
itselfas choppy voice. Thepredictive insertion, indicated byan increase inthe
GapFillWithPrediction value, tendsto manifest itselfas synthetic voice.
To alter the amount of voicesignal that is buffered to avoidjitter buffer underruns or overruns,
issue the playout-delay command. Take care inconfiguring theplayoui-delay command
because it will have animpact onthe overall delay budget of the path of the call. Larger values
result inlarger return-trip delays, but reduce the likelihood of underruns at receiving dejitter
buffers. Dependingon the sc\ erity, this could result in a choppy or robotic voice.
Note Configuring the playout-delay command is not necessarily the appropriate solution in all
cases. Severe jitter should be located and eliminated fromthe network where possible.
6-198 Troubleshooting Cisco Unilied Communications (TVOICE) v8 0
The two modesof configuration for playout delayare adaptiveand fixed:
Adaptive allowsthejitter bufferto growand shrink for the duration of the call withina
configured rangeby issuingthe playout-delay {nominal value | maximum value |
minimum {default i Ion | high}} command.
Fixedis set at the beginning of a call by issuingthc playout-delay mode {adaptive | fixed
[no-timestamps]} command.
Choppy Voice
Ihis section describes how to troubleshoot the issueof choppyvoice.
The problem:
Users are reporting that choppyvoice is being heard when calling other
IP phones. End users were trained to use the QRT tool.
Possible causes:
Multiple dropped packets are causing gaps in the voice packets that
cannot be filled withpredictive insertion on the DSP chip.
- The gateway is dropping packets due to congestion and/or improper
QoS settings.
The switch is dropping voice packets due to congestion and/or improper
QoS settings.
> Awireless IP phone is being used and due to the shared media-based
nature of wireless, collisionsare occurring due to segment saturation.
Thereare duplex mismatches on the portstowhich the IPphones are
connected.
There are duplex mismatches on trunks.
The problem: Users are reponing that choppy voice is being heard when calling other Cisco IP
phones. Thc end users are trained to use QRT.
Narrow the scope of the problem by using theinformation that was gathered, and, potentially, a
network diagram or a call flow diagram (or both).
There are several possible causes of choppyvoicequality:
Multiple dropped packets are causing gaps inthe voicepackets that the predictive insertion
on the DSPchip cannot rill. This was explained in the previous section.
The gateway is dropping packets because of congestion or improper QoS, or both.
Ihe switch is dropping voicepackets because of congestion or improper QoS settings, or
both.
A wireless Cisco IP phone is being used, and because of the shared-media-based nature of
wireless, collisions are occurring because of segment saturation.
There arc duplex mismatches on ihe ports to which the IP phones arc connected.
There are duplex mismatches on ;he trunks.
Gather infonnation to help troubleshoot thc choppy voice issue:
UseQRTrecords to display dropped packets and sent packets. Example result: Packets arc
being dropped.
From theCisco IPphone, press the blue ? button twiceto displaythe number of dropped
packets. You notice that multiple packets are beingdropped.
Use show commands on the routers to see if packet drops are occurring on thc gateway.
Result: Thc packet drops are not occurring on the gateways.
6-200 Troubleshooting Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems, Ire
Use the show mis qos interface statistics command on the switch to see if packet drops
are occurring on thc switch. Exampleresult: The packet drops arc occurring on the
switchport.
Verify that the packets that arc marked with a CoS of 5 (assuming that the Cisco IP phones
are left at the Cisco default) are being sent to the priority queue on the switch. Result: The
CoS-to-queue mappings for voiceare correct. Verifythat thc traffic-producing large IP
packets (FTP, for instance) are not beingsent to the priority queue. For instance, the
following configuration showsthat voice and FTPtrafficare queued in separateclass-based
buffers:
ip access-list extended ftp
permit tcp any any eq ftp
class-map match-all voice
match ip dscp ef
class-map match-any ftp
match access-group name ftp
policy-map WAN
class voice
priority 106
class ftp
bandwidth 64
Echo
I his section describes how to troubleshoot the issue of echo.
i.WESK*=!-i_ii,
.en e
The problem:
An IP phone user reports that distracting echo was heard
on a voice call.
Possible causes:
Loudness on talker's device set to too highhigh input
level
Call proceeds through tail circuits that do not use echo
cancellers
Call path has end-tc-end delay that is not covered by
echo cancellers
Attenuation misconfigured on voice gateways
The problem: An IP phone user reports that distracting echo is heard on a voice call.
They are two types of echo:
Talker echo occurs when thc speech energy of a talker, transmitted down the primary signal
path, is coupled into the receiving path from the far end (or tail circuit). Talkers then hear
their own voice that is delayed by thc total echo path delay time. If the echoed signal has
sufficient amplitude and delay, thc result can be annoying to thc customer and can interfere
with the normal speech process. Talker echo is usually a direct result of thc two-wire to
four-wire conversion that takes place in the PSTN.
Listener echo occurs at thc far end by circulating voice energy. Listener echo is generally
caused by the two- and four-wire hybrid transformers (caused by the echo being echoed).
The voice of the talker is echoed by the far-end hybrid, and when the echo comes back to
the listener, the hybrid on the side of thc listener echoes the echo back toward the listener.
The effect is that thc person listening hears both the talker and an echo of the raiker.
Echo is a function of delay and signal power. If the talker equipment is set to high volume, the
power of thc signal increases and. potentially, more echo is reflected and noticeable.
The telco usually applies its own port-tuning techniques to minimize echo. Kcho is constant in a
telco environment; however, low delay and low amplitude typically render echo not an issue.
Echo cancellers are tools that you can use to control echo. An echo canceller reduces the level
of echo that leaks from the R\ path (from the gateway out into thc tail circuit) into the Tx path
(from the tail circuit into thc gateway.
** Locating and Eliminating Echo
This section describes how to locate and eliminate an echo in thc network.
Locating and Eliminating Echo
Identify the type of echo: loud, long or acoustic.
Try first to move the remote phone from acoustic sources.
Try replacing the speakerphone or headset with a better quality
handset and see if the echo disappears.
If these changes do not help, continue with the echo-eliminating
procedure.
Suspected i E~h~ a~o~isIk f#,"
EchoSource1 Qiîytt^d
Echo canceller enabled'' &^ B
Echo coverage sufficient0 ^tS
! K PBX/CO
Caller is hearing
the echo!
Echo is always caused by the tail circuit on
the opposite side.
The following summarizes the process for dealing with echoes:
Step 1 Identify which tail circuit is causing the echo. Remember, the echo is caused by the
tail circuit on the opposite side of the network from the caller hearing the echo.
Step 2 Check for speakerphones or headsets. If the destination telephone is a speakerphone
or headset, this is probably the source of the echo. Try replacing the speakcrphonc or
headset with a better quality handset and see if the echo dies away normally. Or
simply try to move the phone away from acoustic echo sources.
Step 3 Check at the remote voice gateway that the echo canceller is enabled and that the
coverage is set to maximum. Echo canceller should be enabled by using the echo-
cancel enable voice-port configuration command, and the coverage should be set to
maximum value by using the echo-cancel coverage voice-port configuration
command. The echo canceller faces into a static tail circuit with an input and an
output. If a word enters a tail circuit, the echo is a series of delayed and attenuated
versions of that word, depending on the number of echo sources and delays that are
associated w:iththem. After a certain period, no signal comes out. This time period is
known as the ringing time of the tail circuitthe time that is required for all of the
ripples to disperse. To eliminate all echoes entirely, the coverage of the echo
canceller must be as long as the ringing time of thc tail circuit. Use thc command
echo-cancel coverage <(ime> to set the tail coverage, where time can be 8, 16, 24,
32. 48, or 64 ms, or 128 ms with the latest Cisco IOS Software.
Note You must apply the shutdown command to the voice port and then apply no shutdown
command to that voice port for the changes to take effect.
Step 4
If the echo persists, andyouhave verified thattheechocanceller is working
properly, you can conclude that the echo canceller cannot fix the echo "orone of two
reasons:
The echo is too loud (called a loud echo).
The echo is too delayed (called a long echo).
The echo is acoustically distorted.
Identity' which type of echo you arc experiencing, either long or loud.
6-204 Troubleshooting Cisco Unified Communications (TVCICE) vB 0 ) 201G Cisco Syslems. Inc
* Identifying a Type of Echo
After you have verified that the echo canceller is enabled, you still need to determine thc cause
of the echo: Is the probleminsufficient echo return loss (ERL) in the tail, or is the echo delayed
beyondthc coverage of the echocanceller? Most persistentechoes are loudechoes. Delayed
echoes are common, however, when the tail circuit involves a long-distance PSTN link, a series
of alternating digital and analoglinks, or any other linkwith highlatency.
Identifying a Type of Echo
Gateway*ahow coll acti*
- snip -
NoiaeLevel.o
ACOMLevel-0
OutSigoalleveU-7
InSignalLevel-n
InfaActivity.2
ERLLevel-"?
PBX/CO
ERL should be >15dB
based on ITU-T G.131
Output Level Rout
Input Level Sin
ERL = Rout-Sin
ERL = (-7)-(-14) = 7 dB
Following are descriptions of the primary measurements of relative signal levels that are used
by echo cancellers. They are all expressed in decibels.
Echo return loss (ERL): The reduction in the echo level that is produced by the tail circuit
without the use of an echo canceller. So if an Rx speech signal enters the tail circuit from
the network at a level of X dB, the echo coming back from the tail circuit into the Sin
terminal of the echo canceller is (X - ERL). ERL = ERL through tail - Rout - Sin (dB).
Echo return loss enhancement (ERLE): The additional reduction in echo level that is
accomplished by the echo canceller. An echo canceller is not a perfect device; the best it
can do is attenuate thc level of the returning echo. ERLEis a measure of this echo
attenuation that is performed by the echo canceller. It is the difference between the echo
level that is arriving from the tail circuit at the echo canceller and the level of the signal that
is leaving the echo canceller. ERLE= ERLE through echo canceller = Sin - Sout (dB).
ACombined (ACOM): Thc total echo return loss that is seen across the Rin and Sout
terminals of the echo canceller. ACOM is the sum of ERL + ERLE, or the total echo return
loss that is seen by the network. ACOM = Combined echo return loss through system = Rin
-Sout(dB).
>2010 Cisco Systems. Inc. Voice Quality and Media Resources Issues 6-205
Identifying a Loud Echo
You can use the voicegateway itself to measure thc ERL of the tail circuit by usingthe
gateway echocanceller statistics reporting function. Fora Cisco VoIP gateway, output from the
showcall active voice command contains valuable statistics. To generate thesestatistics, first
establish a voice call over the gateway. Then type the show call active voice command without
pressing the Return key. Make a loud, continuous soundintothe mouthpiece or holddowna
button on your touch-tone keypadto generate a sound, and then press Return to display the call
statistics.
Note Youcan also use commercial test devices (including handheld telecommunications level
meters) to measure ERL for a particular destination circuit.
Remember, you need to look at thc destination voice gateway. The tigure shows that the ERL is
the ditfcrencc in thc reported Tx and Rx levels. Ideally, you want your gatewav to have an ERL
of at least I5 dB. If your ERL is less than 10dB, youprobably have insufficient ERL in the tail
circuit. Perform the tests that are outlinedby usinglouderand softer noises. Verify that the
FRI. is consistent and that, when you varyyour volume, the levelsvaryaccordingly. If these
testsare consistent, you can be confident that the tail circuit is not providing enough echo loss
tor the echo canceller to be able to eliminate thc echo.
Identifying a Long Echo
You can identifya long echo problemwith a technique like the one described previously for
loud echoes. The signature of a loud echo problem is that the echo is somewhat attenuated but
still noticeable. The echo is the same whether or not the echo canceller is enabled. If you
determine that the ERLis reasonable (greater than lOdB) but the echo is still persistent, then
the problem might be a long echo.
If the problem is a longecho, youcannot do muchto solve it. If the tail includes a long-distance
hop, make sure that the PBXthat is terminating the long-distance hop has its own echo
canceller turnedon. If possible, extendthe digital portionof your network as closeas possible
to the endpoint.
Locating and Eliminating Echoes in the Tail Circuit
Because of the variety of possible network scenarios, it is difficult to give specific instructions
for finding and eliminating an echo ina tail circuit. Youcan perform some general steps,
however, to track down the source of an echo and eliminate it.
Drawa diagramof the tail circuit, and include all the digital and analog links between the
destination voice gateway and the destination telephone. This diagramwill likely forma tree:
fromthe voice gateway out. each device will have one or more potential destination branches.
You need to identify the break point off the main branch for which calls give consistent echo.
For example, the gateway might be connected to a PBX with three output cards. It'many of thc
calls through one of these ports exhibit echo, then you have narrowed the source cf the problem
tail to the circuits that are attached to that voice port. Look for clusters of echo that are
associatedwith common links. If you trace your tail out lo the uncontrolled PSTN, then
remember that a certain percentage of PSTN tails always will not provide sufficient ERL, and
you will be unable to correct them. When you find a link that is giving insufficien. ERL,
examine the levels and provisioning of the devices at both ends of the link.
6-206 Troubleshooting Cisco Untied Communications (TVOICE) v8 0 2010 Cisco Systems Inc
'** Eliminating the Echo
mm
This section describes how to further eliminate the echo in the network.
Eliminating the Echo
ln|ecting 1004 Hz Tone at 0 dB
1
-7dB PBX/CO
;nu
Adjust audio levels,
typically at the PBX side.
RaAot-gw#BboH ca]
ACOMLevel.O
OutSignalLevel--7
Ino*ctivlty2
ERLLsvel-20 *
Output level Rout
Input level Sin
ERL = Rout-Sin
ERL = (-7) -(-27)= 20 dB
ERL is now > 15 dB
/t
*>
To thoroughly check the network, you can use a commercial test set for thc PBX and use this
process to eliminate the echo:
1. Verify proper impedance levels on the PBX.
2. Verify proper audio levels.
3. Measure the ERL of the PBX.
Note Note that you do not need an official test set for echo testing. You can use dual tone
multifrequency (DTMF) tones or your own voice to get an approximate indication of level
mismatches.
Impedance level requirements vary per PBX brand and type of circuit that connects the PBX to
Cisco IOS gateway. Generally, the Cisco IOS gateway impedance configuration must match
the PBX expectations. Example of a typical impedance value of thc four-wire ear and mouth
(E&M) circuits is around 600 ohms:
Gateway(config-voiceport]#impedance
600c 600 Ohms complex
600r 600 Ohms real
900c 9 00 Ohms complex
complexl complex 1
comp1ex2 complex 2
Verify proper audio level settings from thc local site to the remote site (for instance, from the
local PBX to the remote PBX). After a call is established, a 1004-Hz tone at 0 dB is injected
into the path. Then measure the audio levels at various points along the voice path. Enter a
show call active voice command on the local router to verify the audio levels. For instance, the
level on the local router is measured as -3 dB, which is the correct level according to the Cisco
guidelines:
>2010 Cisco Systems, Inc Voice Quality and Iviedia Resources Issues 6-207
Gateway^show call active voice
- snip -
MoiseLevel-0
ACOMLevel-0
OutS".gnalLevel = -"T9
InSignalLevel-3
Intheprevious figure, the ERL level was -14 dB. which means that, inrelation tothesignal
going into the PBX, theechois coming back at a level only 7 dBlessthan what wasgoing in.
The (ITU-T) recommendationG.131 states that the ERL of a PBXshould be greater than 15
dB. The LRL was suhstantially higher than an echo canceller can effectively nullify; therefore,
the echo problem was \\ ith the PBX. To further verify the problemlocation, adiust thc audio
level into the PBXup and down. When thc audio level is adjusted, the ERLshould remain
constant.
The figure shows that after the audio level was adjusted at the PBXside (Sin is-27 dB now),
the measured ERL of 20 dBwas reached. The call should not exhibit any echo.
Another way to tunc PBXaudio levels is to configure signal attenuation or input gain at the
CiscoIOSgateway. The OutSignalLevel showsthe valueof thelevel after the output
attenuation is appliedto the signal, and the InSignalLevel showsthc level after the input gain is
applied. Configure attenuation in each direction. Increasedecibel value until echo disappears.
For example, 1dBof gain (inputdirection) or attenuation (outputdirection) is configured on an
H.323 or SIP gateway as follows:
voice-port 1/1:23
input gain -1
output attenuation 1
Note You must apply the shutdown command to the voice port and then apply the no shutdown
command to that voice port for the changes to take effect.
MGCP gateway attenuation is configured at theCiscoUnified Communications Manager
Administration gateway page.
6-208 Troubleshooting Cisco Unified Communications (TVOICE) v8.0 12010 Cisco Systems Inc
One-Way Audio Issue
One-way audio and no audio at all (no-way audio) are common problcms that occur during a
new IP telephony network installation.
One-Way Audio Issue
The problem:
When a phone call is established through a Cisco IOS voice
gateway or router, only one of the parties receives audio
(one-way communication).
Ai!-1i,iOjM"h!\J'.:tl-.
No Audio
Possible causes;
One-way IP reachability exists only
cRTP not configured on both ends of the link
Early version of NAT in the path blocking voice
Access list configured in the blocked direction
Miscon figurations cause the majority of these problems. For one-way audio problems, always
pay attention the direction in which the one-way audio is occurring. For no audio in either
direction, thc troubleshooting methodology is the same. You might need to repeat the procedure
for each direction of audio, but, more likely, you will find the source of the problem when you
try to troubleshoot one direction.
