96xxSIP ReadmeR2 6

96xx Release R2.
6 Readme
Date: June 7, 2010
Avaya one-X Deskphone SIP for 9600 Telephones Release SIP 2.6
This file is the readme for the Avaya one-X™ Deskphone SIP Edition release 2.6 for 9620, 9620C,
9620L, 9630, 9630G, 9640, 9640G, 9650, and 9650C model IP Telephones. This readme file
describes the contents of the Release 2.6 software distribution package.
Avaya one-X® Deskphone SIP 2.6 is software for the 9600 IP Telephones that supports the SIP protocol.
This release is required by and should be use ONLY with Session Manager 6.0 and
Communication Manger 6.0. This release should NOT be used with SES or any release of CM
prior to 6.0 This firmware package is recommended for the setups having Avaya Aura™ Session
Manager 6.0 and Avaya Aura™ Communication Manager 6.0 environments .
New features introduced as part of Release 2.6
Setup Environment Feature Feature description
Common Feature List Applicable for all CM/SES,CM/SM Setup
All CM/SES/SM setup Updated LLDP support Updated handling for LLDP
MED Network Policy TLVs
with Application types 1 and
2 or with a zero value
improves inter-operation with
recent releases of Cisco IOS.
All CM/SES/SM setup Support HTTP 1.1 for Avaya one-X Deskphone SIP
connection reuse shall download information
from PPM using HTTP 1.1 so
that a single connection is
used for all downloads during
the phone’s login sequence.
All CM/SES/SM setup Identification of the type of The phone can identify the
proxy type of registered proxy
server (e.g., SES/Session
Manager/Branch Session
Manager/Generic proxies
Copyright Avaya 2010 All rights reserved

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96xx Release R2.6 Readme
such as Audiocodes, Cisco

2811 ISR, I55).
All CM/SES/SM setup Added Support for third-party When used with CM 4.0+,
MWI Deskphone SIP shall properly
handle a PPM messages
allocating one or more aut-
msg-wt buttons and, when
an MWI notification is
received for other than the
primary extension, sets the
Message Waiting Indicator on
the corresponding button.
Feature List Applicable for CM 6.0 environment
Requires CM 6.0 Call pickup alerting. When used with CM6.0+,

Deskphone SIP alerts (ring or
flash) when a phone in a
provisioned call pickup group
alerts.
Requires CM 6.0 Improved Call Unpark When used with CM6.0+ and
behavior either SES5.2 or SM6.0+, a
call unpark button should
only appear on phones
provisioned on CM or SMGR.
Requires CM 6.0 Support SIP Direct Media. Avaya one-X Deskphone SIP
R2.6 shall support Direct
Media as supported by CM
6.0 and later.
Requires CM 6.0 Support SDP Capability Avaya one-X Deskphone SIP
Negotiation. shall support SDP capability
negotiation as defined by
draft-ietf-mmusic-sdp-
capability-negotiation-10.txt
and supported by CM 6.0
and later.
New and Enhanced features Applicable SM 6.0

Requires SM 6.0 Handling of a redirect When a user is moved from
response to a REGISTER request. Session Manager #1 to SM
#2, if a phone tries to
register the user on SM #1, it
responds with a redirect to
SM #2. This means that a
user will be able to login
after the home SM is
changed, but before the
information is propagated to

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System Manager.
Requires SM 6.0 Setting of voice mail The voice mail destination

destination can be set through PPM number can be set in System
set in the System Manager Manager for individual users
instead of administering it in
the settings file.
Requires SM 6.0 Improved user-friendly phone For example, a telephone

number display format number such as “5381234” in
the display portion of PAI or
Contact header might be
formatted in a more readable
fashion as “538-1234” before
it is displayed on the phone.
Requires SM 6.0 Support geo-redundant Ensures that a telephone

Session user can always receive
incoming calls even when
another call is active and the
call controller has changed
due to maintenance, network
or server failure. Without
this change, an incoming call
would be dropped if it is
received after a controller
switch happens while a user
is on another call. When the
SIPREGPROXYPOLICY
parameter is “simultaneous,”
the phone accepts an
incoming INVITE from any of
its Available Controllers,
regardless of which one is
active.
Requires SM 6.0 Support simultaneous Avaya one-X Deskphone SIP

registration with up to 3 proxies. shall simultaneously register
with the three highest
priority registrars specified
by any of the supported
provisioning mechanisms.
Requires SM 6.0 New extended call pickup An extended call

behavior pickup button will appear on
phones provisioned for
extended call pickup on CM.

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Field Issues Addressed in Release R2.6
Patch releases merged in this release

R2.502 --- When adding Contacts from the Call Log, the name fields are reversed.
R2.504 ---96XX SIP not setting DSCP on 200 OKs from UPDATEs, and Contacts Details from
the Call Log display name as First Name
R2.506 Cut-thru delay audio
Known Limitations of Release 2.6
Recommended SIP Transport Protocols
SIP Proxy / Phone Connection UDP TCP TLS

Type
SES (as primary controller ) Supported Recommended Recommended
Avaya Session Manager (as Supported Recommended Recommended

primary controller)
AudioCodes MP-series analog Supported Recommended Supported

and BRI gateways (as
secondary controller )
Cisco 2811 ISR (as secondary Supported Recommended Not supported

controller)
I55(as secondary controller) Supported Recommended Not supported
Teldat Vyda gateway Supported Recommended Supported
Table of Phone Connection Types for Survivability Configurations

‘Recommended’ means minimal latency in the detection of a Failover condition
¾ The Audiocodes SIP gateways tested for interoperability are the MP114 and MP118; the
recommended firmware revision, 5.60A.010.005, is available at the Avaya Support Site
location
http://support.avaya.com/css/appmanager/public/support/Downloads/P1341/C20090710164
61675250960
¾ The Cisco gateways tested for interoperability are the Cisco [ISR] ; the minimum firmware
revision is c2800nm-ipbasek9-mz.124-20.YA2.bin
¾ When the phone is failed over to a non-Avaya secondary controller (AudioCodes/Cisco): We
expect all the phones in the environment are operating in same transport protocol. There
may be a few call-based issues if the phones are using different transport protocols.

