Audio Codec

Alliance & Application Partner Program
Inter-Working Report
Partner: AudioCodes
Application : Telephone Adapter / VoIP gateway
Application name: MP-202
The product and version listed have been tested with the Alcatel-Lucent Communication Server and the version
specified hereinafter. The tests concern only the inter-working between the Application Partner product and the
Alcatel-Lucent Communication platforms. The inter-working report is valid until the Application Partner issues a
new version of such product (incorporating new features or functionality), or until Alcatel-Lucent issues a new
version of such Alcatel-Lucent product (incorporating new features or functionality), whichever first occurs.
ALCATEL-LUCENT MAKES NO REPRESENTATIONS, WARRANTIES OR CONDITIONS WITH RESPECT TO
THE APPLICATION PARTNER PRODUCT. WITHOUT LIMITING THE GENERALITY OF THE FOREGOING,
ALCATEL-LUCENT HEREBY EXPRESSLY DISCLAIMS ANY AND ALL REPRESENTATIONS, WARRANTIES
OR CONDITIONS OF ANY NATURE WHATSOEVER AS TO THE APPLICATION PARTNER PRODUCT
INCLUDING WITHOUT LIMITATION THE IMPLIED WARRANTIES OF MERCHANTABILITY, NON
INFRINGEMENT OR FITNESS FOR A PARTICULAR PURPOSE AND ALCATEL-LUCENT FURTHER SHALL
HAVE NO LIABILITY TO APPLICATION PARTNER OR ANY OTHER PARTY ARISING FROM OR RELATED IN
ANY MANNER TO THIS CERTIFICATE.
Tests identification
Date of the tests
March 2009
Alcatels representative
Bruno Fouarge
<Bruno.Fouarge@alcatel-lucent.com>
Lars Ruoff
<Lars.Ruoff@alcatel-lucent.com>
Partners representative
Christophe Hubaut
Alcatel Communication Platform

Alcatel compatibility release
Partners application version
OmniPCX Enterprise
R9.0 (H1.301.27)
2.6.4
<christophe.hubaut@audiocodes.com>
Environment (if it has a sense)
Author(s):
Reviewer(s):
ACD
Business
Bruno Fouarge, Lars Ruoff

Rachid Himmi, Denis Lienhart, Bernard Roth
Historic
Edition 1: Creation of the document 2009-03-23
Edition 2: Added note about voice use restriction 2010-08-17
Test results
Passed
Refused
Postponed
Passed with restrictions
Refer to the section 4 for a summary of the test results.
Alliance & Application Partner Program Inter-working report

Copyright 2008 Alcatel-Lucent, All rights reserved
Edition 2 - page 2/41
Company Contact Information

(Information provided by the partner)
Contact name:
Title:
Bertrand Goldfarb
Sales Director
Address 1:
City:
Zip:
Country:
54, route des Sartrouville

LE PECQ
78230
France
Phone:
Fax:
+33 1 74 90 96 00
+33 1 39 76 89 61
Web address:
E-mail:
http://www.audiocodes.com
bertrand.goldfarb@audiocodes.com

TABLE OF CONTENTS
1
INTRODUCTION .......................................................................................................................................... 6
1.1
GLOSSARY ............................................................................................................................................... 6
APPLICATION INFORMATION ................................................................................................................ 7
TESTS ENVIRONMENT ............................................................................................................................ 8

3.1
3.2
3.3
GENERAL ARCHITECTURE ........................................................................................................................ 8

HARDWARE CONFIGURATION ................................................................................................................... 9
SOFTWARE CONFIGURATION .................................................................................................................... 9
SUMMARY OF TEST RESULTS ............................................................................................................ 11