You can take several steps to troubleshoot a one-way or no-way audio problem that occurs as
soon as the call connects:
Stepl Verify bidirectional IP connectivity.
Step 2 Check Cisco IOS Software gateway configurations, especially cRTP on both ends of
the link.
Step 3 Check for NAT or firewall restrictions.
First, verify IP connectivity when troubleshooting a one-way or no-way audio problem. IP
connectivity must be present for voice packets to be exchanged between two devices. The lack
of IP connectivity causes many one-way or no-way audio problems.
A common cause of one-way or no-way audio is the existence of Network Address Translation
(NAT). Port Address Translation (PAT), or firewalls between two endpoints. The SCCP
protocol embeds IP addresses in the payload of the IP packet to signal to which IP address to
send RTP packets. If the device that is performing NAT or PAT is unaware of this fact, the
embedded IP addresses are not translated. This issue results in one-way or no-way audio. You
must run Cisco IOS Software Release 12.1(5)T or later.
2010 Cisco Systems, Inc. Voice Quality and Media Resources issues 6-209
Firewalls canalsobea problem if they areunaware of thevoice traffic that is passing through
them. Firewalls arc oftenconfigured to blockall UDPtrafficthat goes through them. Because
voice traffic is carried over UDP, it might be blocked while the signaling that is carried over
TCP is passed. Fixup protocol makes sure that UDP port numbers are passed through the
firewall. This involves deep packet inspection to open up ports requested for that voice call.
A sniffer is the best tool for debugging such a scenario. If both devices appear to be
transmitting audio but the audio is not reaching the opposite side, take a sniffer trace at each
hop along the way until you find the hop where the audio is not passing through. If the firewall
is blocking UDPpackets, youmight needto opena hole in it lo allowthe voicetraffic to pass
through.
*"* Policy Map Attached to One End of the Link Only
This section descnbes the issue when an IP phone cannot load firmware or configuration over
the IP WAN because a policy map that is implementing LLQ is being attached at one end of the
serial link only.
Policy-Map Attached to One End of the
Link Oniy
The problem:
IP phone at branch fails to load firmware or configuration repeatedly.
Attempting to Load Firmware
Central Branch
Possible cause:
Policy map with LLQ attached at one end of the link onty
If you experience a Cisco IP phone at the branch failing to load firmware or configuration,
verify the QoS policy map configuration on the central and branch routers.
If IP connectivity is maintained between the central and branch sites, very likely, the policy
map with thc LLQ priority command is attached to one end of the serial link only.
Summary
uinru
Voice quality issues in Cisco Unified Communications
networks are caused by lack of bandwidth, long end-to-end
delay, jitter, and echo.
Voice call general requirements are latency of less than 150
ms, jitter of less than 30 ms, loss of less than 1 percent, and
17 to 106 kb/s guaranteed priority bandwidth
Isolate potential voice issues to the most likely causes,
identify the most likely cause of the problem, and take
corrective action, being careful to change only one variable at
a time Then analyze the results.
In a LAN environment, buffers can fill instantaneously so that
the switch drops packets when they attempt to enter the full
interface buffer. This could result in degradation of voice
quality. Another possible source of problems occurs when
there are bandwidth mismatches.
Consider these possible causes when troubleshooting voice
quality issues on a gateway: Speed mismatches, bandwidth
issues on slow WAN links, serialization on slow WAN links,
carrier network issues, and SLAthat is not designed for
voice.
The most common issues that are experienced in converged
networks include delayed voice, synthetic voice, choppy
voice, or echo issues.
In this lesson, you have learned to explain common voice quality issues and identify the most
likelv causes of these issues.
* References
Fxho Analysis for Voice over IP white paper at
http: www.cisco.comenl,Scustomer;docs.;ios/soluttons_docs,Voip_solutions/
EAJSD.html
Troubleshooting Echo Problcms between IP Phones and Cisco IOS Gateways at
http: www.cisco.com en US'CUStoincr'tech.'tkh^/tkCîS/
teehnologics_tech_notc(l91 S6a()0K0149a ?f.shtml
Measuring Delay, Jitter, and Packet Loss with Cisco IOS SAA and RTTMON at
http: www.eiseo.eonfenTJS/tech/tk869.t.k769/
technologtcs_\shitejiaper0918()aOOSr)3blale.shtml
2010CiscoSystems, Inc. Voice Quality and Media Resources Issues 6-213
6-214 TroubleshootingCisco Unified Communications (TVOICEl v8 0 2010 Cisco Systems. Inc
Module Summary
This topic summari7cs the key points that were discussed in this module.
iocfule Summary
When troubleshooting MOH issues, verify that the MOH
server is registered, that it is not running out of resources,
that the MRG lists the correct servers and when multicast
MOH is desired, and that IP multicast routing is enabled.
When MTP is required but does not work, ensure that MTP is
registered, that it is not running out of resources, and that it is
reachable from the endpoints requiring its services.
Ad Hoc and Meet-Me conferencing issues are rooted at
hardware conference bridges and Cisco Unified
Communications Manager. Verify that a conference bridge is
registered and available; that location, region, and media
resource group configurations are not mismatched; and that
the codec is supported by a conference bridge.
loduie Summary (Cont)
The common causes for transcoder issues include a
transcoder that is not registered or that is busy, media
resource misconfiguration, network connectivity issues, or
DSP farm issues.
RSVP agents must be registered for RSVP CAC to work.
When a call is blocked by RSVP CAC, consider whether
bandwidth is exhausted at locations or the IP WAN, whether
IP phones are not associated with RSVP agents or agents
are misconfigured, or whether network connectivity to an
agent or between agents has been lost.
Voice quality issues in Cisco Unified Communications
networks are caused by lack of bandwidth, long end-to-end
delay, jitter, and echo. Identify the issues by searching for
speed mismatches or lack of bandwidth and serialization on
slow WAN links, and verify that the carrier network works as
expected.
In this module, you learned howto explain common voicequalityissuesand identifythe most
likely causes of these issues.
References
for additional information, refer to these resources:
Cisco Unified Communications Manager Features and Services Guide, Release 8.0(2),
Music On Hold at
http: www.eijjco.eom en US partner docsAoiec_ ip eomnv'cucm admin 8_0 2
cemteat fsmoh.htm!
Cisco Unitied Communications Manager System Guide, Release 8.0(2), Transcoders at
hup: www .cisco.com en I S partner docs'voiee ip comnv'eucin.'iidmiii S 0 2
ecnisv s .i05tra;iv!.ti..l
Cisco IOS Quality of Service Solutions Configuration (iuide, Release 15.0, Configuring
RSVP at http: www.ci-.co.com en US docsios qosconliguraiion/guide'
eonfiji ri\p p>1059l T^D Products Configuration Guide Chaptcr.html
Cisco IOS and NX-OS Software. Configuring SIP RSVP Features at
http1 www.cim-'o com en US docVios \ oicesip'Cnnfigui at ion'guide sip cg-rsvp.hunl
Cisco Unitied Communications Manager Administration Guide, Release 8.0(2). Location
Contiguration at https: www.eiico.com enlJS'doc* voice ip comnicucni admin 8 0 2
cemefg hl>2lnL.it html
Echo Analysis for \'oice over IP white paper at
http: www .cisco.com en LS customer does.'ios Solutions docs
\Dip solutions L \_iSL) html
Troubleshooting Echo Problcms between IP Phones and Cisco IOS Gateways at
hup: www.cisco.com en LS cusiomer'iech'tk.652 lkolIH
tcch.io luês iccti noieOl}lS6;iOl)80149al f.shtml
Measuring Delay, Jitter, and Packet Loss with Cisco IOS SAA and RTTMON at
http: www Lis=jn.com en US lech tkS69'tk"W
technologies whiie_ p,tper09 186a00801 b la Ie.shiml
TroutjleshootingCisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems, Inc
Module Self-Check
solutions arc found in thc Module Self-Check Answer Key.
Q1) Which three statements characterize unicast MOI 1?(Choose three.) (Source:
Troubleshooting MOII Issues)
A) Unicast MOH is a point-to-point, two-way audio.
B) Unicast MOH is a point-to-multipoint, one-way audio.
C) Unicast MOH is a point-to-point, one-way audio.
D) Multiple users or connections share a single stream.
E) Unicast MOH uses a separate source stream for each user or connection.
F) As more endpoint devices go on hold via a user or network event, the number
of MOH streams increases.
Ci) As more endpoint devices go on hold via a user or network event, thc numbcr
of MOH streams stays the same.
Q2) What is a typical cause of a call being disconnected when placed on hold'.' (Source:
Troubleshooting MOH Issues)
A) The voice codec for a given device is not allowed by the region.
B) The MOHserver is registered, but all MOH resources are currently in use.
C) Thc voice codec for a given device, as defined by its region, is not in the list of
codecs that are supported by the MOH server.
D) MOH is excluded from the MRG and, therefore, is unavailable to the endpoint.
Q3) When using the Cisco Unified RTMT, what are two factors to check when the number
of currently active MOH devices flaps between one and two when they stream music to
a single device? (Choose two.) (Source: Troubleshooting MOH Issues)
A) Make sure that the MOH server is registered with Cisco Unified
B) Make sure that you have uploaded the MOH file to all the servers in the Cisco
UnifiedCommunications Manager cluster, not only to the publisher.
C) Verify that the MOH server has available resources.
D) Verify that multicast routing is enabled in the network.
E) Make sure that the MOH source filename docs not contain a space.
Q4) What are four typical reasons why the MOII server does not register with Cisco
UnifiedCommunications Manager? (Choosefour.) (Source: Troubleshooting MOH
Issues)
A) Network connectivity issues exist.
B) Cisco Unified Communications Manager might be overly utilized.
C) The Cisco Unified Communications Manager database might be corrupted.
D) The MOII server does not reside on the same LAN with Cisco Unified
E) The Cisco MOH service might not be running.
F) The Cisco IP Voice Media Streaming Application service might not be
running.
G) All the MOH sources (audio files to play) were removed from the MOH server.
II) The MOH resource can be disabled or a software error could occur.
2010 Cisco Systems. Inc. Voice Qualityand Media Resources Issues 6-217
Q5) What are two options for tuning the volume of played music or tones to a comfort
level? (Choose two.) (Source: Troubleshooting MOH Issues)
A) Get a louder type of music and import it to MOII server.
B) Thc advanced Cisco IP Voice Media Streaming Application service parameter
Default MOH Volume Level can be adjusted. Changes to this parameter only
affect audio files that are played after the change occurred.
C) The advanced Cisco IP Voice Media Streaming Application service parameter
Default MOII Volume Level can be adjusted. Changes to this parameter only
affect audio files that arc imported after thc change occurred.
D) Increase thc loudness by using Cisco Unified IP phone settings.
E) You can modify thc decibel level of .wav files to adjust music volume by using
various audio processing applications.
Q6) Cisco Unitied Communications Manager uses the User and Network Hold MOH Audio
settings to play the MOII audio source to the users. You can configure the MOII audio
source at various places in Cisco Unified Communications Manager. Order the
configuration options according to their priority. (Source: Troubleshooting MOH
Issues)
A) Common Deviee Contiguration
Bj line settings
C] Cisco CallManager service parameters
D) phone settings
1. first
_ _ 2. second
3. third
4. fourth
Q7) Cisco IOS routers might not be enabled for IP multicast when multicast MOH is
preferred. Each router on the path must be globally enabled for IP multicast routing by
die command . and the router interfaces must be enabled
to participate in multicast routing by the or
interface configuration command. (Source:
Troubleshooting MOII Issues)
Q8) Multicast MOH from the branch router flash is a feature that allows multicast MOH
streams to be generated by gateways that are located at remote sites instead of
streaming MOH from the main site to the remote site over the IP WAN. (Source:
Troubleshooting MOH Issues)
A) true
B) false
6-218 TroubleshootingCisco Unified Communications (TVOICE) v8.0 2010 Cisco Systems. Inc
Q9) What arc four major scenarios in which MTP is used? (Choose four.) (Source:
Troubleshooting MTP Issues)
A) repacketization of a stream when an MTP can be used to transcode G.729a to
G.729b
B) repacketization of a stream when an MTP can be used to transcode a-law to
mu-law and vice versa
C) to extend supplementary services to H.323 endpoints
D) inbound H.323 fast start
E) outbound H.323 fast start
F) outbound SIP early offer
G) outbound SIP delay offer
H) named telephony events (RFC 2833) DTMF relay
I) H.245-alphanumeric DTMF relay
mm QIO) Which Cisco IOS command shows a registration name of the hardware MTP? (Source:
Troubleshooting MTP Issues)
A) show seep registration
B) show seep mtp
C) show seep ccm group
D) show seep ccm profile
OH) If the reason for the MTP allocation is an activated MTP Required check box at a SIP
trunk and the allocation fails, then which two of the following might happen? (Choose
two.) (Source: Troubleshooting MTP Issues)
A) If the Cisco CallManager service parameter Fail Call Over SIP Trunk if MTP
Allocation Fails is set to True, the call fails.
B) If the Cisco CallManager service parameter Fail Call Over SIP Trunk if MTP
Allocation Fails is set to False, the call fails.
C) If the Fail Call Over SIP Trunk if MTP Allocation Fails parameter is set to
False, call setup depends on whether the call actually requires an MTP. If early
offer is required, the call fails; if delay offer is required, thc call is set up even
without an MTP.
D) If the Fail Call Over SIP Trunk if MTP Allocation Fails parameter is set to
False, call setup depends on whether the call actually requires an MTP. If it
does, the call fails; if it does not, the call is set up without an MTP.
E) If an MTP cannot be allocated, the call fails unconditionally.
Q12) The software-basedconference bridge that is implementedas a Cisco Unified
Communications Managerservicesupports onlyconferences that are usingG.711 mu-
law, (Source: Troubleshooting Issues with Conferences)
A) true
B) false
Q13) If the software conference bridge is deployed, it must be enabled to register properly. A
software conference bridge can be enabled on the
service parameters page by setting the ___^_ option in the CFB parameters
section to True. (Source: Troubleshooting Issues with Conferences)
2010 Cisco Systems, Inc. VoiceQualityand Media Resources Issues 6-219
Q14) Whichtwo of the followingcould be reasons for the message "No Conference Bridge"
to be displayed at an IP phone when a conference originator wants to add another
conference participant? (Choose nvo.) (Source: Troubleshooting Issues with
Conferences)
A) The conference participant phone is not associated with any conference bridge
through MRGL.
B) Ihe conference originator phone is not associated with any conference bridge
through MRGL,
C) Network connectivity issues exist between thc conference originator and the
conference participant.
D) The conference bridge is not registered.
E) The conference bridge is registered but out of resources.
QI5) If the Meet-Me pattern is dialed as a new call to join a conference, but a caller gets a
reorder tone, what arc four of thc most likely causes? (Choose four.) (Source:
Troubleshooting Issues with Conferences)
A) This is normal. The conference should be joined by using thc Meet-Me
softkey.
B) Another Meet-Me conference is in progress at the dialed pattern.
C) The Meet-Me conference is not yet set up.
D) "Ihe Cisco Unified IP phone model does not support Meet-Mc conferencing,
t) I he CAC is rejecting thc call to the Mect-Me pattern.
F) The conference bridge does not support the caller codec.
G) I he conference bridge is not registered.
H) The Meet-Me pattern is in the partition that is not accessible by the calling
search space of a caller phone.
I) The Meet-Me conference bridge does not support the codec of the caller.
Q16) A transcoder takes the stream of one codec and converts it from one compression type
to another. In addition, a transcoder provides MTP capabilities and can enable
supplementary services for H.323 endpoints when required. (Source: Troubleshooting
Transcoder Issues)
A) true
B) false
Q17) Which port number is used for the communications between a hardware transcoder and
the Cisco Unitied Communications Manager cluster? (Source: Troubleshooting
Transcoder Issues)
A) 1720
B) 5060
C) 2428
D) 2000
E) 2427
QIH) If you experience problems with a transcoder allocation and the transcoder is properly
registered with Cisco Unitied Communications Manager, ensure that the transcoder
does not run out of resources. The resources of the Cisco IOS transcoder can be
verified by using the command. (Source:
Troubleshooting Transcoder Issues)
6-220 Troubleshooting Cisco UnifiedCommunicalions (TVOICE) v8.0 2010 Cisco Systems, Inc
Q19) What are three advantages of RSVP-enabled locations over standard locations in Cisco
Unified Communications Manager? (Choose three.) (Source: Troubleshooting Issues
with RSVP Agents)
A) RSVP is less resource-intensive to Cisco Unified Communications Manager
servers.
B) If one link in the IP network goes down and packets are routed at different
paths. RSVP is aware of the change and considers the bandwidth that is now
available at the actually routed path.
C) RSVP can improve the utilization of bandwidth in a WAN.
D) RSVP takes care of routing a call via a path that has the most available
bandwidth.
E) If backup links are added after link failures or if bandwidth on demand is used
to add dial on demand circuits, RSVP is fully aware of the currently used
routing path and thc bandwidth available on each link along that path.
F) RSVP can block the call immediately when the remote destination is busy,
making bandwidth available to other calls.
G) If load sharing is used, RSVP is aware of the overall bandwidth that is
provided by multiple load-shared links.
Q20) The RSVP agent cannot register with Cisco Unified Communications Manager. Given
the following dspfarm profile configuration, what is incorrect or missing? (Source:
Troubleshooting Issues with RSVP Agents)
codec g7iiulaw
codec pass - through
maximum sessions software 2
i^p, A) The rsvp keyword shouldbe used insteadof mtp.