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SRTP
¾ The Avaya Gateway firmware version must support SDES SRTP
¾ Crypto suites must match
¾ Requires CM 4.0.1 or greater
¾ Always provision ‘None’ encryption (setting 9) as one of the encryption selections in the
settings file and on CM
NAT
¾ 96xx SIP telephones should not be provisioned behind a NAT with private network addresses
when the CM and SES switching fabric are provisioned on a different network.
Recommended Failback Policy
¾ Admin failback not recommended in R2.6 and Auto failback policy is recommended.
Known Issues of Release 2.6
Issues Observed specific to 2.6 or previous releases :
¾ If the user makes changes in the options and settings when failover has not yet been
detected, then the phone detects failover and registers with the secondary server and the
user logs out and logs in as a different user, the changes are displayed to the new user.
Workaround: This issue can be avoided by not logging in with different extension after
failover.
¾ 9620C/9620L-- As long as the SIP R2.6 big app is loaded into 9620C and 9650C phones that
already contain a SPICE 3.0 little app, there shouldn't be a problem. If the customer has an
EAP-TLS network /and OR little app is not 3.0,then the customer need to make sure that
they have the S3.0 boot file to load SIP firmware on 9620C/9620L.
¾ Sometimes Phone doesn’t retain value of group and SSON parameter while upgrading from
2.6 binary.
Workaround: Clear the phone and then reprogram it.
¾ HTTP Server a phone can retain a HTTPS file server setting set by DHCP after DHCP is later
set to “No” in CRAFT menu.
Failover Issues
¾ When the phone is in Failover and not connected to a controller, the Emergency Soft key is
displayed even though the phone is not able to connect calls.
¾ Phone will not play dial tone during its in a state of detection of Network failure condition
Workaround: Wait for 20 seconds
¾ When two phones having two different primary controllers and when one of the phone whose
controller is available on active call, tries to hold and resume while the other controller is not
available in network, it may be observed sometimes as No talk path or one way Audio
Workaround: Drop the call
¾ If phone is alerting during the state of Network failure detection for primary Controller then
the phone will be failed over to secondary after delayed time of unto three minutes after
answering the ringing call.
Workaround: Wait for failover to occur.
¾ If a phone with a BCA that is alerting, and if there is failure of its primary SM, and the user
attempts to bridge onto the BCA, then the BCA may appears as lock up.

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Workaround: log off and log back in again.

¾ If users are on active call originated on secondary SM or BSM then manual failback triggered
during the active call won’t work and another trigger needs to be sent after the calls end
causing the phones to failback to primary SM.
Workaround: Admin failback not recommended in R2.6.
¾ Simultaneously Holding the call and resuming it during the state of Network failover detection
phase and when the secondary server is not been detected (Limbo State ) will bring in the
phone in weird state and failback will not happen.
Workaround: Phone require to logout and login back.
Audio Issues
¾ Sometimes distorted noise is sent from the phone Direct IP turned to No on CM
Workaround: It’s recommended to keep Direct Media ‘ON’ in Communication Manager.
¾ No audio path with G.726 codec on non shuffled calls
Workaround: Recommendation is to have shuffled calls in case of G.726 codec.
¾ When SRTP CAPNEG is enabled on CM and phone and if alerting call is answered very quickly
and this happens 3-4 times then there is no more voice path on the phone.
Workaround: Restart the phone to resolve the issue.
UI and application issues

¾ When AutoDial is configured on CM, the “Edit” softkey may not be displayed for the Busy
Indicator feature immediately after editing the label for the AutoDial feature
Workaround: Navigate out of the A-menu and then return – the “Edit” softkey will then be
displayed
¾ Long labels on the SBM button module may be truncated in the middle of a displayed
character
Workaround: shorten button labels on the SBM
¾ Rotating Web displays used as screen saver can cause the phone to hang over long time
periods.
Workaround: Do not use timed, rotating web displays. If phones are found hung as a
result of rotating web displays, reboot them.
¾ If the web screen saver (setting WMLIDLEURI) uses the same image file as the custom logo,
the Web screen saver page is not displayed properly.
Workaround: Do not use the customized logo image on the WMLIDLEURL page.
¾ Not able to retrieve 250 contacts from PPM since PPM doesn't send gCL response.
Workaround: Session Manager Recommendation is to save only 50 contacts in contact
list.
¾ Incorrect character display inside Calendar meeting requests for Hebrew and Arabic
languages.
¾ Phone is displaying long form number in redial list even if user dials the short form number.
• Workaround: Dial with short form number if dialing with Long form is not allowed.
¾ If the contact Nickname been added and modified, then this will not be get reflected on the
phone, and it will again get reflected as administered on the phone.
¾ Formatted Avaya long form number is displayed instead of Avaya Extension short form on
SBM24 and aux buttons on 9650
¾ When the phone is unlocked during an active or alerting call then user cannot invoke any
applications.
Workaround: Drop the call and now the applications can be invoked properly.