4.1
SUMMARY OF MAIN FUNCTIONS SUPPORTED .......................................................................................... 11
4.2
SUMMARY OF PROBLEMS ...................................................................................................................... 11
4.2.1
AudioCodes side ........................................................................................................................ 11
4.2.2
Alcatel-Lucent side ..................................................................................................................... 11
4.2.3
Other problems (origin not identified) ...................................................................................... 11
4.3
SUMMARY OF LIMITATIONS .................................................................................................................... 11
4.4
NOTES, REMARKS ................................................................................................................................. 11
TEST SCENARIOS.................................................................................................................................... 13
5.1
5.2
TEST PROCEDURE ................................................................................................................................... 13

RESULT TEMPLATE ................................................................................................................................. 13
TESTING..................................................................................................................................................... 14
6.1
SIP SET COMMISSIONING ...................................................................................................................... 14
6.1.1
Test objectives ............................................................................................................................ 14
6.1.2
Test procedure............................................................................................................................ 14
6.2
OUTGOING CALLS .................................................................................................................................. 15
6.2.1
Test Objectives .............................................................................................................................. 15
6.2.2
Test Results ................................................................................................................................... 15
6.3
INCOMING CALLS ................................................................................................................................... 16
6.3.1
Test Objectives .............................................................................................................................. 16
6.3.2
Test Results ................................................................................................................................... 17
6.4
FEATURES DURING CONVERSATION ....................................................................................................... 19
6.4.1
Test Objectives .............................................................................................................................. 19
6.4.2
Test Results ................................................................................................................................... 19
6.5
CALL TRANSFER .................................................................................................................................... 19
6.5.1
Test Objectives .............................................................................................................................. 19
6.5.2
Test Results ................................................................................................................................... 20
6.6
ATTENDANT ........................................................................................................................................... 21
6.6.1
Test Objectives .............................................................................................................................. 21
6.6.2
Test Results ................................................................................................................................... 21
6.7
VOICE MAIL ........................................................................................................................................... 21
6.7.1
Test Objectives .............................................................................................................................. 21
6.7.2
Test Results ................................................................................................................................... 21
6.8
DUPLICATION AND ROBUSTNESS ........................................................................................................... 22
6.8.1
Test Objectives .............................................................................................................................. 22
6.8.2
Test Results ................................................................................................................................... 22
6.9
FAX....................................................................................................................................................... 24
6.9.1
Test objectives ............................................................................................................................... 24
6.9.2
Test procedure............................................................................................................................... 24
APPENDIX B : AUDIOCODES CONFIGURATION..................................................................................... 25

APPENDIX D: ALCATEL-LUCENT COMMUNICATION PLATFORM CONFIGURATION
REQUIREMENTS.............................................................................................................................................. 31
APPENDIX E: PARTNER ESCALATION PROCESS.................................................................................. 34

APPENDIX F: AAPP PROGRAM, DOCUMENTATION AND TECHNICAL ASSISTANCE................ 35
APPENDIX G: ALCATEL-LUCENT ESCALATION PROCESS IN CASE OF PROBLEM WITH A
CERTIFIED EXTERNAL APPLICATION (REFERENCED IN THE AAPP)........................................... 39

1 Introduction
The goal of these tests is to qualify an external application as an Alliance & Application
Partner Program solution for the Alcatel-Lucent Communication Platform.
The scope of the tests is the interoperability of the application with the Alcatel-Lucent
Communication Platform. It covers a basic or complex inter-working to ensure that services
requested by the application and provided by the Communication Platform (and/or
conversely) are properly completed.
These tests do not verify the functional achievement of the application as well as they do not
cover load capacity checks, race conditions and generally speaking any real customer's site
conditions.
This document is the interworking report between the AudioCodes MP-202 FXS working in
SIP mode - and an OmniPCX Enterprise (OXE) Call Server.
1.1 Glossary
ABC-F
: Alcatel Business Communication telephone Feature
ARS
: Automatic Route Selection
CS
: Call Server (Alize or Appliance Communication server)
CSTA
: Computer Supported Telecommunications Applications
CPU
: Call Processing Unit (4400 Communication Server)
DTMF
: Dual-Tone Multi-Frequency
GD
: Gateway Driver
IP
: Internet Protocol
ISDN
: Integrated Services Digital Network
MG
: Media Gateway
MLA
: Multi Line Appearance
OXE
: OmniPCX Enterprise
SIP
: Session Initialization Protocol
UA
: Universal Alcatel
UM
: Unified Messaging
: Alcatel-Lucent Analogical Set