B) There are not enough sessions configured. For the RSVP agent to work, at least
three sessions must be specified.
C) The enable rsvp command is missing.
D) The rsvp command is missing.
Q21) Which four configurations represent the RSVP per-flow bandwidth that must be
configured for a call to be admitted if you are using the specified codec? (Choose four.)
(Source: Troubleshooting Issues with RSVP Agents)
A) for G.711 and G.722, at least 96 kb/s
B) for G.711 and G.722, at least 80 kb/s
C) for G.729, at least 40 kb/s
D) for G.729, at least 16 kb/s
E) for iLBC, at least 40 kb/s
F) for iLBC, at least 24 kb/s
G) for iLBC, at least 16 kb/s
H) for G.723, at least 16kb/s
1) for G.723, at least 40 kb/s
)2010 CiscoSystems, Inc Voice Quality and Media Resources Issues 6-221
Q22) SIP Preconditions for RSVP-enabled CAC is supported between Cisco Unified
Communications Manager, Cisco Unified Communications Manager Express. Cisco
IOS SIP gateways, and third-party SIP devices. (Source: Troubleshooting Issues with
RSVP Agents)
A) true
B) false
Q23) How much delay is generally acceptable for networks that transport real-time traffic?
(Source: Troubleshooting Voice Quality Issues)
A) two-way delay below 150 ms
B) one-way delay below 150 ms
C) two-way delay below 350 ms
D) one-way delay below 350 ms
Q24) What is the required bandwidth per single G.711 call when the call is transponed over
Ethernet with IEEE 80LQ VLAN framing at 50 p/s? (Source: Troubleshooting Voice
Quality Issues)
A) ISO kb s
B) 84 kbs
C) 90 kbs
D) 93 kbs
Q25) During a voice call, you can view the call statistics information from the Cisco IP
phone. This information can show properties of the call, such as codec, packetization
period, packets sent, packets received, packets lost, signaling overhead, delay, and
jitter. (Source: Troubleshooting Voice Quality Issues)
A) true
B) false
Q26) Which egress queueing option is recommended on a Cisco Catalyst 3750 Series Sw itch
when transporting voice? (Source: TroubleshootingVoice Quality Issues)
A) 1P1Q3T
B) 2Q3T
C) 1P3Q3T
U) 4Q3T
Q27) Which fragment sizeshouldbe configured at a FrameRelayPVCof 256 kb's for the
target serialization delay of 10 ms? (Source: TroubleshootingVoice Quality Issues)
A) SO bytes
B) 160bvtes
C) 320bvtcs
D) 640 bvtes
Q2K) When identifyingwhich tail circuit is causing the echo, remember that thc echo is
caused b> the tail circuit on thc same side as the caller hearing the echo. (Source:
Troubleshooting Voice Quality Issues)
A) true
B) false
6-222 Troubleshooting Cisco Unitied Communications (TVOICE) vB 0 2010Cisco Systems, Inc
jg^^^
iMr
*
Ql) C.E.F
Q2)
C
03) B. b
04) A. C. F. H
05) C. E
06)
1-B, 2-D. 3-A.4-C
Q7) ip multicast-routing, ip pim or ip pim sparse-dense-mode
Q8) A
09) B.C. E. H
010) C
QUI A.D
QH)
B
Q13) Cisco IP Voice Media Streaming Application. Run Flag
014) B.D
01?) C. E. E. H
QIC') A
017) D
QI8) show dspfarm profile
QI9> B. E.G
Q20) D
021) A, C. E, I
Q22) B
023) 13
Q24) 1)
025) B
026) C
Q27) C
028) B
6-224 Troubleshooting Cisco Unified Lommiimciilions (TVOICE) vB0 2010 Cisco Syslems, Inc
Table of Contents
Lab Guide
tfH Overview 1
Outline 1
Pullout Lab Layout 2
* Description of Simulated PSTN 3
Simulated PSTN Phone 4
Calls Inbound to Simulated PSTN 5
Calls Outbound from Simulated PSTN 8
Importing Cisco Unified Communications Manager Configuration 11
* Uploading Configuration File 11
Importing Configuration File 11
Lab 2-1: Troubleshooting Gateway and Endpoint Registration Issues 14
"* Activity Objective 14
Required Resources 14
^ JobAids 14
Trouble Ticket 1: Troubleshooting Cisco IP Phone Registration 14
Trouble Ticket 2: Troubleshooting MGCP Gateway Registration 17
mU Trouble Ticket 3: Troubleshooting H.323 Gateway Communication 19
Lab 2-2: Troubleshooting LDAP Integration Issues (Optional) 23
Activity Objective 23
am Required Resources 23
Job Aids 23
Trouble Ticket 1: Troubleshooting LDAP Integration 23
mm Lab 3-1: Troubleshooting On-Net Single-Site Calling Issues 27
mm Required Resources 27
Job Aids 27
Trouble Ticket 1: Troubleshooting On-Net Single-Site Deployment 27
n Lab 3-2: Troubleshooting On-Net Multisite Calling Issues 31
ActivityObjective 31
Job Aids 31
Trouble Ticket 1: Troubleshooting Centralized MultisiteDeployment 31
^ Trouble Ticket 2: Troubleshooting Distributed Multisite Deployment with Gatekeeper 34
Trouble Ticket 3: Troubleshooting Distributed Multisite Deployment with Cisco Unified Border
Element at HQ 36
mm Trouble Ticket 4: Troubleshooting Distributed MultisiteDeployment with Cisco UnifiedBorder
Element at BR 39
Lab 3-3: Troubleshooting Off-Net Calling Issues 42
" ActivityObjective 42
Job Aids 42
tmm Trouble Ticket 1:Troubleshooting Off-Net Calling Issuesat HQ 43
Trouble Ticket 2: Troubleshooting Off-Net Calling Issues at BR 47
mg Lab3-4: Troubleshooting Globalized Call Routing Issues 50
mi Job Aids 50
Trouble Ticket 1: Troubleshooting Globalized Call Routing Issues 51
Lab 4-1: Troubleshooting SAF Client and Forwarder Issues 56
"mm Activity Objective 56
mm Job Aids 56
Trouble Ticket 1: Troubleshooting SAF Client to SAF Forwarder Communication 57
Trouble Ticket 2: Troubleshooting Hosted Directory Number Patterns Learning Process 60
M Trouble Ticket 3: Troubleshooting PSTN Failover 63
Lab 5-1: Troubleshooting Device MobilityIssues 67
Job Aids 67
Trouble Ticket 1: Troubleshooting Device Mobility 68
Lab 5-2: Troubleshooting Cisco Extension Mobility Issues 72
Job Aids 72
Trouble Ticket 1: Troubleshooting Intracluster Cisco Extension MobilityIssues 73
Lab 5-3: Troubleshooting Cisco Unified MobilityIssues " 77
Job Aids 77
Trouble Ticket 1: Troubleshooting Inbound Calling 78
Trouble Ticket 2: Troubleshooting Inconsistent Mobile Phone Ringing 81
Lab 5-4: Troubleshooting Cisco Unified Communications Manager Native Presence Issues
(Optional) 85
Job Aids 85
Trouble Ticket 1: Troubleshooting Cisco Unified Communications Manager Native Presence
Issues 85
Lab 6-1: Troubleshooting MOH Issues 89
Job Aids 89
Trouble Ticket 1: Troubleshooting MOH 89
Lab 6-2: Troubleshooting Transcoder Issues 93
Job Aids 93
Trouble Ticket 1: Troubleshooting Transcoder Registration 93
Trouble Ticket 2: Troubleshooting Transcoder Allocation 96
Lab 6-3: Troubleshooting Issues with RSVP Agents 99
Job Aids 99
Trouble Ticket 1: Troubleshooting Issues with Intracluster RSVP Agents 99
Trouble Ticket 2: Troubleshooting Issues with Intercluster RSVP Agents 102
Troublesnooting Cisco Unified Communications (TVOICE) v8 0 2010 Cisco Systems, Inc
TVOICE
Lab Guide
Overview
This guide presents the instructions and other information concerning the lab activities for this
course.
Outline
This guide includes these activities:
Pullout Lab Layout
Description of Simulated PSTN
Importing Cisco Unified Communications Manager Configuration
Lab 2-1: Troubleshooting Gateway and Endpoint Registration Issues
Lab 2-2: Troubleshooting LDAP Integration Issues (Optional)
Lab 3-1: TroubleshootingOn-Net Single-Site Calling Issues
Lab 3-2: TroubleshootingOn-Net Multisite Calling Issues
Lab 3-3: TroubleshootingOff-Net Calling Issues
Lab 3-4; TroubleshootingGlobalizedCall-Routing Issues
Lab 4-1: Troubleshooting SAF Client and Forwarder Issues
Lab 5-1: Troubleshooting Device Mobility Issues
Lab 5-2: TroubleshootingCisco Extension Mobility Issues
Lab 5-3: Troubleshooting CiscoUnifiedMobilityIssues
Lab 5-4: Troubleshooting CiscoUnifiedCommunications ManagerNativePresenceIssues
(Optional)
Lab 6-1: Troubleshooting MOH Issues
Lab 6-2: Troubleshooting Transcoder Issues
Lab 6-3: Troubleshooting Issues with RSVP Agents
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Description of Simulated PSTN
This section describes the functionality of the simulated PSTN.
The lab layout consists of two student pods: pod 1 and pod 2. Each pod consists of HQ-x and
BR-x site. These sites are virtually placed in the following different geographical regions:
The HQ-x site is located in a virtual European country.
The BR-x site is located in a virtual North American country.
This placement detennines the dialing rules that are to be followed when dialing to the
simulated PSTN. The HQ-x site follows European dialing rules. The BR-x site follows North
American dialing rules (that is, the NANP).
Note There is no common European numbering plan in effect within the European Union at the
moment and therefore, for simplicity, the simulated PSTN supports similar dialing rules like
in NANP. This will be explained in more detail in a later section.
Student pods are connected to the simulated PSTN using ISDN PRI interfaces. ISDN PRI
interfaces use one of the two switch types toward the simulated PSTN.
HQ-1 and HQ-2 use switch-type primary-oet5.
BR-1 and BR-2 use switch-type primary-ni.
Be aware that using switch-type primary-ni automatically modifies the type of number (TON)
for outbound ISDN calls if they have one of the following called-partynumber formats:
Seven-digit called-party number gets TON subscriber
10-digit called-party number gets TON national
12-digh called-party number (includes 2-digit country code) prefixed with Oil gets TON
international
This figure shows DID called-party number ranges that are used to dial into individual sites.
These calls can be placed either from a simulated PSTNphone to a site or fromone site to
another via the simulated PSTN. Pods use overlapping directory numbers at HQ-x and BR-x
sites.
>2010 Cisco Systems, Inc. I_abGuide
HQ-1
Pod1
*v.
European Country
5552XXX 5552XXX
5115552XXX 5125552XXX
55-5115552XXX 55-5125552XXX
>iC
^<
ISDN PRI p: misty-net6
Simulated PSTN
ISDN PRI pnidiy-r
HQ-2
Pod 2
V-
The individual sites use DID number ranges listed in this table. Area codes 511, 512, 521, and
522 are assigned to individual sites. Country codes 55 and 66 represent the virtual Europetin
(EU) and the virtual North American (NA) countries.
Site DID Number Ranges
HQ-mSite(EU) BR-m Site (NA)
Site internal directory numbers 2XXX 3XXX
Local PSTN DID range 555-2XXX 555-3XXX
National PSTN DID range 51m-555-2XXX 52m-555-3XXX
International PSTN DID range 55-51 m-555-2XXX 66-52m-555-3XXX
Note m is your pod number.
Simulated PSTN Phone
Each of the student pods has been using its own simulated PSTNphone: PSTNphone 1(for
pod 1) and PSTN phone 2 (for pod 2).
The simulated PSTNphone is used to iniliatc simulated PSTNcalls to sites of the pod to which
the simulated PSTNphone has been associated, or the simulated PSTN phone is used to
terminate simulated PSTN calls that originated from a student pod. For this purpose, the
simulated PSTNphone has been usingsix lines that represent different types of calls.
This table outlines which line buttons are available at the simulated PSTN phone as well as
their functions.
Troubleshooting Cisco Unified Communications (TVOICE] v8 D ) 2010 Cisco Systems, Inc
^
Lines at the Simulated PSTN Phone
Button Position Button Name* Function
1 Local Terminates valid local PSTN calls
2 National Terminates valid national PSTN calls
3 Intemtl Terminates valid international PSTN calls
4 800 Terminates valid toll-free PSTN calls
5 Premium Terminates valid premium (900) PSTN calls
6 Emergency Terminates emergency PSTN calls
*Button names as they appear on the simulated PSTN phone display (button labels)
The definition of valid calls for each type of call is explained in later sections. In these later
sections, it will also be explained how the simulated PSTN phone can be used to initiate calls of
various types.
Caution Placing calls between simulated PSTN phones of two pods is not supported.
Calls Inbound to Simulated PSTN
This section describes which calls are recognized by simulated PSTN in the inbound direction
when dialed from student pods and how these calls are routed.
Calls Inbound from HQ Sites
HQ sites are virtually located in Europe and, therefore, all calls from HQ sites into the
simulated PSTN follow European dialing rules. For simplicity however, variable-length
numbers are implemented for international calls only and not for local and national calls as is
common in many European countries.
A call from an HQ site, depending on its called number, is either terminated at the simulated
PSTN phone or the call is routed to another site (HQ or BR) of pod 1 or pod 2. The call can
also be returned back (hairpinned) to the site it came from.
This table lists all the types of valid inbound calls to the simulated PSTN from HQ sites.
Valid Inbound PSTN Calls from HQ Sites
Valid Called Number Valid TON Call Routed To Calling Number
Presentation Format
112 unknown Emergency line* preserved
[2-9JXX-XXXX (7 digits} unknown Local line* preserved
(2-9JXX-XXXX (7 digits) subscriber Local line* preserved
0-(2-9]XX-[2-9]XX-XXXX(11 digits) unknown National line* preserved
[2-9]XX-i2-9]XX-XXXX(10 digits) national National line* preserved
00-any number of digits unknown Intemtl line* preserved
Any number of digits international Intemtl line* preserved
0-800-{2-9]XX-XXXX (11 digits) unknown 800 line* preserved
800-[2-9]XX-XXXX(11 digits) national 800 line* preserved
>2010 Cisco Systems, Inc. Lab Guide
Presentation Format
0-900-[2-9]XX-XXXX(11 digits) unknown Premium line* preserved
9Q0-[2-9]XX-XXXX (11 digits) national Premium line* preserved
555-2XXX (7 digits) unknown HQ-m subscriber
555-2XXX (7 digits) subscriber HQ-m subscriber
0-511-555-2XXX(11 digits) unknown HQ-1 national
511-555-2XXX (10 digits) national HQ-1 national
00-55-511-555-2XXX (14 digits) unknown HQ-1 national
55-511-555-2XXX (12 digits) international HQ-1 national
0-512-555-2XXX (11 digits) unknown HQ-2 national
512-555-2XXX (10 digits) national HQ-2 national
00-55-512-555-2XXX (14 digits) unknown HQ-2 national
55-512-555-2XXX (12 digits) international HQ-2 national
00-66-521-555-3XXX (14 digits) unknown BR-1 international
66-521-555-3XXX (12 digits) international BR-t international
00-66-522-555-3XXX (14 digits) unknown BR-2 international
66-522-555-3XXX (12 digits) international BR-2 international
"Lines at the simulated PSTNphone that are associated with the call originating pod
Note Where m is the pod number that originated the call
Note Allcalls being routed to pod sites present the called-party number in the national format,
regardless of the call type.
Calls Inbound from BR Sites
BR sites are virtually located in North America. The simulated PSTN follows the NANP
number fonnats.
A call from a BR site, depending on its called number, is either terminated at the simulated
PSTN phone or the call is routed to anotlier site (HQ or BR) of pod 1 or pod 2. The call can
also be returned back (hairpinned) to the site from which it came.
*
This table lists all the types of valid inbound calls to the simulated PSTN from BR sites.
Valid Inbound PSTN Calls from BR Sites
Presentation Format
911 unknown Emergency line* preserved
[2-9JXX-XXXX (7 digits) unknown Local line* preserved
[2-9JXX-XXXX (7 digits) subscriber Local line* preserved
1-[2-9]XX-[2-9]XX-XXXX(11 digits) unknown National line* preserved
[2-9]XX-(2-9]XX-XXXX(10 digits) national National line* preserved
011-any number of digits unknown Intemtl line* preserved
Any number of digits international Intemtl line* preserved
1-800-{2-9]XX-XXXX (11 digits) unknown 800 line* preserved
800-[2-9]XX-XXXX(11 digits) national 800 line* preserved
1-900-[2-9]XX-XXXX (11 digits) unknown Premium line* preserved
900-[2-9]XX-XXXX(11 digits) national Premium line* preserved
555-3XXX (7 digits) unknown BR-m subscriber
555-3XXX (7 digits) subscriber BR-m subscriber
1-521-555-3XXX(11 digits) unknown BR-1 national
521-555-3XXX (10 digits) national BR-1 national
011-66-521-555-3XXX (15 digits) unknown BR-1 national
66-521-555-3XXX (12 digits) international BR-1 national
1-522-555-3XXX(11 digits) unknown BR-2 national
522-555-3XXX (10 digits) national BR-2 national
011-66-522-555-3XXX (15 digits) unknown BR-2 national
66-522-555-3XXX (12 digits) international BR-2 national
011-55-511-555-2XXX (15 digits) unknown HQ-1 international
55-511-555-2XXX (12 digits) international HQ-1 international
011 -55-512-555-2XXX (15 digits) unknown HQ-2 international
55-512-555-2XXX (12 digits) international HQ-2 international
*Lines at the simulated PSTNphone that are associated with the call originating pod
Where m is the pod number that originated the call
AHcalls being routed to pod sites present the called-party number in the national format,
regardless of the call type.