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Features Related Issues
¾ Assigning number to unlabeled autodial configured feature gives error beep and not allowed
message on the top line and hence not able to assign the number when phone is registered
with Session manager.
¾ Call park feature invoked for active call on Bridged Call Appearance is not working.
¾ Auto Call Back feature invoked by holding the active call received on bridged call appearance
does not work.
¾ Audix-Record is not displayed on aux button in 9650/9650C phones.
Workaround: Use the configured for Audix Record.
¾ When the phone is unlocked during an active or alerting call then user cannot invoke any
applications.
Workaround: Drop the call and now the applications can be invoked properly.
¾ If any user is logged in with enterprise canonical long form number for which there is short
form mapping on CM, and if the user has invoked (enabled) features such as EC500, Send-
calls, will show deactivated after logout/login.
Workaround: If the user invokes the feature twice, then the feature indication status is
available on phone .This problem is not been observed when the user is log in with short
form.
Configuration Issues
¾ Any configuration parameter that is set through the CRAFT menu to its default value will be
ignored and the value set via DHCP or 46xxsettings file will be used.
Workaround: To explicitly set default values for parameters, set them in DHCP or the
46xxsettings file.
¾ Public Numbering Call Appearance will not activated at principal if Bridge Appearance Button
was pushed on NotPrincipal station
Workaround—Configuration of the Avaya Handles with Avaya SIP type recommended on
System Manager

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Feature Compatibility Matrix for Communication Manager (CM) and SIP

Enablement Server (SES)
A number of the new features in Release 2.6 depend on recent CM and SES versions as shown in the
table below.
Minimum CM Minimum SES Version

Release 2.6 Feature Version Required Required
Compatibility Matrix
SBM-24 button module – 5.0 5.1

Feature button assignments and
operation
SBM-24 button module – 5.0 5.1.2

Autodial assignments and
operation
SBM-24 button module – 5.1.2 5.1

Bridged Call Appearance (BCA)
and Call Appearance (CA)
operation
Auto Answer 5.0 5.0
Deactivation of Call Forward On 5.2 5.2

Busy Do Not Answer
Features from prior SIP phone releases that have dependencies on CM and SES versions are shown
in the table below:
Minimum CM Minimum SES Version

Prior SIP Release Feature Version Required Required
Compatibility Matrix
Visiting User (Applicable for SES 5.0 5.0

environment only)
Extend Call 5.0 5.0
SRTP 4.0.1 n/a
All features other than those 4.0 4.0

noted in this these tables
Basic steps for setting up 96xx SIP telephones:
The basic steps for setting up 96xx SIP telephones:

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Define an HTTP Server to host the telephone firmware downloads and settings files
Define a list of SIP Proxies
Define SIPDOMAIN
Define a Time Server (SNTP protocol) and parameters
Provision CM/SES for 96xx telephones (For CM 5.2 and later provision as
9620SIP,9630SIP,9640SIP,9600SIP set type , 4620SIP and for CM 5.1 and prior releases
provision as -9600SIP set types )
Optionally, if DHCP is used in the network, appropriate DHCP scope needs to be provisioned on
the phone.
Recommended configuration is only long form should be configured on system manager and user
should register with only the long form.
The following changes must be done for R2.5 and later for enabling PUSH functionality on the
phone.
¾ The string representing the Trusted server list, specified in TPSLIST, should contain a server
specification that completely matches the server specified in the push URI.
(e.g.: if the push URI is: http://192.168.12.63:80/display.wml then the TPSLIST should
contain 192.168.12.63:80 that is it includes the port number).
Setting up Primary and Secondary Controllers
The minimal settings required for provisioning SIP Proxies are:

SIPDOMAIN: The domain name that the phones operate in. All proxies are assumed to support
the same domain.
SIP_CONTROLLER_LIST: This parameter consolidates SIP controller parameters for IP
address, port, and transport protocol into a single configuration parameter. The parameter
setting should be a list of controller information where the format for each controller entry is
“host[:port][;transport=xxx]”
Example: SET SIP_CONTROLLER_LIST proxy1:5060;transport=tcp,proxy2:5060;transport=tcp

SIP_CONTROLLER_LIST (setting through Windows DHCP server):
In DHCP scope option set as
SIP_CONTROLLER_LIST="Proxy1:<port>;transport=xxx,Proxy2:<port>;transport=yyy"
Where: Proxy1 is the primary SES server address and Proxy2 is the secondary SIP server
address. And xxx and yyy are the transport protocols of respective server.
OR:
SIP_CONTROLLER_LIST="Proxy1,Proxy2:<port>;transport=xxx”
In the later case, the first controller will take default port (5061) and transport type (TLS) and
second controller will take assigned values in the SIP_CONTROLLER_LIST.
SIP_CONTROLLER_LIST (setting through Linux DHCP server (httpd.conf)