2 Application information
Application type:
(Sets, Faxes).
Telephone Adapter / VoIP Gateway for Analog equipments
Application commercial name:
MP-202-B
Application version:
2.6.4_p7_build_9
Interface type:
SIP
Brief application description:

AudioCodes MP202 is a telephone adapter that allows connecting ordinary analog
telephones or fax machines to a Voice over Broadband service. It is typically adapted
for Branch Offices.
The MP202 connects to a Service Provider by using its IP uplink connection. It
proposes up to 2 VoIP ports for connecting up to 2 analog sets or faxes.
MP202 also supports the SIP protocol, used in the present case for communicating
with OXE. The equipments connected on the MP202 ports will therefore be declared
as SIP terminals (extern station) and will register on OXE.
MP202 also supports additional features such as routing, NAT, DHCP, PPPoE
functions as well as a built-in firewall. These features are not covered by the present
test campaign.

3 Tests environment
3.1 General architecture
The tests are performed on the Alcatel-Lucent TSS Applications International platform in the
following environment:
IP phones
Node 1
O XE/Aliz hardware
Duplicated call servers,
spatial redundancy
4645 Voice M ail
PSTN Netw ork

Sim ulated by loopback
Analog set
FAX
AudioCodes
M P-202-B
IP Netw ork
ABCF/IP
hybrid link
FAX
Node 2
4400/crystal hardware
Single CPU
SIP phone
Figure 1 - Test System Architecture

3.2 Hardware configuration
Alcatel-Lucent Communication Platform:
Node1:
o
o
o
o
OmniPCX Enterprise common hardware (Aliz)

Duplicated call servers
Spatial redundancy (Different IP subnetworks)
Two media gateways:
Cristal 0 :
+-------------------------------------------------------------------+
| Cr | cpl| cpl type
| hw type
| cpl state | coupler ID
|
|----|----|------------|-----------|--------------|-----------------|
| 0 | 6 |
CPU_CS|---------- |
IN SERVICE |
NO PCMS CODE |
| 0 | 10 |
CPU_CS|---------- |
IN SERVICE |
NO PCMS CODE |
+-------------------------------------------------------------------+
Cristal 1 :
+-------------------------------------------------------------------+
| hw type
|
|----|----|------------|-----------|--------------|-----------------|
| 1 | 0 |
GD|---------- |
IN SERVICE |
NO PCMS CODE |
| 1 | 1 |
UAI 8|---------- |
IN SERVICE |
NO PCMS CODE |
| 1 | 3 |
PRA T2|---------- |
IN SERVICE |
NO PCMS CODE |
| 1 | 6 |
PRA T2|---------- |
IN SERVICE |
NO PCMS CODE |
+-------------------------------------------------------------------+
Cristal 2 :
+-------------------------------------------------------------------+
| hw type
|
|----|----|------------|-----------|--------------|-----------------|
| 2 | 0 |
GD|---------- |
IN SERVICE |
NO PCMS CODE |
| 2 | 1 |
MIX448|---------- |
IN SERVICE |
NO PCMS CODE |
+-------------------------------------------------------------------+
Node2:
OmniPCX Enterprise crystal hardware (4400) voice hub
Single CPU
One INTIP:
Cristal 0 :
+-------------------------------------------------------------------+
| hw type
|
|----|----|------------|-----------|--------------|-----------------|
| 0 | 1 |
CPU7|---------- |
IN SERVICE |
BAD PCMS CODE |
| 0 | 3 |
UA32|---------- |
IN SERVICE |
BAD PCMS CODE |
| 0 | 4 |
INTIPA|
INT-IP |
IN SERVICE |
BAD PCMS CODE |
+-------------------------------------------------------------------+
Application platform:
AudioCodes MP-202-B
3.3 Software configuration
Alcatel-Lucent Communication Platform:

OmniPCX Enterprise R9.0 h1.301.25

Note: Handsets are declared as SIP extensions (SIP Endpoint Level of Service - SEPLOS). This
means that all telephone features must be tested once by prefixes and once locally on the
phone, if available.
Recommended configuration:
OXE > SIP > SIP Registrar > SIP Min Expiration Date = 60.
This is to allow for a 60 secs registration time for the IPBS, see IPBS recommended
configuration below.
Application platform:
AC MP-202 2.6.4

4 Summary of test results

4.1 Summary of main functions supported
The main supported functions implied in the test campaign are:
Installation and configuration of the AudioCodes gateway
Configuration of SIP users connected to the AudioCodes gateway on the OXE
registrar server
Registration and call authentication*
Basic communications (internal and external, intra-node, inter-nodes)
Supports telephonic features (Broker call, Transfer*, Conference, Call back, Waiting*,
Forwards*, Do Not Disturb, in intra-node and inter-nodes communications)..
Access to the OXE Voice mail boxes and MF dialing for navigation
Multi Frequency sending in conversation/connected state
Framing, VAD, Codec impact*
Fax (T38) transmission in intra and inter node
Simple and spatial call server redundancy
4.2 Summary of problems

4.2.1 AudioCodes side
Some transfer cases not working. See section 6.3.
Note: Due to this limitation we do not recommend to use AudioCodes MP202 for
use as a voice gateway (with analog voice phones). FAX operations are OK.
The restriction is present for both SIP device and SIP extension mode.
4.2.2 Alcatel-Lucent side
Forward to external number not possible. See section 6.3.
4.2.3 Other problems (origin not identified)
Music on hold : in case of remote calls, when the called set put the calling (MP202)
on hold, no music on hold is heard.
Local call to SIP terminal in forward on busy (CFB) state NOK.
4.3 Summary of limitations
Overlap dialing not supported.

After a second call, if you press the R key, you go in conference. If you press again
the R key, you release call the conference.=> Not possible to keep the second call.
Not possible to call sip device with attendant.
Message Waiting Indication (MWI): The message led on the analog phone is not
blinking.
4.4 Notes, remarks

Analog phones have been tested in SIP extension (SEPLOS) mode.

Fax has been tested in SIP device (trunk) mode. It can only operate in that mode.
Both modes can coexist on different lines, i.e. one line can be configured as SIP
extension, the other one as SIP device.

5 Test Scenarios
5.1 Test procedure
Step
Action
N/A
Result
Origin of the problem
Comment
Step: a test may comprise multiple steps depending on its complexity. Each step has to be completed
successfully in order to conform to the test. Step 0 when present represents the initial state for all the
following steps.
Action: describes which action to realize in order to set-up the conditions of the test.
N/A: the step within this test is not applicable to this application. This has to be filled in only if the test
is checked as mandatory in the applicability box. In that case, the column comment must indicate the
reason of the non-applicability (e.g.: service not supported).
Result: describes the result of the test from an external point of view. If it is positive, it describes
which application's trigger was checked. If it is negative, it describes as precisely as possible the
problem.
Origin of the problem: this column has to be filled in when a problem occurs during the test. It must
contain a high level evaluation of the localization of the responsibility: Alcatel-Lucent or the Partner.
it is not intended during this test session to debug and fix problems.
5.2 Result template

The results table must be formatted as indicated in the example below:
Step
1
2
Action
N/A
Result
. action 5
NOK
The application
waits for PBX
timer or phone
set hangs up
OK
. action 3
. action 4
X
Comment
OK
. action 1
. action 2
OK
3
4
Origin of the
problem
Relevant only if
the CTI
interface is a
direct CSTA
link
No indication,
no error
message

6 Testing
6.1 SIP set commissioning
6.1.1 Test objectives
These tests shall verify that the analog devices ( phones and fax machine ) connected to the MP-202
can be declared on OmniPCX Enterprise as SIP extension (SEPLOS) and are able to register properly
on the OXE server registrar.
6.1.2 Test procedure