) 201C Cisco Systems, Inc
Lab Guide
Calls Outbound from Simulated PSTN
This section explains howoutbound calls fromthe simulated PSTNtoward student pods can be
placed.
Predictable Calling-Party Numbers
For placing calls to student pods, each pod is using its own simulated PSTN phone. Calling-
party numbers for the simulated PSTN phone-initiated calls are determined depending on
which simulated PSTN phone line was selected to originate the call. This figure shows calling-
party numbers and the associated TONs, based on a line selected. If a call is originated from the
800 (toll-free) line, instead of from the calling number, the calling-parly name "PSTN" is
presented (the calling-party number field is erased completely). If a call is originated from the
Premium line, the presentation of the calling-party number is restricted.
Troubleshooting Cisco Unified Coinmuni cations (TVOICE) v8 0 >201Q Cisco Systems, Inc
ling-Ha rt;
initiated from Simu
PSTN-Phone-x
555444, subscriber
606-555-4444, national
Local
National
Internt) #
800
Premium
Emergency #
77-606-555-4444, international
"PSTN"
CLIR
112or911, unknowi
To Pod x
Sites
Each line can place any of the valid supported call types. The line selection influences the
calling-partypresentation only. The call types and called-partynumber formats that are valid is
explained in later sections.
If no line is selected before dialing at the simulated PSTN phone and the called number is
dialed immediately, then a call is placed as if it was dialed fromthe Local line.
Valid Outbound Calls from Simulated PSTN Phone
The calls that are placed from the simulated PSTN phone are routed to the HQ and BR sites of
pod 1 or pod 2. Because HQ and BR sites are virtually located in different geographical
regions, an outbound call must complywiththe dialingrules in that particularregion. The
simulatedPSTNphone covers bothregionsas if it was locatedin bothEuropeand North
America at the same time.
This tablelists all the call types and their accepted called-party numbersthat are permittedto
dial fromthe simulated PSTNphone.
Valid Outbound PSTN Calls
Accepted Called Number at Simulated
PSTN Phone
Call Routed To Called Number
Presentation Format
555-2XXX (7 digits) HQ-m subscriber
555-3XXX (7 digits) BR-m subscriber
0-511-555-2XXX(11 digits) HQ-1 national
0-512-555-2XXX(11 digits) HQ-2 national
1-521-555-3XXX(11 digits) BR-1 national
1-522-555-3XXX(11 digits) BR-2 national
00-55-511-555-2XXX {14 digits) HQ-1 international
) 2D10 Cisco Systems. Inc
Lab Guide
Accepted Called Number at Simulated
PSTN Phone
Call Routed To Called Number
Presentation Format
0O-55-512-555-2XXX (14 digits) HQ-2 international
00-66-521-555-3XXX (14 digits) BR-1 international
00-66-522-555-3XXX (14 digits) BR-2 international
011-55-511-555-2XXX (15 digits) HQ-1 international
011-55-512-555-2XXX (15 digits) HQ-2 international
011 -66-521-555-3XXX (15 digits) BR-1 international
011-66-522-555-3XXX (15 digits) BR-2 International
Note Where m is the pod number that is associated with the s
originating the call.
mulated PSTN phone that is
mm
Importing Cisco Unified Communications
Manager Configuration
This section explains how to import the configuration of Cisco Unified Communications
Manager using the Bulk Administration Tool import feature. Individual lab exercises will need
to follow this procedure to set up trouble tickets before actual troubleshooting can take place.
Uploading Configuration File
Before a configuration can be imported in Cisco Unified Communications Manager
Administration, the file containing the configuration needs to be uploaded to the Cisco Unified
Communications Manager publisher. The file is a package of individual configuration files in
CSV format and the package file has a .tar extension.
If, for example, one of the next lab exercises asks to import the podl-onnet-multisite-1
configuration file, perform the following procedure to upload the file to the publisher.
Step 1 In Cisco Unified Communications Manager Administration, choose Bulk
Administration >Upload/Download Files.
Step 2 Click Add New.
- Upload thecs* Me
rita:
Select The Target *
Select Transaction Type *
|C;\TFTP\pocil-onnet-iTiulti*.te-I,Iaf
ImporVExpOrt
Import Configuration *\
P Overwrite File if it exists."
Biowie.
Step 3 Choose Import/Export fromthe Select The Target drop-down list.
Step 4 Choose Import Configuration fromthe Select Transaction Type drop-down list.
Step 5 Click Browse and locate the .tar file to import on your administrator PC.
Step 6 Check the Overwrite File if it exists check box.
Step 7 Click Save. The uploaded file should appear in the Upload/Download Files list
shortly.
Importing Configuration File
Afterthe configuration file has been uploaded, the configuration canbe imported in Cisco
Unified Communications Manager Administration.
Note For this procedure, BulkAdministration service has to be running at the publisher.
Step 8 In Cisco Unified Communications Manager Administration, choose Bulk
Administration > Import/Export > Import.
i-SelectFit
file Name*
Next]
podi-onnet-mufti-iite-l.tar
1
Lab Guide
Step 9 Choose the uploaded file to import from tire File Name drop-down list.
Step 10 Click Next. The Import Configuration page is displayed.
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Step 11 Click Select All to check all the Import Configuration check boxes. The number of
check boxes that are displayed on the page will vary, but for the proper
configuration import, all of diem must be checked.
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Step 12
Step 13
Step 14
Step 15
Step 16
Scroll to the bottom of the Import Configuration page and check the Override the
existing configuration check box.
Select the Run Immediately radio button to activate the import job instantly.
Click Submit.
The results of the submitted job can be verified on the Job Scheduler page. Choose
Bulk Administration > Job Scheduler.
Click Find. If the submitted task does nol report the status "Completed," click Find
again periodically and as many times as required until you see that the job
completed.
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Note It may take even 4 to 5 minutes until the job completes. Do not continue until it is completed
Step 17 Click the Job Id link to inspect the results.
Step 18 On the Job Results page, verify the results of the individual configuration
components in the Job Results Status column.
12 Troubleshooting Cisco UnifiedCommunicalions (TVOICE] v8.0
-!-
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Step 19 If the status shows as "Error," the component was not properly imported and it
might need your attention. There are several components that may fail to import as a
matter of normal import procedure. They are as follows:
Summary component (usually at the top of the report)
Service parameters
Enterprise parameters
Certificates
Step 20 If the Job Results page shows any component that is listed with "Error," other than
those listed in Step 19, report it to the instructor.
>201C Cisco Systems, Inc.
Lab Guide 13
Lab 2-1: Troubleshooting Gateway and Endpoint
Registration Issues
Complete this lab activity to practice what you learned in the related module.
Activity Objective
In this activity, you will be assigned trouble tickets for gateway and endpoint registration
problems (hat you need to troubleshoot and resolve. After completing this activity, you will be
able to meet these objectives:
Identify and resolve Cisco IP phone registration issues
Identify and resolve MGCP gateway registration issues
Identify and resolve H.323 and SIP gateway to Cisco Unified Communications Manager
communications issues
Required Resources
These are the resources and equipment that are required to complete this activity:
Cisco Unified Communications Manager cluster
HQ and BR routers with WAN and simulated PSTN connectivity
Three IP phones that are distributed between the sites
Simulated PSTN phone
Job Aids
These job aids are available to help you complete the lab activity.
Cisco Unified Communications Manager Sites Numbering Plan
Internal directory
number
Local HQ Site (EU)
2XXX
Local BR Site (NA)
3XXX
Trouble Ticket 1: Troubleshooting Cisco IP Phone Registration
In this trouble ticket, you will troubleshoot the problem of Cisco IP phones that are unable to
register with Cisco Unified Communications Manager.
Network Discovery
In this section, you wilt discover the lab topology and basic configuration, as well as the
expected functionalities that the Cisco Unified Communicationssystemshould offer.
Caution The initial configurations might be preloaded as a part of the lab equipment initialization
procedure. Confirm withthe instructor whether you are supposed to performthe following
procedure to prepare the first trouble ticket.
Troubleshooting CiscoUnified Communications (TVOICE) v80 2010 CiscoSystems, Inc
Perform the following steps to prepare the trouble ticket:
Step 1 Delete all the IP phones in Cisco Unified Communications Manager Administration,
if any exist.
Step 2 Download and apply the following initial configurations for this trouble ticket:
Cisco Unified Communications Manager: podx-regist, where x is your pod
number
BR router: brx-regist, where x is your pod number
HQ router: hqx-regist, where x is your pod number
Step 3 Update the IP phones in Cisco Unified Communications Manager Administration
with their MACaddresses that apply to your pod. Make sure that the IP phones
register.
This figure shows the network layout.
Lab 2-1: Trouble Ticket 1-
Cisco IP Phone Registration
Pod x Ouster
3001
Theconfiguration (onlythe important components that arelisted) andtheexpected Cisco
Unified Communicationssystemfunctionality are as follows:
Three CiscoIPphonesshouldbe registeredwithCiscoUnifiedCommunications Manager
HQ-x and BR-x gateways connect the sites to the IP WAN
DHCP servers for theIPphones that areimplemented at HQ-x andBR-x gateways
Any-to-any calls shouldbe possiblebetweenthe IP phones
Problem Definition
The following two problems were experienced. Resolve theproblems intheorder that they are
listed.
Problem 1: IPphones at HQcannot register withCiscoUnified Communications Manager.
Problem 2: IPphone at BRcannot register with Cisco Unified Communications Manager.
Lab Guide 15
Gather Facts
While testing the current functionality of your Cisco Unified Communications system, fill out
the Gather Facts table with the steps used to gather facts. To ensure that all participants have
the most effective lab experience possible, please keep your results confidential. Please use as
many different tools as possible to find, diagnose, and verify the symptoms of the problems.
Gather Facts for Problem 1
Tools Used Results and Relevant Information
List the possible causes of the problem, with the most likely cause listed first.
Possibilities to Consider for Problem 1
Possible Cause
Possible Cause
Troubleshooting Cisco UnifiedCommunications (TVOICE]v8 0 )2010 Cisco Systems, Inc
Create and Implement Action Plan
In this section, you will create and implement your action plan and observe the results.
Make a change to fix the most likely cause of the problem and verify the results. If the problem
was not fixed, continue with next possible cause. Log your activities in the table.
The problemcan be caused by multiple possible causes, not by a single one. Take this into
consideration while progressing through your action plans.
Action Plans for Problem 1
Action Plan Log Observed Results
1
2
3
4
l
2
3
4
Problem Resolved
You have resolved the problem when you attain these results:
Three Cisco IP phones have been registered successfully and phone calls can be made
between the HQ site and the BR site.
Trouble Ticket 2: Troubleshooting MGCP Gateway Registration
In this trouble ticket, you will troubleshoot the problem of an MGCP gateway that is unable to
register with Cisco Unified Communications Manager.
Network Discovery
In this section, you will discover the lab topology and basic configuration, as well as the
expected functionalities that the Cisco Unified Communications system should offer.
>201C Cisco Systems. Inc. Lab Guide 17
Pod x Cluster
2001 2002 ^ HQ"*
(MGCP Gatevây)
3001
BR-x
PSTN
The configuration(only the important components that are listed) and the expected Cisco
Unified Communications system functionality arc as follows:
Cisco IOS MGCP gateway HQ-x should be registered to Cisco Unified Communications
Manager
Problem Definition
The following problem was experienced. Resolve the problem.
The problem: The MGCP gateway at HQ cannot register with Cisco Unified Communications
Manager.
Gather Facts
Gather Facts
i
1
Troubleshooting Cisco Unified Communications (TVOICEl v8.Q >201QCi5co Systems, Inc
List the possiblecauses of the problem, withthe most likelycauselistedfirst.
Possibilities to Consider
Possible Cause
Create and implement Action Plan
Make a change to fix the most likely cause of the problemand verify the results. If the problem
consideration while progressingthrough your action plans.
Action Plans
1
2
3
4
5
6
Problem Resolved
You have resolved the problem when you attain this result:
The Cisco IOS MGCP gateway at the HQ site successfully registers with Cisco Unified
Trouble Ticket 3: Troubleshooting H.323 Gateway
Communication
In this trouble ticket, you will troubleshoot the problem of an H.323 gateway that is unable to
communicate with Cisco Unified Communications Manager.
Network Discovery
>201C Cisco Systems, Inc. Lab Guide 19
L323 G
Pod x Cluster
52x-555-3XXX
BR-x
(H.323 Gate4y) Pod xPSTN
r Phone
uu,;.i
The configuration (onlythe important components that are listed)and the expectedCisco
Unified Communications system functionality are as follows:
The IP phone at the BR site should be registered with Cisco Unified Communications
Manager.
Cisco IOS H.323 gateway BR-x that is used for inbound North American calling to BR
phone.
Inbound calling to BR phone is performed by dialing fromthe simulated PSTNphone as
either 152x5553001(where x is your pod number) or 5553001.
Outbound calling to PSTN is not configured in this trouble ticket and it should not be a
subject for troubleshooting.
Problem Definition
The problem: The inbound PSTN call to the BR phone dirough the H.323 gateway at the BR
sile does not work. The PSTN phone has dialed the correct number (152x5553001, where x is
your pod number).
Gather Facts
20
While testing the current functionality of your Cisco Unified Comimmications system, fill out
Troubleshooting Cisco Unified Communicalions (TVOICE) vB 0 >2010 Cisco Systems, Inc
Gather Facts
Possible Cause
The problem can be caused by multiple possible causes, not by a single one. Take this into
Action Plans
1
2
3
4
5
6
i 2010 Cisco Systems. Inc. Lab Guide
Problem Resolved
The inboimdPSTN call to the BR phone through the 11.323 gateway at the BRsite works
(by dialing 152x5553001 fromthe PSTNphone, where x is your pod number).
Troubleshooting Cisco Unified Communications (TVOICE) v8.0 2010 Cisco Systems Inc
Lab 2-2: Troubleshooting LDAP Integration
Issues (Optional)
Complete this lab activity topractice what you learned inthe related module.
Activity Objective
In this activity, you will be assigned trouble tickets for LDAP integration problems that you
need to troubleshoot and resolve. After completing thisactivity, youwill beable tomeet these
objectives:
Identify and resolve issues that are related toLDAP synchronization
Identify and resolve issues thatarerelated toLDAP authentication
Required Resources
These aretheresources and equipment thatarerequired tocomplete this activity:
Microsoft Active Directory server
Job Aids
Thereare nojob aids for this labactivity.
Trouble Ticket 1: Troubleshooting LDAP Integration
Inthis trouble ticket, you will troubleshoot the problem ofLDAP synchronization when four
LDAP database users fail tobesynchronized toCisco Unified Communications Manager. You
will also troubleshoot aproblem with anend user not being able tologintoCisco Unified
Communications Manager user pages.
Network Discovery
This lab uses the initial configurations of the previous trouble tickets. The previous trouble
tickets must have been successfully resolved.
) 2010 Cisco Syslems, Inc.
Lab Guide 23
Pod x Cluster
LDAP (AD)
10x1 9
The configuration (only the important components that are listed) and the expected Cisco
Unified Communications system functionality areas follows:
Microsoft Active Directory' isreachable at 10.x. 1.9, where xisyour pod number.
Four end users are preeonfigured at the Active Directory and they should be synchronized
to CiscoUnified Communications Manager.
All four end users should use the password Cisco, 123 that isconfigured inthe Active
Directory database to reach their Cisco Unified Communications Manager user web pages.
Problem Definition
The following two problems were experienced. Resolve the problems in the order that they are
listed.
Problem 1: Four end users are not synchronized from the LDAP database (Active Directory) to
CiscoUnified Communications Manager.
Problem 2: Users are not able to log in to their Cisco Unified Communications Manager user
pages (htlps./VlO.vl.l/ccmuser. where x is your pod number) with their LDAP password
Cisco, 123.
Gather Facts
24
White testing thc current functionality ofyour Cisco Unified Communications system, fill out
the (iather Fads table with the steps used to gather facts. To ensure that all participants have
the most effective labexperience possible, please keep your results confidential. Please use as
many different tools as possible tofind, diagnose, and verify the symptoms of the problems.
TrouWeshooting Cisco Unified Communications(TVOICE) v8 0
D2010 Cisco Systems, Inc
Tools Used
Results and Relevant Information
Tools Used
List the possible causes ofthe problem, with the most likely cause listed first.
Possible Cause
Possible Cause
Inthis section, you will create and implement your action plan and observe the results.
Make achange to fix the most likely cause ofthe problem and verify the results. Ifthe problem
was not fixed, continue with next possible cause. Logyour activities inthetable.
The problem can be caused by multiple possible causes, not by asingle one. Take this into
consideration while progressing throughyour actionplans.
Lab Guide
Action Plan Log
Observed Results
1
2
3
4
Action Plan Log
Observed Results
l
2
3
4
Problem Resolved
Four end users are successfully synchronized from the LDAP database toCisco Unified
Users are able to log in to their Cisco Unified Communications Manager user pages.