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In below examples Proxy1 is the primary SES server address and Proxy2 is the secondary SIP
server address. And xxx and yyy are the transport protocols of respective server.
For Linux,we would have to use '\' as an escape character for quoting the value inside the already
quoted name = value pair.
eg. When using single SIP CONTROLLER
"HTTPSRVR=aaa.aaa.aaa.aaa,SIP_CONTROLLER_LIST=\"xxx.xxx.xxx.xxx:5060;transport=t
cp\"";
When using Multiple SIP CONTROLLERs , please follow below examples
"HTTPSRVR=aaa.aaa.aaa.aaa,SIP_CONTROLLER_LIST=\"xxx.xxx.xxx.xxx:5060;transport=t
cp,yyy.yyy.yyy.yyy:5060;transport=tcp\"";
o Note: When viewing the SIP proxy list from the Craft menu, it may contain additional
proxies obtained from other data sources which cannot be deleted through the Craft
menu. Only proxies entered through the Craft menu can be deleted. Further settings
are optional:
CONFIG_SERVER_SECURE_MODE When the phone is using HTTPS to communicate with
PPM, it reuses the connection per HTTP 1.1.Specifies the communication mode used to access
the configuration server. This parameter applies only to 96xx model phones.
0 for use HTTP
1 for use HTTPS(default)
2 for use HTTPS if SIP transport mode is TLS; otherwise, use HTTP
Note : Default value is 1 in SIP 2.6 and 0 in SIP 2.5 release.
DISCOVER_AVAYA_ENVIRONMENT: Indicates how server support for Avaya SIP Telephony (AST)
capabilities will be determined. The default setting is "1" (AUTO), which means that the
support for AST will be determined automatically. Setting DISCOVER_AVAYA_ENVIRONMENT
to "0" (NO) means that the SIP proxy server does not support AST.
FAILBACK_POLICY: This is a string 'admin' or 'auto'. When set to 'auto' (the default), if the
phones detect that a disabled primary server has recovered, they will attempt to register to
the highest priority available controller in the SIP proxy server list, and will stop using the
Secondary. If set to ‘admin’, the phones will never autonomously attempt to leave a working
server. Admin failback needs to be invoked from the system manager for that particular
user. Please leave this parameter to its default value i.e. ‘Auto’
RDS_INITIAL_RETRY_TIME: Parameter that indicates the initial delay for a retry for
connecting to the PPM server. The default value is 2 seconds and provisioning is through the
settings file and can be set in the range of 2 to 60 seconds.
RDS_MAX_RETRY_TIME: Parameter that indicates the maximum delay interval before giving up
on connecting to the PPM server. The default value is 600 seconds and provisioning is
through the settings file and can be set in the range of 2 to 3600 seconds.
RDS_INITIAL_RETRY_ATTEMPTS: Parameter that indicates how many times PPM adaptor
should try to download from PPM before giving up on connecting to the PPM server. The
default value is 15 attempts and provisioning is through the settings file and can be set in the
range of 1 to 30 attempts.
SIPREGPROXYPOLICY: This is a string, 'alternate' or 'simultaneous'. If set to 'alternate' (the
default), the phones will only register to one server at a time. If set to 'simultaneous', the

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phones will maintain an active registration on all provisioned servers which is only supported
for SM environment.
o Recommendation is alternate for SES environment or SM-nonAST. For SM
environment recommendation is simultaneous.
Note: From 2.6 onwards the default value if CONFIG_SERVER_SECURE_MODE is 1 indicating
that PPM download will happen over HTTPS.
100REL_SUPPORT: If we set 100REL_SUPPORT 0 then phone doesn't 100rel in Supported

header and so in case of any 18x response phone doesn't respond with PRACK. So this
disables the reliable response capability on the phone. When we set it to default value as 1,
then phone sends 100rel in Supported header and responds any 18x response with PRACK.
o Recommendation is alternate for SES environment or SM-nonAST. For SM
environment recommendation is simultaneous
ASTCONFIRMATION: Sets the time that the phone waits to validate an active subscription when
it subscribes to the "avaya-cm-feature-status" package. The range is from 16-3600 seconds.
The default value for ASTCONFIRMATION is 32 seconds for 2.6 and above.
SIMULTANEOUS_REGISTRATIONS: The number of Session Managers in the configuration that
the phone will simultaneously register with. The range is from 1-3. The default value for
SIMULTANEOUS_REGISTRATIONS 3 for 2.6 and above.
ENFORCE_SIPS_URI: Controls the enforcement of SIPS URI with SRTP. The range is from 0-1.
The default value for ENFORCE SIPS URI is 1 for 2.6 and above releases.
SDPCAPNEG: Controls the SDP capability negotiation. The range is from 0-1. The default
value for this SDP CAP NEG is 1 for 2.6 and 0 for 2.5 releases respectively.## SET
SDPCAPNEG 1
Invalid Proxy Settings
If an invalid proxy server address is programmed on the phone (via the administration screen,
accessed through Craft procedures, or via the settings file) and a login is attempted and cancelled,
rebooting the phone is not sufficient to recover as the phone will continue to attempt to contact the
invalid proxy.
Procedure for erasing an invalid proxy server address: During the re-boot sequence, press the
“Program” soft key to present the CRAFT menu, select the CLEAR option to clear the phone values
and reboot the phone. The valid proxy values can be re-programmed by adding them to the settings
file before the reboot, or through phone programming from the CRAFT menu during the boot.
Specification of SEND_DTMF_TYPE Setting
The 46xxsettings.txt file supports the inclusion of one setting, SEND_DTMF_TYPE, that characterizes
the endpoint’s DTMF treatment, with respect to DTMF being sent in-band or out-of-band. The official
documentation for this setting is as follows:

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## DTMF Transmission Method

## Specifies whether DTMF tones are sent in-band, as
## Regular audio, or out-of-band, using RFC 2833
## procedures.
## 1 for in-band
Please note that, no matter what setting is provisioned for SEND_DTMF_TYPE2 (1 or 2), the
endpoint will treat DTMF as though 2 were specified for this setting. The endpoint will always send
DTMF out-of-band, following the scheme specified in IETF RFC 2833.
Specification of SYMMETRIC_RTP Setting
Beginning with 96xx SIP Release 2.0, the 46xxsettings.txt file supports the inclusion of one setting,
SYMMETRIC_RTP, which characterizes the endpoint’s treatment of RTP port security. By default, the
SIP 96xx endpoints enforce the restriction that ports be symmetric. This new setting allows the
relaxation of the port restriction so that the ports on each end of a session can be asymmetric.
Please note that this relaxation is only true for the port in the RTP stream. The IP address must be
the same no matter which setting is chosen for the port value (symmetric or asymmetric).
The variable is named SYMMETRIC_RTP. It can take on two values: 0 and 1.
If this setting is not included in a setting file, it behaves as though it were included with a value of 1,
as in:
SET SYMMETRIC_RTP “1”
This is the default behavior (which pre-dates 96xx SIP Release). The endpoint will enforce the
symmetric restrictions on RTP streams when the SYMMETRIC_RTP is set to 1, or not included in the
setting file at all.
If this setting is included in the setting file with a value of 0, as in:
SET SYMMETRIC_RTP “0”
The endpoint will relax the restrictions on the RTP streams so that they can terminate on differently-
numbered ports on each end of an RTP session.
Please note that this setting has no influence on the IP address on either end of the RTP stream.
That still cannot change after the establishment of the session.
Communication Manager (CM) Compatibility and Encryption
SRTP is not enabled in the endpoints, out of the box. As noted earlier, SRTP is supported, but
requires that the gateway software versions also support SRTP (not all do). Once this has been
verified SRTP may be provisioned in the following manner.
For SRTP operation, it is recommended that the CM must be provisioned with the SRTP IP codec
policy of:
1-srtp-aescm128-hmac80
9-none

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On the 96xx SIP phone, this can be accomplished with the inclusion of the following setting in the
46xxsettings.txt file:
SET MEDIAENCRYPTION "1,9"
When the ENFORCE_SIPS_URI parameter is 1, the phone accepts and uses only SIPS URI for
incoming and outgoing calls with SRTP media encryption. When the ENFORCE_SIPS_URI paramter is
0, the phone allows either SIP URI or SIPS URI for incoming SRTP media encryption calls and uses
only SIP URI for outgoing SRTP media encryption calls. This parameter has been introduced in 2.6.
In general, when any encryption is selected, it is necessary to also include ‘9-none’ as a fallback. For
more details, see Administering Avaya Aura™ Communication Manager Administering Avaya Aura™
Communication Manager (Doc. ID 03-300509) at
http://support.avaya.com/css/P8/documents/100059292
Note that, having different Media Encryption settings on different sets requires the union of all the
encryption settings in CM. For example, if, in one phone the following media encryption setting is
provisioned:
And, in another phone, the following media encryption setting is provisioned:
Then, CM setting must include all unique media encryption values – in this case 1, 2, and 9, or just 9
(if no media encryption is desired).
96xx SIP Release 2.6 Package Contents
The package contains the readme file and all the files necessary to upgrade Avaya 9600 IP telephones to 96xx SIP
Release 2.6. The following files are included in each package
hb96xxua3_00.bin – The 9600 R2.6 SIP boot application
SIP96xx_2_6_0_0.bin – The 9600 R2.6 SIP phone application
96xxupgrade.txt – This file is downloaded by the Avaya 9600 IP telephones and instructs
the telephones on how to upgrade. DO NOT EDIT this file. You MUST USE the 96xxupgrade.txt
file included in this package to upgrade to R2.6 SIP.
¾ Note that this file is designed for environments in which all 9620,9620C,9620L
9630, 9630G, 9640, and 9640G endpoints will be upgraded to the 9600 R2.6 SIP
phone application.

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¾ Alternative 96xxupgrade file for mixed H.323 and SIP 96xx deployments - Available
at support.avaya.com the alternate_96xxupgrade.txt file is designed for environments
that will support 96xx upgrades for both the H323 and SIP modes of operation. See the
Avaya support website for additional information for how to edit this file and setup the SIG
setting (the Signaling Procedure) in the 96xx phones.
release.xml – An XML-format text file that is designed to be used with Avaya
Integrated Management (AIM) through the Software Update Manager (SUM).
Fifteen predefined language files:
¾ Mlf_Arabic.xml
¾ Mlf_BrazilianPortuguese.xml
¾ Mlf_CanadianFrench.xml
¾ Mlf_CastilianSpanish.xml
¾ Mlf_Chinese.xml
¾ Mlf_Dutch.xml
¾ Mlf_English.xml
¾ Mlf_German.xml
¾ Mlf_Hebrew.xml
¾ Mlf_Italian.xml
¾ Mlf_Japanese.xml
¾ Mlf_Korean.xml
¾ Mlf_LatinAmericanSpanish.xml
¾ Mlf_ParisianFrench.xml
¾ Mlf_Russian.xml
Eight extended Korean ring tone files:

¾ KoreanRT1.xml
¾ KoreanRT2.xml
¾ KoreanRT3.xml
¾ KoreanRT4.xml
¾ KoreanRT5.xml
¾ KoreanRT6.xml
¾ KoreanRT7.xml
¾ KoreanRT8.xml
Four certificate files:

¾ av_csca_pem_2032.txt Avaya Call Server CA certificate with an expiration date of 2032
¾ av_prca_pem_2033.txt Avaya Product Root CA certificate with an expiration date of 2033
¾ av_sipca_pem_2011.txt Avaya SIP Root CA certificate with an expiration date of 2011
¾ av_sipca_pem_2027.txt Avaya SIP Root CA certificate with an expiration date of 2027
System specific parameters should be entered into the 46xxsettings.txt file, which is available for
separate download at support.avaya.com.
To upgrade your Avaya 9600 IP Telephones to 96xx SIP Release 2.6:
Unzip executable file in the root directory of the outbound file directory of your HTTP server.
Make any adjustments required by your environment to your 46xxsettings.txt file.