Troubleshooting CiscoUnified Communications (TVOICE) v8.0
>2010Gsco Systems, Inc
Lab 3-1: Troubleshooting On-Net Single-Site
Calling Issues
Activity Objective
In this activity, you will be assigned trouble tickets for on-net single-site calling problems that
you need totroubleshoot and resolve. After completing this activity, you will be able tomeet
these objectives:
Identify and resolve issues inon-net single-site deployment
Identifyandresolveissuesof call forwarding
Required Resources
These aretheresources and equipment that arerequired tocomplete this activity:
Three Cisco IP phones
Job Aids
These job aidsareavailable tohelpyoucomplete the labactivity.
Local HQ Site (EU)
Local BR Site (NA)
Internal directory
number
2XXX
3XXX
Trouble Ticket 1: Troubleshooting On-Net Single-Site
Deployment
In this troubleticket, you will troubleshoot a problemwith single-site deployment.
Network Discovery
Inthis section, you will discover thelabtopology andbasic configuration, aswell asthe
expected functionalities that theCisco Unified Communications system should offer.
Performthe following steps to prepare the trouble ticket:
Step 1 Delete all theIPphones inCisco Unified Communications Manager Administration.
Step2 Download andapply thefollowing initial configurations for thistrouble ticket:
Cisco Unified Communications Manager: podx-onnet-singlesite-1. where x is
your pod number.
Router configurations remain unchanged if theprevious trouble tickets have
been successfully resolved.
Step3 Update theIPphones inCiscoUnified Communications Manager Administration
withtheir MACaddresses that applyto your pod. Makesure that the IPphones
register.
2010CiscoSystems, Inc. LabGuide 2
This figure shows thenetwork layout.
i m*j J-l fumble i.r^l 1 -,t vihte !*/.,.mg
Pod x Cluster
.twIT'
i
2001 2002
3001
y-*
Unified Communications system functionality areasfollows:
Three IP phones that are registered to Cisco Unified Communications Manager.
Any-to-any call should bepossible between theIPphones.
If HQ Phone 1(2001) does not answer a call, the call should beautomatically forwarded to
BR Phone (3001), where it should ring
Problem Definition
listed.
Problem 1: The HQ Phone 1user hears the reorder lone (or annunciator) when calling HQ
Phone 2.
Problem 2 (optional, lime permitting): When HQ Phone 2calls HQ Phone 1and HQ Phone 1
does not answer, the call does not forward to 3001. The reorder tone or annunciator is heard
instead.
Gather Facts
the Gather Facts table with the steps used togather facts. Toensure that all participants have
the most effective labexperience possible, pleasekeepyour resultsconfidential. Pleaseuse as
many different tools as possible tofind, diagnose, and verify thesymptoms of theproblems.
Gather Facts for the Problem 1
Possibilities to Consider for the Problem 1
Possible Cause
1
2
3
4
Possible Cause
l
2
3
4
>2010 Cisco Syslems, Inc. Lab Guide
Action Plans for the Problem 1
Action Plan Log
Observed Results
1
2
3
4
1
2
3
4 |
Problem Resolved
HQ phones can call each other in bodi directions.
An unanswered call at HQ Phone 1 is successfully forwarded to 3001.
Troubleshooting Cisco Unified Communications (TVOICE) v8 0 >2010Cisco Systems, Inc
Lab 3-2: Troubleshooting On-Net Multisite Calling
Issues
Complete this labactivity to practice whatyoulearned inthe related module.
Activity Objective
In this activity, you will be assigned trouble tickets for on-net multisite calling problcms that
youneed to troubleshoot andresolve. After completing this activity, you will be able to meet
these objectives:
Identify and resolve issues in on-net multisite centralized deployment
Identify and resolve issues in on-net multisite distributed deployment with gatekeepers
Identify and resolve issues in on-net multisite distributed deployment with Cisco Unified
Border Element
Required Resources
These are the resources and equipment that are required to completethis activity:
Cisco Unified Communications Manager cluster with two sites
IIQ and BR routers with WAN connectivity
Three IP phones that are distributed between the sites
Job Aids
Local HQ Site (EU) Local BR Site (NA)
Internal directory
number
2XXX 3XXX
Site code 51 m 52m
Trouble Ticket 1: Troubleshooting Centralized Multisite
Deployment
In this trouble ticket, you will troubleshoot a problem with centralized multisite deployment.
Network Discovery
Step 1 Delete all the IP phones in Cisco Unified Communications Manager Administration.
Step 2 Download and apply the following initial configurations for this trouble tickel:
2010 Cisco Systems, Inc. Lab Guide 3
Step 3
Step 4
Cisco Unified Communications Manager: podx-onnet-niultisite-1. where x is
your pod number
BR router: brx-onnet-multisite-1, where x is your pod number
HQ router: hqx-onnet-multisite-1. where x is your pod number
Updatethe IP phones in CiscoUnifiedCommunications Manager Administration
with theirMAC addresses that apply toyourpod. Make surethatthe IPphones
register.
Remove all unassigned directory numbers usingthe Call Routing> Route Plan
Report in Cisco Unified Communications Manager Administration.
Pod x Cluster
2001 2002
HQ-x
3001
BR-x
All calls between the sites within a cluster are supposed to use G.729. Within each site,
G.711 should be used.
Site-to-site calling within a cluster should allow for two parallel G.729 calls.
Problem Definition
The following problem was experienced. Resolve Ihe problem.
The problem: HQ phones cannot call a BR phone. Make sure that two parallel G.729 calls can
be established between the sites.
Gather Facts
32
While testing the current functionality of your Cisco Unified Communications syslem. fill out
Troubleshooting Cisco Unified Communications (TVOICE) v8 0 ) 2010 Cisco Systems, Inc
mm
the most effectivelab experience possible, pleasekeep your resultsconfidential. Pleaseuse as
manydifferent tools as possibleto find, diagnose, andverifythe symptoms of the problems,
Gather Facts
Possible Cause
Action Plans
1
2
3
4
5
6
'2010 Cisco Systems, Inc Lab Guide
Problem Resolved
HQphones canreacli a BRphone andvice versa, andtwo parallel G.729calls canbe
established between the sites.
Trouble Ticket 2: Troubleshooting Distributed Multisite
Deployment with Gatekeeper
This trouble ticket troublcshoots problems of distributed multisite deployment witha
gatekeeper being used in between the clusters.
Network Discovery
In this section, you will discoverthe labtopology and basic configuration, as well as the
expected functionalities that the Cisco Unitied Communicationssystemshould offer.
77i is- trouble ticket docs not neednewinitialconfigurations, ft uses the configurationsthat were
loaded in the previoustrouble ticket. The previous troubleticket must have beensuccessfully
resolved.
lab 3-2: Yrouhio Ticket 2-Troubleshooting Distnbu
Multisite Deployment with Gatekeeper
Cluster in Pod 1
H 225 Gaieke
Can! soiled
"irunk
IPVWJ
-*<
3001
BR-1
Ouster in Pod 2
3001
BR-2
SKii i.oci*; rr,!
Tiie configuration (only the important components that are listed) and the expected Cisco
Two pods cooperate to resolve this trouble ticket.
The H.225 gatekeeper-controlled trunk has been preeonfigured between the clusters.
The HQ routers act as gatekeepers.
All calls between the two clusters, as well as individual sites within each cluster, are
supposed to use G.729. Within each site. G.71 f should be used.
Site-to-site calling within each cluster should allow for two G.729 calls.
Troubleshooting Cisco Unified Communications (TVOICE) v8 0 ) 2010 Cisco Systems, Inc
Calling between thetwoclusters should allowfor twoG.729 calls.
Clusters useoverlapping directory numbers. Sitecodes areusedto produce unique
numbers.
Callbackmust be possible. The callingnumbermust have the correct format, including Ihe
site code.
Problem Definition
The following problemwas experienced. Resolve the problem.
The problem: Pod 1phonescannot call Pod 2 phones. Makesure that two parallel G.729calls
can be established between the clusters.
Gather Facts
While testing the current functionalityof your Cisco Unified Communications system, fill out
manydifferent tools as possible to find, diagnose, andverifythe symptoms of the problcms.
Gather Facts
Possible Cause
>2010 Cisco Systems, Inc. Lab Guide
Possible Cause
10
Make a change to fixthemost likely cause of theproblem andverify theresults. If theproblem
The problemcan be caused by multiple possible causes, not by a single one. fake this into
consideration while progressingthrough your action plans.
Action Plans
1
2
3
4
5
6
7
8 !
9
10
Problem Resolved
Pod 1 phones can reach Pod 2 phones and vice versa, and two parallel G.729 calls can be
established between the two clusters.
Deployment with Cisco Unified Border Element at HQ
This trouble ticket troubleshoots a problem of distributed multisite deployment with Cisco
Unified Border Element used in between the HQ sites.
Network Discovery
In this section, you will discover ihe lab topology and basic configuration, as well as the
Perform the following steps to prepare thc trouble ticket:
Step 2 Delete the Cisco Unified Communications Manager dial plan, including all route
patterns, route lists, route groups, gateways, and trunks.
Troubleshooting Cisco Umfied Communications (TVOICE] v8.0 ) 2010 Cisco Systems, Inc
Step 3 Download and apply thefollowing initial configurations for this trouble ticket:
Cisco Unified Communications Manager: podx-onnet-multisite-2, where x is
your pod number
BR router: brx-onDet-multisite-2, where x is your pod number
]iQrouter: hqx-onnet-multisite-2, where x is yourpodnumber
Step 4 Update the IPphones inCisco Unified Communications Manager Administration
withtheir MACaddresses that applyto your pod. Makesure that the IPphones
register.
Step 5 Remove all unassigned directory numbers, if any, using theCall Routing > Route
PlanReport inCisco Unified Communications Manager Administration.
l.a!) 3-2: Trouble Ticket 3Troubleshooting Distributed
Multisite Deployment with Cisco Unified Border Element at HQ
Cluster in Pod 1
BR-1
Cluster in Pod 2
HQ-2
IPb
SIP-H323)
2001 2002
Site Code S1? j
3001
BR-2
The SIP-H.323 Cisco Unified Border Element connection is preeonfigured between the IIQ
sites of the two clusters.
Calls being routed between the HQ sites go via HQ Cisco Unified Border Elements.
From the perspective of Cisco Unified Communications Managers
in pod 1: HQ-1 is the SIP-connected Cisco Unified Border Element
In pod 2: HQ-2 is the H.323-connected Cisco Unified Border Element
All calls between the clusters, as well as individual sites within each cluster, are supposed
to use G.729. Within a site, G.711 should be used.
Clusters use overlapping directory numbers. Site codes are used to produce unique
numbers.
201C Cisco Systems, Inc. Lab Guide
Callback must be possible. The calling number must have the correct format, including the
site code.
Problem Definition
The following problem wasexperienced. Resolve the problem.
The problem: When calling from SlP-side Cisco Unified Border Element to H.323-side Cisco
Unified Border Element (pod 1dials topod 2as 512XXXX inthis problem), the call setup
fails.
Gather Facts
While testing the current functionality of your Cisco Unified Communications syslem, fill out
theGather facts table with the steps used togather facts. Toensure thatall participants have
themost effective labexperience possible, please keepyourresults confidential. Please useas
many different toolsas possible to find, diagnose, andverify thesymptoms of theproblems.
Gather Facts
fist the possible causes of the problem, with the most likely cause listed first.
Possible Cause
1he problemcan be caused by multiple possible causes, not by a single one. Take this into
Troubleshooting Cisco Unified Communications (TVOICE] v8.0 >2010 Cisco Systems, Inc
Action Plans
l
2
3
4
5
6
Problem Resolved
The HQsites of the two pods can mutually set up calls via HQ Cisco Unified Border
Elements in both directions.
Deployment with Cisco Unified Border Element at BR
This trouble ticket troubleshoots problem of distributed multisite deployment with Cisco
Unified Border Element used in between the BR sites.
Network Discovery
expectedfunctionalities that the Cisco Unified Communications systemshouldoffer.
Lah 3-2: Trouble Ticket 4Trout
Deployment with Cisco Unified Border Elemental BF
Cluster in Pod 1
$
2001 2002
3001
HQ-1
BR-1
(H.323-SIP)
IPvWJ
SIP
Ouster in Pod 2
BR-2
(SIP-SIP)
Site Code 52?
Lab Guide
The configuration (only theimportant components that arelisted) and the expected Cisco
Hie SIP-II.323 Cisco Unified Border Element comiection is preeonfigured between the BR
sites of the two clusters.
Calls being routed between the BRsites go via BRCisco Unified Border Elements.
Fromthe perspective of CiscoUnified Communications Managers
In pod 1: BR-1 is the H.323-connected Cisco Unified Border Element
In pod 2: BR-2 is Ihe SIP-connected Cisco Unified Border Elemenl
All calls between theclusters, as well as individual sites within each cluster, aresupposed
to use G.729. Within a site. G.711 should be used.
Clusters use overlapping directory numbers. Site codes areused toproduce unique
numbers.
Callback must bepossible. The calling number must have thecorrect format, including the
site code.
Problem Definition
The following problemwas experienced. Resolve the problem.
"Hie problem: When calling totheremote podBRsite(from anylocal phone, dial 52y300l,
where y is the remote pod number), the called phone rings but, when answered, the reorder tone
is heard. The issue is seen in both directions.
Gather Facts
While testingthe current functionality of your CiscoUnified Communications system, fill out
theGather Facts table withihestepsused togatherfacts. To ensure that all participants have
Gather Facts
Troubleshooting Cisco Unified Communications (TVOICE) w8.0 12010 Cisco Systems. Inc
Possible Cause
In this section, youwill createand implement your actionplan andobservethe results.
Make a change to fixthemost likely cause of theproblem andverify theresults. If the problem
Theproblem canbe caused bymultiple possible causes, not by a single one. Takethis into
Action Plans
1
2
3
4
5
6
Problem Resolved
The BRsites of the two pods can mutually set up calls via BRCisco Unified Border
Elements in both directions.
) 2010 Cisco Syslems. Inc. Lab Guide
Lab 3-3: Troubleshooting Off-Net Calling Issues
Complete this lab activity to practicewhat you learned in the relatedmodule.
Activity Objective
Inthis activity, you will beassigned trouble tickets foroff-net calling problems that youneed to
troubleshoot and resolve. After completing this activity, youwill be ableto meet these
objectives:
Identify and resolve off-net calling issues when traditional call routing is used
Required Resources
These aretheresources andequipment that arerequired to complete this activity:
IIQandBRrouters with WAN connectivity as well as connectivity loanemulated PSTN
Three IP phones that are distributed between the cluster sites
Emulated PSTN network
Emulated PSTN phone
Job Aids
Thesejob aids are available to help you complete die labactivity.
Cisco Unified Communications Manager Sites PSTN Numbering Plan
HQ Site (EU) BR Site (NA)
Internal directory
numbers
2XXX 3XXX
PSTN access code 0 g
Local DID range 555-2XXX 555-3XXX
National DID range 51m-555-2XXX 52m-555-3XXX
International DID
range
55-51m-555-2XXX 66-52U1-555-3XXX
Caution All outbound PSTNcalls must be dialedwith the correct PSTNaccess code (0 at HQ, and 9
at BR).
Valid Numbers in Simulated PSTN
Calls from HQ (EU) to PSTN
Calls from BR (NA) to PSTN
Local calls
NXX-XXXX (7 digits), TON: unknown
NXX-XXXX (7 digits), TON. subscriber
Example: 455-8000
NXX-XXXX (7 digits), TON: subscriber
Example: 455-8000
National
calls
O-NXX-NXX-XXXX, TON: unknown
(0 + 3-digtt area + 7 digits)
NXX-NXX-XXXX, TON: national
(3-digit area + 7 digits)
Example: 0-455-455-8000
1-NXX-NXX-XXXX, TON: unknown
(1 + 3-digit area + 7 digits)
Example: 1-455-455-8000
International
calls
00 + any number of digits, TON:
unknown
Any number of digits, TON:
international
Example: 00-23-455-455-8000
unknown
international
Example: 011-23-455-455-8000
Emergency
calls
112, TON: unknown
911, TON: unknown
Note N represents a digit between 2 and 9.
Trouble Ticket 1: Troubleshooting Off-Net Calling Issues at HQ
Inthis trouble ticket, you will troubleshoot a problem of off-net calling using traditional call
routing on Cisco Unified Communications Manager at theHQsite.
Network Discovery
tn this section, youwill discover thelabtopology and basic configuration, aswell asthe
Performthe following stepsto preparethe troubleticket:
Step1 Delete all theIPphones inCisco Unified Communications Manager Administration.
Delete the Cisco Unified Communications Manager dial plan, including all route
patterns, route lists, routegroups, calling partytransformation patterns, gateways,
and trunks.
Download and applythe following initial configurations for this troubleticket: .
Cisco Unified Communications Manager: podx-oflnet-1, where x is your pod
number
BR router: brx-oflnet-1, where x is your pod number
HQ router: hqx-offhet-1, where x is your pod number
Restart the Cisco CallManager service in Cisco Unified Serviceability.
Reset the MGCPgatewayHQ-xusingthe no mgcp command, followed by thc
mgcpcommand. Make surethatthe MGCP gateway HQ-x registers withtheCisco
Update theIPphones inCisco Unified Communications Manager Administration
with their MAC addresses that apply to your pod. Make sure that the IPphones
register.
Step 2
Step 3
Step 4
Steps
Step 6
Lab Guide
Step 7 Remove all unassigned directory numbers, ifany, using the Call Routing >Route
Plan Report inCisco Unified Communications Manager Administration.
This figure showsthe network layout.
Cluster in Pod 1
~L=, ^
^3
2001 2002
HQ-1
f
^3^Y
3001
BR-1
\ /
Pod 1 PSTN
Phone
i
PSTN
*?
^
Pod 2 PSTN
Phone
Bi d&i!ai>3U diiL.**=&&. iftsj
Ouster in Pod 2
i
HQ-2
2001 2002
3001
BR-2
Unified Communications systemfunctionality are as follows:
Cisco Unified Communications Manager is preeonfigured with atraditional dial plan.
HQ-1 and HQ-2 are MGCP gateways. BR-1 and BR-2 areH.323 gateways.
International callsthat areplaced manually from pods out tothe PSTN must bedialed with
a trailing # to avoid postdial delay. Received international calls should be called back from
caii lists without number editing.
PSTNaccess code 0 is dialed fromHQ sites (EU). PSTNaccess code 9 is dialed from BR
sites (North America). PSTN access codes must be dialed for all outbound PSTNcalls.
At HQ. all dialingfollows European PSTN rules. In BR, all dialingfollows North
American PSTN rules, as illustrated in the Job Aids section.
Review dialing rules from simulated PSTN phones tobe able to dial into clusters correctly.
Problem Definition
The following two problems were experienced. Resolve the problems inthe order that they are
listed.
Problem 1:All types of calls that areplaced from the HQ site (dial without pressing the
NewCall softkey button) tothe PSTN appear tothePSTN as international calls (calls ring at
the Intemtl line). All types ofcalls that are placed from the HQ site present the same calling-
party-number for any type of call (5552000).
Problem 2: When anHQ phone places an emergency call (0112) using the NewCall softkey
button, after the second digit is entered, die dial tone isheard. Also, a long postdial delay is
experienced.
Gather Facts
While testing thecurrent functionality of yourCiscoUnified Communications system, fill out
theGather Facts tablewiththe steps usedtogatherfacts. To ensure that all participants have
the most effectivelabexperience possible, please keepyour resultsconfidential. Pleaseuse as
many different toolsas possible to find, diagnose, andverifythesymptoms of theproblems.
List the possible causesof the problem, withthe most likelycause listedfirst.
Possible Cause
>2010 Cisco Systems, Inc Lab Guide 45
Possible Cause
Inthis section, youwill createand implement your actionplanand observeIheresults.
Make a change to fixthemost likely cause of theproblem andverify the results. If theproblem
Theproblem canbe caused by multiple possible causes, not bya single one. Takethis into
consideration while progressing throughyour actionplans.
1
2
3
4
5
6
1
2
3
4
5
6
Problem Resolved
Calls placed from the HQ site present the correct calling-party number.
No postdial dela; is experienced on outbound PSTN calls.
Outbound PSTN calls ring at the correct PSTNphone lines.
>2010CiscoSystems, Inc
Trouble Ticket 2: Troubleshooting Off-Net Calling Issues at BR
Inthistrouble ticket, youwill troubleshoot a problem of off-net calling usingtraditional call
routingon CiscoUnifiedCommunications Managerat the BRsite.
Network Discovery
expectedfunctionalities that the CiscoUnifiedCommunications systemshouldoffer.
This troubleticketdoes not need new initial configurations. It uses the configurations that were
loadedin the previoustroubleticket. The previoustroubleticket must have beensuccessfully
resolved.
Lab 3-3: Trouble Ticket 2Troubleshooting
Cluster in Pod 1
2001 2002
SUSSO^XXX
3001
;2i53F3,w
Pod 1 PSTN
Phone
Cluster in Pod 2
2001 2002
3001
All received calls should be dialable fromIP phone call lists (Received, Missed Calls)
without number editing.
Problem Definition
listed.
Problem 1: When an international call is dialed from a BR phone (for instance 9011-45-45?-
455-100O#), the call fails and the caller hears a reorder tone.
Problem 2: Inbound PSTN calls to a BR phone present a calling-party number that prevents
callback using call lists.
2010 Cisco Systems, Inc Lab Guide
Gather Facts
While testing thecurrent functionality of yourCisco Unified Communications system, fill out
the Gather Facts table with the steps used togather facts. Toensure that all participants have
the most effective labexperience possible, pleasekeepyour resultsconfidential. Please use as
many different tools aspossible tofind, diagnose, and verify thesymptoms of theproblems.
Tools Used
Possible Cause
Troubleshooting Cisco Unified Communications (TVOICE) v8 0 >2010 Cisco Systems, Inc
Possible Cause
Inthis section, you will create and implement your action plan and observe theresults.
Make achange tofix the most likely cause ofthe problem and verify the results. Ifthe problem
wasnot fixed, continue with next possible cause. Logyouractivities inthe table.
consideration wfiile progressing throughyour actionplans.
Action Plan Log
Observed Results
1
2
3
4
5
6
Action Plan Log
Observed Results
i
2
3
4
5
6
Problem Resolved
You have resolved the problemwhen you attain these results:
All types of outbound calls from a BRphoneto thesimulated PSTN that aresupported
should be successful.
All inbound PSTNcalls to the BR phone should present the calling-party number that can
be dialed back without editing.
) 2010 Cisco Systems, inc
Lab Guide 49
Lab 3-4: Troubleshooting Globalized Call-Routing
Issues
Complete this labactivity topractice what youlearned intherelated module.
Activity Objective
In this activity. \ouwill be assigned trouble tickets for calling problems when globalized call
routing is used. You will need to troubleshoot and resolve these problems. After completing
this activity, you will beable tomeet these objectives:
Identify and resolve off-net calling issues when globalized call routing is used
Required Resources
These are the resources and equipment that arc required to complete this activity:
Cisco Unified Communications Manager clusler
HQ and BR routers with WAN connectivity aswell asconnectivity toasimulated PSTN
Three IP phones that are distributed between the cluster sites
Simulated PSTN
Job Aids
These jobaids areavailable to help youcomplete thelabactivity.
50
HQ Site (EU)
BR Site (NA)
Internal directory
numbers
2XXX
3XXX
PSTN access code 0
g
Local DIDrange 555-2XXX
555-3XXX
National DID range 51m-555-2XXX
52m-555-3XXX
International DID
range
55-51 m-555-2XXX
66-52m-555-3XXX
Caution All outbound PSTN calls must bedialed with thecorrect PSTN accesscode (0at HQ, and 9
at BR)
TroubleshootingCisco Unified Communications(TVOICE) v8 0
Calls from HQ(EU) to PSTN Calls from BR (NA) to PSTN
Local calls NXX-XXXX (7 digits), TON: unknown
NXX-XXXX(7 digits), TON: unknown
Example: 455-8000
Example: 455-8000
National 0-NXX-NXX-XXXX, TON: unknown
calls
(0 + 3-digit area + 7 digits) (1 + 3-digit area + 7 digits)
NXX-NXX-XXXX, TON: national NXX-NXX-XXXX, TON: national
Example: 0-455-455-8000 Example: 1-455-455-8000
International
calls
unknown
unknown
international
international
Example: 00-23-455-455-8000 Example: 011-23-455-455-8000
Emergency
calls
112, TON: unknown 911, TON: unknown
Trouble Ticket 1: Troubleshooting Globalized Call-Routing
Issues
In this trouble ticket, you will troubleshoot a problem with off-net calling using globalized
routing at Cisco Unified Communications Manager.
Network Discovery
expectedfunctionalities that the CiscoUnifiedCommunications systemshouldoffer.
Step1 Deleteall the IPphones in CiscoUnifiedCommunications ManagerAdministration.
patterns, route lists, route groups (detachLRGfromHQdp andBR dp dev icepools
first), calling party transformationpatterns, gateways and trunks, and calling search
spaces and partitions.
Cisco Unified Communications Manager: podx-offnet-2, where x is your pod
number
Note The import process can take approximately 4 to 5 minutes to complete. In order to optimize
time, continue with applying router configurations and, once those are completed, return to
Cisco Unified Communications Manager Administration to verify the import results.
Step 4
i 2010 Cisco Systems. Inc.
BR router: brx-offnet-2, where x is your pod number
HQ router: hqx-offnet-2, where x is your pod number
Restart the Cisco CallManager service in Cisco Unified Serviceability.
Lab Guide
Step 5 Reset the MGCP gateway HQ-x using the nomgcp command, followed by the
mgcpcommand, Make sure that theMGCP gateway HQ-x registers with Cisco
Step6 Update the IPphones inCisco Unified Communications Manager Administration
with theirMAC addresses thatapply toyour pod. Make sure that theIPphones
register.
Step 7 Remove all unassigned directory numbers, if any, using theCall Routing > Route
PlanReport in CiscoUnified Communicalions Manager Administration.
Cluster in Pod 1
I
2001 2002
HQ-1
3001
y
BR-1
Pod 1 PSTN
Phone
i
PSTN
/
Pod 2 PSTN
Phone
Ouster in Pod 2
^
2001 2002
BR-2
3001
The configuration (onlythe important components that are listed)and theexpectedCisco
UnifiedCommunications systemfunctionality are as follows:
HQ-1 and HQ-2 are MGCP gateways. BR-1 and BR-2 are H.323 gateways.
International numbers that are placed from clusters out to the PSTN must be dialed with a
trailing #.
PSTN access code 0 is dialed from HQ sites (EU). PSTN access code 9 is dialed from BR
sites (North America).
At HQ, all dialingfollows European PSTN rules. At BR, all dialingfollows North
American PSTN rules, as illustrated in Job Aids section.
Review the dialing rules from the emulated PSTN phones to be able to dial into clusters
correctly.
Problem Definition
52
The following tiiree problems wereexperienced. Resolvethe problcms in the order that theyare
listed.
Problem 1: Outbound PSTN calling for EU long distance and international numbers do not
work from the HQ site. Other number types work correctly.
Troubleshooting Cisco Unified Communications (TVOICE) v8 0 >2010&SCO Systems, Inc
Problem 2: Inbound calling fromthe PSTN does present correct calling-party numbers at ihe
HQ site but callback is not possible.
Problem 3: Outbound PSTN calling does not work from the BR site.
Gather Facts
the Gather facts table with the steps used to gather facts. To ensure that all participants have
Lab Guide 53
Possible Cause
Possible Cause
Possible Cause
Troubleshooting Cisco Unified Communications (TVOICE) wB 0 12010 Cisco Systems, Inc
mm
1
2
3
4
5
6
l
2
3
4
5
6
l
2
3
4
5
e
Problem Resolved
HQ phones and the BR phone can reach the simulated PSTN phone and vice versa.
Lab Guide 55
Lab 4-1: Troubleshooting SAF Client and
Forwarder Issues
Activity Objective
In this activity, you will be assigned trouble tickets for SAF and CCD problems that you need
to troubleshoot and resolve. After completing this activity, you will be able to meet these
objectives:
Identify and resolve issues of the SAF forwarder and the SAF client
Identify and resolve issues of the process of learninghosted patterns in CCDdeployment
Identify and resolve issues of PSTN failover in CCD deployment
Required Resources
Two Cisco Unified Communications Manager clusters
Two sites with Cisco Unified Communications Manager Express gateways and
connectivity to an emulated PSTN
Two IP phones in each Cisco Unified Communicalions Manager clusler
A single phone for each Cisco Unified Communications Manager Express site
An MGCP gateway connecting each Cisco Unified Communications Manager cluster to the
emulated PSTN
An IP WAN interconnecting both Cisco Unified Communications Manager clusters and
both Cisco Unified Communications Manager Express sites
An emulated PSTN with two PSTN phones
Job Aids
These job aids arc available to help you complete the lab activity.
56
Internal directory
numbers
2XXX 3XXX
PSTN access code 0 9
Internationa! DID
range
55-51 m-555-2XXX 66-52m-555-3XXX
Troubleshooting Cisco Unified Communications (TVOICE) v8 0 >2010 Cisco Systems, Inc
Caution All outbound PSTN calls must be dialed with the correct PSTN access code (0 at HQ and 9
at BR).
Calls from HQ(EU) to PSTN Calls from BR (NA) to PSTN
Local calls NXX-XXXX (7 digits), TON: unknown NXX-XXXX (7 digits), TON: unknown
NXX-XXXX (7 digits), TON: subscriber NXX-XXXX (7 digits), TON: subscriber
Example: 455-8000 Example: 455-8000
National O-NXX-NXX-XXXX, TON: unknown 1-NXX-NXX-XXXX, TON: unknown
calls
(3-digit area + 7 digits) (3-digit area + 7 digits)
Example: 0-455-455-8000 Example: 1-455-455-8000
International
calls
unknown
unknown
international
international
Example: 00-23-455-455-8000 Example: 011-23-455-455-8000
Emergency
calls
Trouble Ticket 1: Troubleshooting SAF Client to SAF Forwarder
Communication
In this trouble ticket, you will troubleshoot the problem of an SAF client that is unable to
communicate with the SAF forwarder.
Network Discovery
patterns, route lists, route groups (detach LRG from HQdp and BRdp device pools
first), translation patterns, calling, and called-party transformation patterns.
gateways, and trunks.
Step 3 Download and apply the following initial configurations for this trouble licket:
Cisco Unified Communications Manager: podx-safccd, where x is your pod
number
BR router: brx-safccd, where x is your pod number
HQrouter: hqx-safccd, where x is your pod number
Step4 Restart the Cisco CallManager service in Cisco Unified Serviceability.
Lab Guide
Step5 Update the newIPphones in Cisco Unified Communications Manager
Administration with the MAC addresses that applyto your pod HQsite. Makesure
that theHQIPphones register withCisco Unified Communications Manager.
Step6 Update an IPphone MAC address in Cisco Unified Communications Manager
Express BR-x with the MAC addresses that apply to your pod BRsite. Make sure
that the IP phone registers with Cisco UnifiedCommunications Manager Express
BR-x.
Step 7 Remove all unassigned directory numbers, if any, using the Call Routing> Roule
Plan Report in Cisco Unified Communications Manager Administration.
o i >lc Ti
nt to SAF ro
Cluster in Pod x
I
External SAF Gk.-r
^ HQ-1
SAF Foavardei-
Unified CME' in Pod x
BR-1
SAF Forwarder
SAF Network
Uiified C^lE = Qsco Ijnrfied Conmun(cations Manager Express
Unified Communications system functionalily are as follows:
Cisco Unified Communications Manager acts as an external SAF client to the HQ-x SAF
forwarder.
Cisco Unified Communications Manager should be registered with the HQ-x SAF
forwarder.
BR-x is Cisco Unified Communications Manager Express that acts as the SAF forwarder
and internal SAF client. (This functionality is not used in this trouble licket.)
Problem Definition
Theproblem: The SAF forwarder at the IIQ site does not register with Cisco Unified
Gather Facts
Whiletestingthe current functionality of your CiscoUnifiedCommunications system, fill out
the most effectivelab experience possible, pleasekeep your results confidential. Pleaseuse as
Gather Facts
Possible Cause
was not fixed, conlinue with next possible cause. Log your activities in the table.
Lab Guide
Action Plans
1
2
3
4
5
6
Problem Resolved
Cisco Unified Communications Manager registers with the SAF forwarder HQ-x. "Hie
registration type should be primary.
Trouble Ticket 2: Troubleshooting Hosted Directory Number
Patterns Learning Process
In this trouble ticket, you will troubleshoot the problem of a CCD client not showing some of
the hosted directory number patterns.
Network Discovery
expected functionalities that the Cisco Unified Coimntinications system should offer.
This trouble ticket does not need to load new initial configurations; however, the previous
trouble tickets must have been successfully resolved.
Dm.:.t< HiunbC'r
Cluster in Pod 1 Ouster in Pod 2
^
n
2001 2002
s213XXX a
HQ-2
2001 2002
s IP WAN -^
SAF Enabled
Unified CME in Pod 1 Unified CME in Pod 2
3001 3001
y
BR-1 BR-2
Unified Communications system functionality are asfollows:
The Cisco Unified Communications Manager cluster registers two HQ phones.
Cisco Unified Communications Manager acts as an external SAF client to the HQ-x SAF
forwarder.
Cisco Unified Communications Manager at pod xadvertises its hosted directory number
pattern 51 xXXX, where xisthe pod number.
Cisco Unified Communications Manager Express BR-x registers the BR phone.
Cisco Unified Communications Manager Express acts as the SAF forwarder and internal
SAF client.
Cisco Unified Communications Manager Express at pod yadvertises its hosted directory
number pattem 52yXXX, where yisthe pod number.
Four sites should mutually exchange their hosted directory number patterns: 511 XXXX.
512XXXX. 521XXXX, and 522XXXX.
When the patterns are learned, calls should be possible to the advertising sites from any
phone.
Problem Definition
The following two problems were experienced. Resolve the problems in the order that the; are
listed.
Problem 1: The patterns that are hosted by Cisco Unified Communications Manager Express
are not showing up inthe Learned Pattern report in the Cisco Unified RTMT.
Problem 2: Calling toany ofthe learned patterns does not work.
Gather Facts
many different tools as possible to find, diagnose, and verify the symptoms ofthe problems.
Tools Used
>201C Cisco Systems. Inc. Lab Guide
Toots Used
List the possible causes ofthe problem, with die most likely cause listed first.
Possible Cause
1
2
3
4
5
6 j
Possible Cause
l
2
3
4
5
6
Make achange to fix the most likely cause ofthc problem and verify the results. Ifthe problem
was not fixed, continue with next possible cause. Log your activities inthe table.
62
Troubleshooting Cisco Unified Communications (TVOICEl w8.0
1
2
3
4
5
6
l
2
3
4
5
6
Problem Resolved
The Learned Pattern report in the Cisco Unified RTMT lists all the hosted directory number
patterns that are available in the network with the status of Reachable.