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One provisioning example may be helpful. Assumptions:
96xx SIP telephone IP address: 10.10.10.20

SIP telephone network mask: 255.255.255.0
SIP telephone gateway: 10.10.10.1
HTTP file server: 10.10.20.200
SES IP address (primary controller): 10.10.20.192
o TLS transport protocol, default so not explicitly specified in setting
AudioCodes/Cisco IP address(secondary controller ): 10.10.20.193
o TCP transport protocol, explicitly set in setting file
SNTP Server IP address: 10.10.30.147
The SIP domain: mynetwork.com
Time zone where IP telephone is provisioned: Eastern Daylight Savings Time (following the
current rules where Daylight Savings Time begins on the first Sunday of April and ends on
the last Sunday of October – for more information on provisioning DST see the 4600
Series IP Telephone LAN Administration Guide)
The following are a minimum group of settings for the 46xxsettings.txt file to make the 96xx
telephone operational:
SET SNTPSRVR 10.10.30.147

SET GMTOFFST -5:00
SET DSTOFFSET 1
SET DSTSTART 1SunApr2L
SET DSTSTOP LSunOct2L
SET SIP_CONTROLLER_LIST
10.10.20.192:5061;transport=tls,10.10.20.193:5060;transport=tcp
SET SIP_MODE 0
SET SIPDOMAIN mynetwork.com
If the 96xx SIP telephone is not acquiring its address courtesy of a DHCP server, it must be
provisioned on the phone, itself, through the use of the phone administration menu. Once in the
“Admin Procedures” menu, the “ADDR” submenu will allow the administrator to provision the
following fields:
Phone: 10.10.10.20
Router: 10.10.10.1
Mask: 255.255.255.0
HTTP File Server: 10.10.20.200
Reset your Avaya 9600 IP Telephones.

Page 15 of 22
License Agreements
T9 Text Input and other products are covered by one or more of the following patents: U.S. Pat. Nos.
5,187,480,5,818,437, 5,945,928, 5,953,541, 6,011,554, 6,286,064, 6,307,548, 6,307,549, and
6,636,162,6,646,573, 6,970,599; Australia Pat. Nos. 727539, 746674, 747901; Austria Pat. Nos.AT225534,
AT221222; Brazil P.I. No. 9609807-4; Canada Pat. Nos. 1,331,057, 2,227,904,2,278,549, 2,302,595; Japan Pat.
Nos. 3532780, 3492981; United Kingdom Pat. No.2238414B; Hong Kong Standard Pat. No. HK1010924; Republic
of Singapore Pat. Nos.51383, 66959, 71979; European Pat. Nos. 1 010 057 (98903671.0), 1 018 069
(98950708.2);Republic of Korea Pat. Nos. KR201211B1, KR226206B1, 402252; People's Republic of China Pat.
No. ZL96196739.0; Mexico Pat. Nos. 208141, 216023, 218409; RussianFederation Pat. Nos. 2206118, 2214620,
2221268; and additional patent applications are pending.
This program includes Fribidi software to which Avaya is licensed under a commercial license from Behdad
Esfahbod and Peter Grobgeld
GoAhead Server
OpenSSL 0.9.7g
Copyright (c) 1998-2008 The OpenSSL Project.
Copyright (C) 1995-1998 Eric Young (eay@cryptsoft.com), and
Copyright (C) 1995-1998 Tim J. Hudson (tjh@cryptsoft.com)
All rights reserved
Copyright (c) 1998-2007 The OpenSSL Project. All rights reserved.

Redistribution and use in source and binary forms, with or without modification, are permitted provided that the
following conditions are met:
1. Redistributions of source code must retain the above copyright notice, this list of conditions and the following
disclaimer.
2. Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following
disclaimer in the documentation and/or other materials provided with the distribution.
3. All advertising materials mentioning features or use of this software must display the following acknowledgment:
"This product includes software developed by the OpenSSL Project for use in the OpenSSL Toolkit.
(http://www.openssl.org/)"
4. The names "OpenSSL Toolkit" and "OpenSSL Project" must not be used to endorse or promote products derived
from this software without prior written permission. For written permission, please contact openssl-
core@openssl.org.
5. Products derived from this software may not be called "OpenSSL" nor may "OpenSSL" appear in their names
without prior written permission of the OpenSSL Project.
6. Redistributions of any form whatsoever must retain the following acknowledgment:
"This product includes software developed by the OpenSSL Project for use in the OpenSSL Toolkit
(http://www.openssl.org/)"
THIS SOFTWARE IS PROVIDED BY THE OpenSSL PROJECT ``AS IS'' AND ANY EXPRESSED OR IMPLIED
WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND
Page 16 of 22
FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE OpenSSL PROJECT
OR ITS CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE
GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE,
EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
This product includes cryptographic software written by Eric Young (eay@cryptsoft.com). This product includes
software written by Tim Hudson (tjh@cryptsoft.com).
Original SSLeay License
-----------------------
Copyright (C) 1995-1998 Eric Young (eay@cryptsoft.com) All rights reserved.
This package is an SSL implementation written by Eric Young (eay@cryptsoft.com).