Calls can be successfullyplaced to any of the hosted directory number patterns that were
learned.
Trouble Ticket 3: Troubleshooting PSTN Failover
In this trouble ticket, you will troubleshoot the problem of CCD PSTN failover not working
when the learned pattern state is UnReachable.
Network Discovery
expected functionalitiesthat the Cisco Unified Communications systemshould offer.
This trouble ticket does not need to load initial configurations; however, the previous trouble
tickets must have been successfully resolved.
To trigger the CCD PSTN failover, make the learned patterns unreachable via the IP network
by breaking the connectivity to the SAF forwarder using the following procedure:
Step 1 SAF forwarders at the HQ site have the access list 101 preeonfigured. Make sure
that the access list exists in the configuration of your HQ router.
Step 2 At HQ-x, apply the access list 101 in the inbound direction to the Ethernet interface
connecting the Cisco Unified Communications Manager server as follows:
! y is your rack number and x is your pod number
) 2010 Cisco Systems, Inc. Lab Guide 63
interface FastEthernet0/0.yxl
ip access-group 101 in
Step3 Verifythat thestatus of hosteddirectory numberpatterns that were learned showsas
UnReachable.
Note It takes approximately 1 minute to show the statusof UnReachable using the Cisco Unified
RTMT.
Tliis figure shows the network layout.
...^ thk im \ t i
lOUDiC/srio
Cluster in Pod 1
i
2001 2002
1 av;,^-
Unified CME in Pod 1
3001
y
BR-1
CCD PSTN
f a\cve
PSTN
* .w-ti ~4 ' t.
Ouster in Pod 2
2001 2002
HQ-2
Unified CME in Pod 2
3001
BR-2
Theconfiguration (onlythe important components that are listed)and the expectedCisco
Unified Communicationssystem functionalityare as follows:
HQ-xacts as a PSTNgateway to Cisco Unified Communications Manager.
BR-x is Cisco Unified Communications Manager Express.
Globalized call routingis preeonfigured in CiscoUnified Communications Manager.
Call routingto the PSTN is preeonfigured in CiscoUnified Communications Manager
Fxpress.
All four sites exchange hosted directory number patterns along with their "toDID" rules.
Note There is a known defect at an internal SAF client that is implemented in Cisco Unified
Communications Manager Express. In the toDIDfield, you cannot configure international
escape character"+"
When a hosted directory number pattem is marked as UnReachable (in the IP network),
calling to the pattem should still be possible using CCD PSTN failover.
Trouble shooting Cisco Unified Communications (TVOICE) v8 2010 Cisco Systems, Inc
Problem Definition
The following problem wasexperienced. Resolve theproblem.
The problem: CCD PSTN failover does notwork for the hosted directory number patterns that
have been learned.
Gather Facts
Whiletestingthe current functionality of your CiscoUnifiedCommunications system, fill out
the Gather Factstable withthe steps usedto gatherfacts. To ensurethat all participants have
the most effectivelab experience possible, please keepyour results confidential. Pleaseuse as
manydifferent tools as possible to find, diagnose, andverify thesymptoms of theproblems.
Gather Facts
List the possiblecausesof the problem, withthe most likelycause listedfirst.
Possible Cause
>2010 Cisco Systems, Inc Lab Guide 65
Action Plans
Action Plan Log
Observed Results
1
2
3
4
5
6
Problem Resolved
CCDPSTN failover worksfor all hosteddirectory numberpatlems that havebeenlearned.
Troubleshooting Cisco Unified Communications (TVOICE) v8.0 )2010 Cisco Systems, Inc
Lab 5-1: Troubleshooting Device Mobility Issues
Complete this lab activity topractice what you learned inthe related module.
Activity Objective
In this activity, you will be assigned trouble tickets for device mobility problems that you need
totroubleshoot andresolve. After completing this activity, youwill be abletomeetthese
objectives:
Identifyand resolvethe issuesof DeviceMobility
Required Resources
These aretheresources andequipment that arerequired to complete this activity:
HQandBRrouters withWAN andsimulated PSTN connectivity
One IPphone, roamingbetweenHQandBRsites
Job Aids
Thesejob aidsareavailable tohelp youcomplete the labactivity.
Internal directory
numbers
2XXX 3XXX
International DID
range
55-51ITI-555-2XXX 66-52m-555-3XXX
Caution All outbound PSTN calls must be dialed with the correct PSTN access code (0 at HQ, and 9
at BR).
) 2010 Cisco Systems, Inc Lab Guide
Local calls
National
calls
international
calls
Emergency
calls
Calls from HQ(EU) to PSTN
Example 455-8000
O-NXX-NXX-XXXX, TON. unknown
NXX NXX-XXXX TON: national
(3-digitarea + 7 digits)
Example 0-455-455-8000
unknown
international
Example: 00-23-455-455-8000
112. TON: unknown
Example: 455-8000
1 NXX-NXX-XXXX, TON: unknown
Example: 1-455-455-8000
unknown
international
Example: 011-23-455-455-8000
911, TON: unknown
Trouble Ticket 1: Troubleshooting Device Mobility
Inthis trouble ticket, you will troubleshoot the problem of aroaming IPphone dial is unable to
place calls via a local gateway at the roamingsite.
Network Discovery
In this section, you will discoverthe labtopology and basic configuration, as well as the
expected functionalities thattheCisco Unified Communications system should offer.
Perform the following stepsto prepare the trouble ticket:
Step1 Delete all the IPphones inCisco Unified Communications Manager Administration.
Cisco Unified Communications Manager: podx- devicemob, where x is your
pod number
BR router: brx-apps. where x is your pod number
HQ router: hqx-apps. where x is your pod number
Step 3 Update HQPhone 1 in Cisco UnifiedCommunications Manager Administration
with its MAC address that applies toyourpodHQsite. Make surethat the IPphone
registers.
Step 4 Remo\ e all unassigneddirectory numbers, if any, using the Call Routing> Route
PlanReport in CiscoUnitiedCommunications Manager Administration.
Step 5 Unplug HQ Phone 1 from its VLAN and wait until Cisco Unified Communications
Manager shows "Unregistered" for the phone in Cisco Unified Communications
Manager Administration. (This takes approximately 30 seconds.)
Step 6 Plug HQ Phone 1 into BR VLAN to simulate the phone roaming to the BR site.
Troubleshooting Cisco Unified Communications (TVOICE) v8 0 >2010 Cisco Systems, Inc.
Note
After this trouble ticket, connect HQ Phone 1 back into its home site.
Pod x Cluster
$
2001
Riming V
BR-x
Pod x PSTN
Phone
The configuration (onlythe important components that are listed)and the expectedCisco
Unified Communications systemfunctionality are as follows:
HQ Phone 1 is roaming to the BR site.
Globalized call routing is preeonfigured in Cisco Unified Communications Manager,
Calling privileges arepreeonfigured inCisco Unified Communications Manager.
HQand BRgateways connectthe sitesto the IP WANand simulatedPSTN.
A simulated PSTN phone is available for testing.
Problem Definition
The following twoproblems wereexperienced. Resolvethe problems in the order that theyare
listed.
Caution Wien resolving the problems, do not add or remove route partitions from any of the calling
search spaces. Search for other causes for the problems.
Problem 1: When HQ Phone 1 roams to the BR site, the "Device in Roaming Location"
message does not show on the display of the phone. All calls that are placed route through the
home gateway HQ. instead of the local gateway BR.
Problem2: When HQ Phone 1 roams to the BR site and correctly shows the "Device in
Roaming Location" message, no calls can be placed fromthe phone. When calling from the
roaming site (BR), the user wants to use home dialing rules (European).
12010 Cisco Systems. Inc. Lab Guide 69
Gather Facts
While testing the current functionality ofyour Cisco Unified Communications system, fill out
the Gather Facts table with the steps used togather facts. To ensure that all participants have
the most effective labexperience possible, please keep your results confidential. Please use as
main different tools as possible lofind, diagnose, and verify the symptoms ofthe problems.
Tools Used
Tools Used
List the possible causesof the problem, withthe most likelycause listedfirst.
Possible Cause
Possibte Cause
mm
Inthis section, you will create and implement your action plan and observe the results.
Make achange tofix the most likely cause oftheproblem and verify theresults. Ifthe problem
was not fixed, continue with next possible cause. Log your activities inthetable.
consideration while progressing through your actionplans.
1
2
3
4
l
2
3
4
Problem Resolved
HQPhone 1roamingto the BRsite can placecalls usinghomedialingrules (European)
and the call is routed via the local BR gateway.
12010 Cisco Systems, Inc. Lab Guide
Lab 5-2: Troubleshooting Cisco Extension
Mobility Issues
Complete this labactivity to practice what you learned inthe relatedmodule.
Activity Objective
In this activity, you will be assigned trouble tickets for Cisco Extension Mobility problems that
you need to troubleshoot and resolve. After completing this activity, you will beable tomeet
these objectnes:
Identify andresolve theissues of intraclusler Cisco Extension Mobility
Required Resources
These are the resources and equipment that arerequired tocomplete this activity:
Cisco Unified Communications Managercluster
HQ and BR routers with WAN connectivity aswell asconnectivity toa simulated PSTN
Three IP phones
Simulated PSTN
Job Aids
These jobaids areavailable lo help youcomplete thelabactivity.
Cisco Unified Communications Manager Sites PSTNNumbering Plan
HQ Site (EU)
BR Site (NA)
Internal directory
numbers
2XXX 3XXX
International DID
range
55-51 m-555-2XXX 66-52m-555-3XXX
Caution All outbound PSTNcalls must be dialedwith the correct PSTNaccess code (0 at HQ, and 9
at BR)
'2010 Cisco Systems, Inc
Calls from HQ (EU) to PSTN Calls from BR (NA) to PSTN
Local calls
NXX-XXXX (7 digits). TON: unknown NXX-XXXX (7 digits), TON: unknown
NXX-XXXX (7 digits), TON. subscriber
Example: 455-8000 Example: 455-8000
National O-NXX-NXX-XXXX, TON: unknown
calls
(3-digit area + 7 digits) (3-digit area + 7 digits)
Example: 0-455-455-8000 Example: 1-455-455-8000
International
calls
unknown
unknown
international
international
Example: 00-23-455-455-8000 Example: 011-23-455-455-8000
Emergency
calls
Trouble Ticket 1: Troubleshooting Intracluster Cisco Extension
Mobility Issues
In this trouble ticket, you will troubleshoot a problem with intracluster Cisco Extension
Mobility.
Network Discovery
Inthis section, youwill discoverthe labtopology and basic configuration, as well as the
expected functionalities that the Cisco Unified Communications systemshould offer.
Step 1 Delete all the IP the phones in Cisco Unified Communications Manager
Administration.
patterns, route lists, route groups(detachLRGfromHQ dp and BR_dpdev icepools
first), calling and called-partytransformation patterns, gateways, and trunks.
Cisco Unified Communications Manager: podx-em, where x is your pod
number.
Router configurations remain unchanged.
Step 4 Restart the Cisco CallManager service in Cisco Unified Serviceability.
Step 5 Reset the MGCP gateway HQ-x using the no mgcp command, followed by the
mgcp command. Make sure that the MGCP gateway HQ-x registers with the Cisco
) 2010 Cisco Systems. Inc. Lab Guide
Step 6
Step 7
Update the IPphones inCisco Unified Communications Manager Administration
with their MAC addresses that apply toyour pod. Make sure that the IPphones
register.
Remove all unassigned directory numbers, il any, using (he Call Routing > Route
PlanReport in CiscoUnifiedCommunications ManagerAdministration.
2; lr
Cluster in Pod 1
Ouster in Pod 2
The configuration (only theimportant components that arelisted) and theexpected Cisco
Unified Communicationssystem functionality are as follows:
End user is preeonfigured: "jdoe" in pod 1, "pbrown" in pod 2. PIN 12345.
User device profile is configured for the end user.
Whenphones are logged out of CiscoExtension Mobility, they can reachonlycluster
interna] exiensions.
Globalized routing and calling privileges are preeonfigured in Cisco Unified
Problem Definition
The following two problems wereexperienced, Resolve the problems inthe order that theyare
listed.
Problem 1: Hie end user cannot log in at HQPhone 1 (2001).
Problem 2: Whenthe end user logs in to CiscoExtension Mobility at BRl phone, the PSTN is
not reachable outbound. (Placingan outbound PSTN call fromthe HQphoneis not the subject
of this trouble ticket.)
Troubleshooting Cisco Unified Communications (TVOICE] v8 0 ) 2010 Cisco Systems, Inc.
Gather Facts
While testing thecurrent functionality of yourCiscoUnified Communications system, fill out
theGather Fads table with thesteps used togather facts. Toensure thatall participants have
List thepossible causes of theproblem, with themost likely cause listed first.
Possible Cause
Lab Guide
Possible Cause
1
2
3
4
5
6
In this section, you will createand implement your actionplanand observethe results.
Make a change to fixthemost likely cause of the problem andverify theresults. If thcproblem
was not fixed, continue with next possible cause. Log your activities in the tabic.
Theproblem canbecaused by multiple possible causes, not bya single one. Takethis into
1
2
3
4
5
6
Action Plan Log
Observed Results
1
2
3
4
5
6
Problem Resolved
Youhave resolved the problem whenyou attaintheseresults:
Theenduser can log intoCisco Extension Mobility at bolh sites and. while logged in, calls
to the PSTN can be placed successfully.
76
Troubleshooting CiscoUnified Communications (TVOICE) v8.0
Lab 5-3: Troubleshooting Cisco Unified Mobility
Issues
Activity Objective
In this activity, you will be assigned trouble tickets for Cisco Unified Mobility problems that
you need to troubleshoot and resolve. After completing this activity, you will be able to meet
these objectives:
Identity and resolve Cisco Unified Mobility issues for inbound and outbound calls
Identify and resolve Cisco Unified Mobility issues when the associated remote destination
does notring forsome type of calls
Required Resources
HQ and BR routers with WAN and simulated PSTN connectivity
Three IPphones that are distributed between the sites
Job Aids
HQSite (EU)
BR Site (NA)
Internal directory
numbers
2XXX
3XXX
PSTN access code 0
9
Local DID range
555-2XXX
555-3XXX
National DIDrange
51m-555-2XXX
52m-555-3XXX
International DID
range
55-51 m-555-2XXX
66-52m-555-3XXX
Note
m is your pod number.
Caution
All outboun
at BR).
1 PSTNcalls must be dialed with th
e correct PSTN access code (0at HQ. and 9
I2010CI sco Systems. Inc.
Lab Guide
Local calls
National
calls
International
calls
Emergency
calls
Calls from HQ(EU) to PSTN
NXX-XXXX (7 digits), TON subscriber
Example; 455-8000
O-NXX-NXX-XXXX, TON: unknown
(0 + 3-digitarea + 7 digits)
Example: 0-455-455-8000
unknown
international
Example: 00-23-455-455-8000
112. TON: unknown
Note
Nrepresents a digit between 2 and9,
NXX-XXXX (7 digits), TON; subscnber
Example: 455-8000
NXX-NXX-XXXX, TON national
(3-digit area +7 digits)
Example: 1-455^55-8000
011 + any number of digits, TON;
unknown
Anynumber of digits,TON:
international
Example; 011-23-455-455-8000
911, TON: unknown
Trouble Ticket 1: Troubleshooting Inbound Calling
In this trouble ticket, vou will troubleshoot problems with inbound PSTN calling when Cisco
Unified Mobility- is used. You will also troubleshoot the problem of acall being handed over to
the mobile phone when Cisco Unified Mobility isused.
Network Discovery
expected functionalities that the Cisco Unified Communications system should ofier.
Perform the following steps toprepare ihe trouble ticket:
Delete the end user.
Download and apply the following initial configurations for this trouble ticket:
Cisco Unified Communications Manager: podx- unifiedmob, where xis your
pod number.
Reset both gateways in Cisco Unified Communications Manager Administration.
Reset the MGCP gateway HQ-x using the no mgcp command, followed by the
mgcp command. Make sure that ihe MGCP gateway HQ-x registers with the Cisco
Update the IP phones in Cisco Unified Communications Manager Administration
with their MAC addresses that apply to your pod. Make sure that the IP phones
register.
Remove all unassigned directory numbers, ifany, using the Call Routing >Route
78
Step 2
Step 3
Step 4
Step 5
Step 6
Step 7
Troubleshooting Inbound Calling
Pod x Cluster
End-User
Desk Phone
2001 2002 HQ-x
'-55-51X5552XXX
3001
BR-x
-65-52X5553XXX
Podx PSTN nd-User
Phone Vtob-'le Phone
Y
PSTN
+77-806-555-44=.
" dratw^ CaH rUting ^ bee" PreCOnfigUred in Cisc0 Unified Communications Manager
End user "jdoe" has been associated with these phones: the desk phone 3001 and the
mobile phone +77-606-555-4444.
The user mobile phone is implemented as the International button on the simulated PS IN
phone.
^r inbound internal calls from the mobile phone, the associated desk phone extension
S^^ l thC C8lled Party inSte3d f** mbile Phne PS numbcr +77-
' ^tilSQrZT ^ abl t0 h3nd Ver mintemal cal1 tl,at is >*'he associated
desk phone 3001 to the mobile phone, using the Mobility softkey button.