The implementation was written so as to conform with Netscapes SSL.
This library is free for commercial and non-commercial use as long as the following conditions are aheared to. The
following conditions apply to all code found in this distribution, be it the RC4, RSA, lhash, DES, etc., code; not just
the SSL code. The SSL documentation included with this distribution is covered by the same copyright terms
except that the holder is Tim Hudson (tjh@cryptsoft.com).
Copyright remains Eric Young's, and as such any Copyright notices in the code are not to be removed.
If this package is used in a product, Eric Young should be given attribution as the author of the parts of the library
used.
This can be in the form of a textual message at program startup or in documentation (online or textual) provided
with the package.
1. Redistributions of source code must retain the copyright otice, this list of conditions and the following disclaimer.
2. Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following
disclaimer in the documentation and/or other materials provided with the distribution.
3. All advertising materials mentioning features or use of this software must display the following acknowledgement:
"This product includes cryptographic software written by Eric Young (eay@cryptsoft.com)"
The word 'cryptographic' can be left out if the rouines from the library being used are not cryptographic related :-).
4. If you include any Windows specific code (or a derivative thereof) from the apps directory (application code) you
must include an acknowledgement:
"This product includes software written by Tim Hudson (tjh@cryptsoft.com)"
THIS SOFTWARE IS PROVIDED BY ERIC YOUNG ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES,
INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE
LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
POSSIBILITY OF SUCH DAMAGE.

Page 17 of 22
The licence and distribution terms for any publically available version or derivative of this code cannot be changed.
i.e. this code cannot simply be copied and put under another distribution licence [including the GNU Public
Licence.]
libcurl 7.15.1
Copyright (c) 1996 - 2009, Daniel Stenberg, <daniel@haxx.se>.
All rights reserved.
The MIT License Copyright (c) <year> <copyright holders> Permission is hereby granted, free of charge, to any
person obtaining a copy of this software and associated documentation files (the \"Software\"), to deal in the
Software without restriction, including without limitation the rights to use, copy, modify, merge, publish, distribute,
sublicense, and/or sell copies of the Software, and to permit persons to whom the Software is furnished to do so,
subject to the following conditions: The above copyright notice and this permission notice shall be included in all
copies or substantial portions of the Software. THE SOFTWARE IS PROVIDED \"AS IS\", WITHOUT WARRANTY
OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT
SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR
IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
Expat 2.0
Copyright (c) 1998, 1999, 2000 Thai Open Source Software Center Ltd and Clark Cooper
Copyright (c) 2001, 2002, 2003, 2004, 2005, 2006 Expat maintainers.
Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated
documentation files (the "Software"), to deal in the Software without restriction, including without limitation the rights
to use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the Software, and to permit
persons to whom the Software is furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in all copies or substantial portions of the
Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,EXPRESS OR IMPLIED,
INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A
PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE
OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
MD5 Message-Digest Algorithm (RFC 1321)

Copyright (C) 1991-2, RSA Data Security, Inc. Created 1991. All rights reserved.
License to copy and use this software is granted provided that it is identified as the "RSA Data Security, Inc. MD5
Message-Digest Algorithm" in all material mentioning or referencing this software or this function.

Page 18 of 22
License is also granted to make and use derivative works provided that such works are identified as "derived from
the RSA Data Security, Inc. MD5 Message-Digest Algorithm" in all material mentioning or referencing the derived
work.
RSA Data Security, Inc. makes no representations concerning either the merchantability of this software or the
suitability of this software for any particular purpose. It is provided "as is" without express or implied warranty of any
kind.
These notices must be retained in any copies of any part of this documentation and/or software.
ConvertUTF
Copyright 2001-2004 Unicode, Inc.
Disclaimer
This source code is provided as is by Unicode, Inc. No claims are made as to fitness for any particular purpose. No
warranties of any kind are expressed or implied. The recipient agrees to determine applicability of information
provided. If this file has been purchased on magnetic or optical media from Unicode, Inc., the sole remedy for any
claim will be exchange of defective media within 90 days of receipt.
Limitations on Rights to Redistribute This Code
Unicode, Inc. hereby grants the right to freely use the information supplied in this file in the creation of products
supporting the Unicode Standard, and to make copies of this file in any form for internal or external distribution as
long as this notice remains attached.
PCRE 7.3
Copyright (c) 1997-2008 University of Cambridge

Copyright (c) 2007-2008, Google Inc.
PCRE is a library of functions to support regular expressions whose syntax and semantics are as close as possible
to those of the Perl 5 language.
Release 7 of PCRE is distributed under the terms of the "BSD" licence, as specified below. The documentation for
PCRE, supplied in the "doc" directory, is distributed under the same terms as the software itself.
The basic library functions are written in C and are freestanding. Also included in the distribution is a set of C++
wrapper functions.
THE BASIC LIBRARY FUNCTIONS

---------------------------
Written by: Philip Hazel
Email local part: ph10
Email domain: cam.ac.uk
University of Cambridge Computing Service,
Cambridge, England.
Page 19 of 22
Copyright (c) 1997-2008 University of Cambridge

THE C++ WRAPPER FUNCTIONS
-------------------------
Contributed by: Google Inc.
Copyright (c) 2007-2008, Google Inc.
THE "BSD" LICENCE

-----------------
* Redistributions of source code must retain the above copyright notice, this list of conditions and the following
disclaimer.
* Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the
following disclaimer in the documentation and/or other materials provided with the distribution.
* Neither the name of the University of Cambridge nor the name of Google Inc. nor the names of their
contributors may be used to endorse or promote products derived from this software without specific prior written
permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
POSSIBILITY OF SUCH DAMAGE.
HTTP Server functionality is provided by the GoAhead WebServer 2.1, Copyright (c) 2004 GoAhead Software, Inc.
All Rights Reserved.

Page 20 of 22
2010 Avaya Inc. All Rights Reserved.
Notice
While reasonable efforts were made to ensure that the information in this document was complete
and accurate at the time of printing, Avaya Inc. can assume no liability for any errors. Changes and
corrections to the information in this document may be incorporated in future releases.
Documentation disclaimer.
Avaya Inc. is not responsible for any modifications, additions, or deletions to the original published
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License
USE OR INSTALLATION OF THE PRODUCT INDICATES THE END USER’S ACCEPTANCE OF THE TERMS
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WITHIN TEN (10) DAYS OF DELIVERY FOR A REFUND OR CREDIT.Avaya grants End User a license
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capacity for which the license is granted will be one (1), unless a different number of licenses or
units of capacity is specified in the Documentation or other materials available to End User.
“Designated Processor” means a single stand-alone computing device. “Server” means a Designated
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Except where expressly stated otherwise, the Product is protected by copyright and other laws
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Avaya’s web site at: http://support.avaya.com/ThirdPartyLicense/
Avaya fraud intervention

Page 21 of 22
If you suspect that you are being victimized by toll fraud and you need technical assistance or
support, call Technical Service Center Toll Fraud Intervention Hotline at +1-800-643-2353 for the
United States and Canada. Suspected security vulnerabilities with Avaya Products should be
reported to Avaya by sending mailto: securityalerts@avaya.com.
For additional support telephone numbers, see the Avaya Web site:
http://www.avaya.com/support
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Avaya, AUDIX, DEFINITY, and COMMUNICATIONS MANAGER and the Avaya logo are registered
trademarks and/or service marks of Avaya Inc. in the U.S. and certain other jurisdictions.
All other trademarks are the property of their respective owners. Document ordering information:
For the most current versions of documentation, go to the Avaya support Web site:
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Avaya support
Avaya provides a telephone number for you to use to report problems or to ask questions about
your contact center. The support telephone number is 1 800 242 2121 in the United States. For
additional support telephone numbers, see the Avaya Web site: http://www.avaya.com/support
~~END OF FILE~~

Page 22 of 22

96xxSIP ReadmeR2 6

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96xx Release R2.

Date: June 7, 2010

New features introduced as part of Release 2.6

 Setup Environment  Feature  Feature description

Common Feature List Applicable for all CM/SES,CM/SM Setup

Copyright Avaya 2010 All rights reserved

such as Audiocodes, Cisco

 Requires CM 6.0  Call pickup alerting. When used with CM6.0+,

New and Enhanced features Applicable SM 6.0

Copyright Avaya 2010 All rights reserved

 Requires SM 6.0  Setting of voice mail The voice mail destination

 Requires SM 6.0  Improved user-friendly phone For example, a telephone

 Requires SM 6.0  Support geo-redundant Ensures that a telephone

 Requires SM 6.0  Support simultaneous Avaya one-X Deskphone SIP

 Requires SM 6.0  New extended call pickup  An extended call

Copyright Avaya 2010 All rights reserved

Field Issues Addressed in Release R2.6

Patch releases merged in this release

Known Limitations of Release 2.6

 Recommended SIP Transport Protocols

SIP Proxy / Phone Connection UDP TCP TLS

SES (as primary controller ) Supported Recommended Recommended

Avaya Session Manager (as Supported Recommended Recommended

AudioCodes MP-series analog Supported Recommended Supported

Cisco 2811 ISR (as secondary Supported Recommended Not supported

I55(as secondary controller) Supported Recommended Not supported

Teldat Vyda gateway Supported Recommended Supported

Table of Phone Connection Types for Survivability Configurations

Copyright Avaya 2010 All rights reserved

Known Issues of Release 2.6

 Issues Observed specific to 2.6 or previous releases :

Copyright Avaya 2010 All rights reserved

 Workaround: log off and log back in again.

 UI and application issues

Copyright Avaya 2010 All rights reserved

 Features Related Issues

Copyright Avaya 2010 All rights reserved

Feature Compatibility Matrix for Communication Manager (CM) and SIP

Minimum CM Minimum SES Version

SBM-24 button module – 5.0 5.1

SBM-24 button module – 5.0 5.1.2

SBM-24 button module – 5.1.2 5.1

Auto Answer 5.0 5.0

Deactivation of Call Forward On 5.2 5.2

Minimum CM Minimum SES Version

Visiting User (Applicable for SES 5.0 5.0

Extend Call 5.0 5.0

SRTP 4.0.1 n/a

All features other than those 4.0 4.0

Basic steps for setting up 96xx SIP telephones:

The basic steps for setting up 96xx SIP telephones:

Copyright Avaya 2010 All rights reserved

Setting up Primary and Secondary Controllers

The minimal settings required for provisioning SIP Proxies are:

Example: SET SIP_CONTROLLER_LIST proxy1:5060;transport=tcp,proxy2:5060;transport=tcp

Copyright Avaya 2010 All rights reserved

eg. When using single SIP CONTROLLER

When using Multiple SIP CONTROLLERs , please follow below examples

Copyright Avaya 2010 All rights reserved

 100REL_SUPPORT: If we set 100REL_SUPPORT 0 then phone doesn't 100rel in Supported

Invalid Proxy Settings

Specification of SEND_DTMF_TYPE Setting

Setup Environment Feature Feature description

Requires CM 6.0 Call pickup alerting. When used with CM6.0+,

Requires SM 6.0 Setting of voice mail The voice mail destination

Requires SM 6.0 Improved user-friendly phone For example, a telephone

Requires SM 6.0 Support geo-redundant Ensures that a telephone

Requires SM 6.0 Support simultaneous Avaya one-X Deskphone SIP

Requires SM 6.0 New extended call pickup An extended call

Recommended SIP Transport Protocols

Issues Observed specific to 2.6 or previous releases :

Workaround: log off and log back in again.

UI and application issues

Features Related Issues

100REL_SUPPORT: If we set 100REL_SUPPORT 0 then phone doesn't 100rel in Supported

Eight extended Korean ring tone files:

Four certificate files:

96xx SIP telephone IP address: 10.10.10.20

Reset your Avaya 9600 IP Telephones.