Problem Definition
o^.bhe^TNWhr 3v r1"1" Ca'!,S fFOm *e mbUe PW<ca,Iin fr0m the International line
on the PSTN phone) to an tnternal extension at HQ (for example, 005551x5552002 where vis
e^hoT ^ Tn,'ete PS mimbCT f the mbile Ph0ne' tead <*f * --ciated
desk phone extension 3001, is presented to the called party.
tnot wL ^^ C' mbile Phne by USing ^ Mbili^Softke* bn
Lab Guide 79
Gather Facts
man) different tools as possible to find, diagnose, and verify the symptoms of the problems.
Tools Used
Tools Used
Possible Cause
Possible Cause
80 Troubleshooting Cisco Unified Communications (TVOICE) vB 0
In this section, you will create and implement your action plan and observe the resulls.
n3. r'Tt0 ^ *e T" Iikdy CaUSC f the Pr0blem and verif* *e results. If the problem
Action PlanLog
Observed Results
1
2
3
4
Action Plan Log
Observed Results
l
2
3 -
4
Problem Resolved
Calling from the mobile phone (remote destination) to an internal extension is successful
S56TfCt fXlenS1n f'he aSSOdated d6Sk Ph0ne is Presen,ed totead <>f *e complete
rb 1N mobile phone number.
' Xr^^Ca" ^th end"USer ^ Phne Cm te ha"ded Ver l ** assîated mobile
Trouble Ticket 2: Troubleshooting Inconsistent Mobile Phone
Ringing
In this trouble ticket, you will troubleshoot the problem ofthe mobile phone ringing onlv for
some types ofinbound calls to the associated desk phone.
Network Discovery
S.eC'i0n;.y0l'wi11 d;scover the ^ apology and basic configuration, as well as the
Download and apply the following initial configurations for this trouble ticket:
^H T'f 'icke,doeS not "eed" '^/configurations. It uses the configuration that was
loaded mthe previous trouble ticket. The previous trouble ticket must have Li succe.ss,I
Lab Guide
rrd Use'
Dfi$\ Pr.o.-e
, <>Uhl'' lt"K' 1 ? - i" I**-Si*.
Pod x Cluster
I
2001 2002 HQ-x
^^ <
I 3001
/3?.\XX
BR-x
. / -:_x;
Pod x PSTN
Phone
/
^
PSTN
End Use.'
fvbbile Phone
For inbound calls to the associated desk phone, thc mobile phone must also ring (a little
delay is normal) for any type of inbound call (both internal and PSTN).
Problem Definition
The problem: The mobile phone does not ring for some calls dialing to the associatedI desk
phone from the PSTN. Test it by placing acall to the associated desk phone (152x5553001.
where xis pod number) from the Premium line at the simulated PSTN phone. If .Us working
properly, this call should ring up the International line (button flashing) at the PSTN phone that
is associated with ihe desk phone.
Gather Facts
While testing the current functionality of your Cisco Unified Communications system fill out
he most efiect.ve lab experience possible, please keep your results confidential Please use as
82
Gather Facts
Possible Cause
In this section,you will createand implement your actionplanand observethe results.
Make a change tofix themost likely cause of theproblem and verify theresults. If theproblem
was not fixed, continuewithnext possible cause. Log your activities in the table.
Theproblem can becaused bymultiple possible causes, not bya single one. Take this into
consideration while progressing through youraction plans.
Action Plans
Action Plan Log
Observed Results
1
2
3
4
5
6
Lab Guide 83
Problem Resolved
You have resolved Ihe problem when you attain this result:
All types of inbound calls to the end-user desk phone ring at the associated mobile phone as
well.
TroubleshootingCisco Unified Communicalions(TVOICE) v8.0
Lab 5-4: Troubleshooting Cisco Unified
Communications Manager Native Presence
Issues (Optional)
Complete this labactivity to practice whatyoulearned in therelated module.
Activity Objective
Inthis activity, youwillbe assigned a trouble ticketfor CiscoUnified Communications
Manager native presence problems thatyouneed to troubleshoot and resolve. After completing
this activity, you will be able to meet theseobjectives:
Identify and resolveCiscoUnifiedCommunications Managernativepresenceissues
Required Resources
HQ and BR routers with WAN connectivity
Three IPphones that are distributedbetween the sites
Job Aids
Thesejob aids are available to help you complete the lab activity.
Local HQ SKe (EU) Local BR Site (NA)
Internal directory
number
2XXX 3XXX
Trouble Ticket 1: Troubleshooting Cisco Unified
Communications Manager Native Presence Issues
In this trouble ticket you will troubleshoot problems with Cisco Unified Communications
Manager native presence, where the presence status is not exchanged as expected.
Network Discovery
Cisco Unified Communications Manager: podx-presence, where x is your pod
number.
) 2010 Cisco Systems. Inc. Lab Guide
Step 3 Update the IPphones inCisco Unified Communications Manager Administration
register.
Step 4 Remove all unassigned directory' numbers, if any, using the Call Routing > Route
PlanReport in Cisco Unified Communications ManagerAdministration.
Lab G-4: "iroubie Ticket 1Ti
Commuhiealtons Manager Native Presence Issue*
Pod x Cluster
m
2001 2002 HQ-x
Presence Status
t-
'>
V'jatchir-^ Region ships
3001
BR-x
VW\N
Unified Communications syslem functionality are as follows:
For the watcher, presence enlity relationships should be as follows:
2001 can watch the status of 2002 and 3001.
2002 can watch the status of 2001.
3001 can watch the status of 2002.
"Ihe presence status should also be shown in call lists.
Problem Definition
listed.
Problem 1: HQ Phone 1 cannot watch the status of BR Phone 3001.
Problem 2: Call lists do not display the presence status.
Gather Facts
the most etTective lab experience possible, please keep your results confidential. Please use as
Troubleshooting Cisco Unified Communicalions (TVOICE) v8.0 12010 Cisco Systems, Inc
Tools Used
List the possiblecausesof the problem, withthe most likelycause listedfirst.
Possible Cause
1
2
3
4
Possible Cause
l
2
3
4
The problem can be caused by multiple possible causes, not by a single one. Take this inio
i 2010 Cisco Systems, Inc. Lab Guide
Action Plan Log
Observed Results
Action Plan tog
Observed Results
1
2
3
4
Problem Resolved
All watchers canseethepresence status of theirpresence entities as requested.
The presence status is also shown in call lists.
Troubleshooting Cisco Unified Communications (TVOICE) vB.O 2010 Cisco Systems, Inc
Lab 6-1: Troubleshooting MOH Issues
Activity Objective
In this activity, you will be assigned trouble tickets for MOH problems that you need to
troubleshoot and resolve. After completing this activity, you will be able to meet these
objectives:
Identify and resolve issues ofMOH in amultisite deployment
Identifyand resolveissuesof multicastMOH
Required Resources
Three IPphones thataredistributed between thesites
Job Aids
Internal directory
number
Local "QSite (EU) Loca, BR Site (NA)
2XXX
3XXX
Trouble Ticket 1: Troubleshooting MOH
In this trouble ticket, you will troubleshoot problems with multicast MOH.
Network Discovery
In this seciion, you will discover the lab topology and basic configuration, as well as the
expected functionalities that the Cisco Unified Communicalions system should offer.
Perform the following steps toprepare the trouble ticket:
Cisco Unified Communications Manager: podx-moh, where xis your pod
number
BR router: brx-moh, where xisyour pod number
HQ router: hqx-moh, where xisyour pod number
Step 3 Update the IP phones in Cisco Unified Communications Manager Administration
register.
Lab Guide
Step 4 Remove all unassigned directory numbers, ifany, using the Call Routing >Route
Tilis figure showsthe network layout.
Pod x Ouster
2001
MOH Serve!
(Mul!..:as!) ^
i
2002
\ HQ-x
~.-^~
1
3001
A &-M
^y
r'
BR-x
IPWAN
The MOH server is enabled at the Cisco Unified Communications Manager cluster.
Multicast MOHshouldbe streamedto both sites.
IP phones are associated with the multicast MOH server through MRGL.
Problem Definition
lire following two problems were experienced. Resolve the problems in the order that they are
listed.
Problem 1: When an HQ phone is put on hold by another HQ caller, there is complete silence
insteadof the requested multicast MOH.
Problem 2: When the BR phone is put on hold by an HQ caller, instead ofmulticast MOH.
TOH is heard on the BR phone.
Gather Facts
While testing the current functionalitv of your Cisco Unified Communications system, till out
90
Troubleshooting Cisco Unified Communications (TVOICE) vB.O
Tools Used
Results and Reisvant Information
Tools Used
Possible Cause
Lab Guide
Possible Cause
Make achanse to fix the most likely cause of the problem and verify the results. Ifthe problem
was not fixed", continue with next possible cause. Log your activities in the table.
Action Plan Log
Observed Results
1
2
3
4
5
6
1
Action Plan Log
Observed Results
l
2
3
4
5
6
_ ..
Problem Resolved
. Multicast MOH is streamed to all three IP phones when they are put on hold.
92
Lab 6-2: Troubleshooting Transcoder Issues
Activity Objective
In this activity, you will be assigned trouble tickets for problems with Ihe transcoder that yc
need to troubleshoot and resolve. After completing this activity, you will be able to meet Ihese
objectives:
Identify and resolve registration issues ofthe hardware transcoder
Identify andresolve issues of transcoder allocation
Identify and resolve issues ofatranscoder that supports an ad hoc conference
Required Resources
Tiiree IPphones thataredistributed between thesites
Job Aids
Internal directory
number
Local HQSite (EU)
2XXX
Local BR Site (NA)
3XXX
r'Oli
Trouble Ticket 1: Troubleshooting Transcoder Registration
In this trouble ticket, you will troubleshoot problems with hardware transcoder registration.
Network Discovery
Cisco Unified Communications Manager: podx-xcode, where xis your nod
number
BR router: brx-xcode, where xisyour pod numbcr
HQ router: hqx-xcode, where xisyour pod number
Step 3 Update the IP phones in Cisco Unified Communications Manager Adminislration
register. r
Lab Guide
Step 4 Remove all unassigned directory numbers, ifany, using Ihe Call Routing >Route
Pod x Cluster
I
2001 202 \ HQ-x
Transcodc.
-^
3001
/ BR-x
IPVWN
The IIQ-x gateway should be used as ahardware transcoder.
The IIQ-x transcoder should register with Cisco Unified Communications Manager.
HQ-x is preeonfigured for transcoding between G.711 and G.729. which will be needed to
suppon conferencing.
Problem Definition
The problem: The hardware transcoder at ihe HQ site cannot register with Cisco Unified
Gather Facts
While testing the current functionality of your Cisco Unified Communications system, till out
manv different tools as possible to find, diagnose and verify the symptoms ofthe problems.
Troubleshooling Cisco Unified Communications (TVOICE) vB.O
Gather Facts
Tools Used
Possible Cause
Make achange to fix the most likely cause of the problem and verify the results. If the problem
Action Plans
Action PlanLog
Observed Results
Lab Guide 95
Problem Resolved
You have resoh ed the problem when you attain this result:
The hardware transcoder atthe HQ site has successfully registered wiih Cisco Unified
Trouble Ticket 2: Troubleshooting Transcoder Allocation
In this trouble ticket, you will troubleshoot problems with transcoder allocation.
Network Discovery
t it f 5SC Ovid MllOC
Podx Cluster
Conference
Bndge
Transcocfet
2001 2002
HQ-x
i.o Ho;. Conference \
-*r
3001
BR-x
IPV\AN
Hie conference bridge is configured in Cisco Unified Communications Manager.
All tiiree IP phones should participate in an ad hoc conference.
Thc ad hoc conference should be set up from the HQ site. The BR phone is nol permitted to
set up an ad hoc conference; it can only participate in an existing conference.
All voice traffic toand from the BR site should use the G.729 codec.
Problem Definition
1he following problem was experienced. Resolve the problem.
The problem- If the HQ Phone 1user sets up an ad hoc conference and invites the HQ Phone 2
user and the BR phone user, the BR phone drops out of the conference suddenly.
) 2010 Cisco Systens, Inc
Gather Facts
While testing the current functionality ofyour Cisco Unified Communications system, fill mil
Gather Facts
Tools Used
List the possible causes of the problem, with the most likely cause listed firet.
Possible Cause
Make achange to fix the most likely cause ofthe problem and verify the results. If the probk
>2Q10 Cisco Systems, Inc
Lab Guide
Action Plans
Action Plan Log
Observed Results
Problem Resolved
You have resolved the problem when youattain this result:
The BR phone can successfully participate in an ad hoc conference that was initiated from
the HQ site.
troubleshooting Cisco Unified Communications (TVOICEl v8.0
Lab 6-3: Troubleshooting Issues with RSVP
Agents
Activity Objective
In this activity, you will be assigned trouble tickets for intercluster calling problems that you
need to troubleshoot and resolve. After completing this activity, you will be able to meet these
objectives:
Identify and resolveissueswith intracluster RSVP
Identify and resolve issues with intercluster RSVP (SIP precondition)
Required Resources
Two Cisco Unified Communications Manager clusters
Three IP phones per cluster that are distributed between the cluster sites
Job Aids
Internal directory
number
Local HQSite (EU)
2XXX
Local BRSite (NA)
3XXX
Trouble Ticket 1: Troubleshooting Issues with Intracluster
RSVP Agents
In this trouble ticket, you will troubleshoot aproblem with intracluster RSVP agents,
Network Discovery
Cisco Unified Communications Manager: podx-rsvp, where xis your pod
number
BR router: brx-rsvp, where x is your pod number
HQ router: hqx-rsvp, where xisyour pod number
Lab Guide
Step 3
Step 4
Update the IP phones in Cisco Unified Communications Manager Administration
with their MAC addresses that apply toyour pod. Make sure that the IP phones
register.
Remove all unassigned directory numbers, ifany, using the Call Routing >Route
This shows the network layout.
Pod x Ouster
2001 2002
4L
RSVPAgeni
HQ-x
V W
2xG729 '
BR-x
RSVP Agent
tJiL^ca
IP\AWN
Tlie configuration (only the important components that are listed) and the expected Cisco
Unified Communications system functionality areasfollows:
RSVP agents arepreeonfigured at HQ and BRrouters
G.729 should be used between HQ and BR, allowing for amaximum oftwo parallel calls
Problem Definition
Tlie following two problems were experienced. Resolve the problems in the order that they are
listed.
Problem 1: The RSVP agent at BR cannot register with Cisco Unified Communications
Manager.
Problem 2: When acall is placed between tlie HQ and BR sites, Ihe call fails.
Gather Facts
While testis the current functionality of your Cisco Unified Communications system, fill out
the mos. effective lab experience possible, please keep your results confidential Please use as
Tools Used
Tools Used
Possible Cause
Possible Cause
Make achange to fix the most likely cause of the problem and verify the results. If the problem
consideration while progressing through youraction plans.
Action Plan Log
Observed Results
Action Plan Log
Observed Results
l
2
3
4
5
6
Problem Resolved
You ha\e resoKedthe problem when you attain this result:
Two parallel C3.729 calls can be placed between HQ and BR sites.
Trouble Ticket 2: Troubleshooting Issues with Intercluster
RSVP Agents
In this trouble ticket, you will troubleshoot aproblem with intercluster RSVP agents (SIP
precondition).
Network Discovery
expected functionalities that the Cisco Unified Communications system should oiler.
Download and apply ihe following initial configurations for this trouble ticket:
Ihis trouble ticket does no, need new initial configurations. It uses thc configurations that were
loaded in the previous trouble ticket. Thc previous trouble ticket must have been successfully
resolved.
Reset all trunks in Cisco Unified Communications Manager Administration to prepare for this
trouble ticket.
Lab 6-3; Trouble Ticket 2Troubleshooting
Issues with Intercluster RSVP Aqents
Cluster in Pod 1
2001 2002
S.tft Or* .^1
RSVPAgent
HQ-1
2xG729
IP WAN
Ouster in Pod 2
RSVP Agent
HQ-2
2001 2002
tt
SiteCode 51?^
J
Twopods cooperateto resolvethis troubleticket.
RSVP agents are preeonfigured at HQ-1 and HQ-2 routers.
G.729 should be used between the HQ sites, allowing for two parallel calls.
Problem Definition
The following two problems were experienced. Resolve the problems in the order that thev ar
listed.
Problem 1: When calling between the HQ sites, thecall fails.
Problem 2: When calling between the HQ sites, only asingle call is allowed. The second call
fails.
Gather Facts
While testing the current functionality of your Cisco Unified Communications system, fill o
he Gather Facts table with the steps used to gather facts. To ensure that all participants have
the most effective lab experience possible, please keep your results confidential. Please use a<
many different tools as possible to find, diagnose, and verify the symptoms of the probl
tit
-IVC
as
ems.
Lab Guide 103
Tools Used
Tools Used
Possible Cause
1
2
3
4
5
6
Possible Cause
l
2
3
4
5
6
104 Troubleshooting Cisco Unified Communications (TVOICE) v8 0
In this section, youwill createand implement your actionplan andobservethe results.
Makea changeto fix the most likely causeof the problemand verifythe results. If the problem
was not fixed, continuewithnext possible cause. Log your activities in the table.
The problem canbe causedby multiplepossiblecauses, not by a singleone. Takethis into
Action Plan Log
Observed Results
1
2
3
4
5
e
Action Plan Log
Observed Results
i
2
3
4
5
6
Problem Resolved
You have resolved the problemwhen you attain this result:
Twoparallel G.729calls canbe placedbetweenthe HQsites.
Lab Guide 105
106 Troubleshooting Cisco UnifiedCommunications (TVOICE) v8 i

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