Test Action
1 Provisioning
2 DHCP registration
(with OXE internal DHCP server)
3 NTP registration
/ The SIP phone is configured to
retrieve the date and time from the
OXE internal NTP server
Check the phone retrieves the
information and displays it.
SIP set registration to OXE in static IP
addressing mode / The phone is
configured to use IP static parameters
4A SIP registration, using OXE MAIN IP
addresse(s)
(without authentication)
4B SIP registration, using DNS
(without authentication) / The phone
is configured to use a domain name
as registrar / proxy server address.
The DNS IP address is the OXE CPU
main address
5 Support of 423 Interval Too Brief
(1)
6 SIP registration with authentication:
Turn on SIP Digest authentication,
specify realm on OXE, and specify
user name and password on SIP
client.
/ SIP digest authentication is
activated on OXE and phone side.
Check also that outgoing call is
authenticated.
Result
OK
Not
tested
OK
Comment
The NTP server of the MP202 is set as the

MAIN 1 CPU address (not the DNS name)
OK
OK
OK
OK
OK
The MP202 DNS is specified as Main1 and

the secondary DNS as MAIN 2.
By default,
OXE>SIP>SIP Registrar>Min Expires=1800s
It works for Registration authentication.
Not for Outgoing call with authentication.
Hence the OXE parameter in the SIP proxy
must be set as :
Only authenticated incoming call == false

Notes:
(1) On the SIP client, specify a default registration period inferior to that of OXE SIP registrar. OXE will
reject with error 423 Interval Too Brief. Check that SIP set increases registration period accordingly.
6.2 Outgoing Calls

6.2.1 Test Objectives
The calls are generated to several users belonging to the same network.
Called party can be in different states: free, busy, out of service, do not disturb, etc.
Calls to data devices are refused.
Points to be checked: tones, voice during the conversation, display (on caller and called party),
hang-up phase.
Note: dialing will be based on direct dialing number but also using programming numbers on the SIP
phone.
6.2.2 Test Results

Test Action
1 Call to a local user
2 Call to a local user with overlap
dialing:
Dial a part of the number, wait and
continue.

dialing,
timeout:
Result
OK
NOK
NOK
Comment
Overlap dialing not supported
Dial a part of the number, wait and stop.

dialing, release:
Dial a part of the number, wait and release

the call.
5 Call to local user with no answer.

Check timeout.
6 Call to another SIP set /
The SIP phone under test calls
another SIP phone. Check the display
and audio during all steps (dialing,
ring back tone, conversation, release).
7 Call to wrong number
(SIP: 404 Not Found)
8 Call rejected by call handling
(SIP: 183 Progress/487 Request
Terminated)
e.g. max number of calls reached
etc.
9 Call to busy user
(SIP: 486 Busy Here)
Check busy tone.
NOK
OK

Cancel sent by SIP device after few
seconds
OK
OK
OK

10
11
12
13
14
15
16
17
18
19
Call to user in Out of Service state

(SIP: 480 Temporarily Unavailable)
Call to user in Do not Disturb state
(SIP: 603 Decline)
Call to local user, immediate forward
(CFU).
(SIP: 302 Moved Temporarily)(1)
Call to local user, forward on no
reply (CFNR). (1)
Call to local user, forward on busy
(CFB). (1
Call to a local user with proxy
Authentication
Call within same IP domain.
SIP set in domain A (intradomain=without compression).
Call to OXE set in domain A (intradomain=without compression).
Call to another IP domain.
SIP set in domain A (extradomain=with compression).
Call to OXE set in domain B (extradomain=with compression).
Call to external number (via T2
loopback)
(Check ring back tone, called party
display)
/ Dial external number 03881234XY
=>1XY
SIP session timer expiration:
Check if call is maintained or
released after the session timer has
expired.
OK
OK
OK
OK
OK
OK
OK
Check order of codecs in SDP list.

Check chosen codec. We expect G.711.
See note (2)
OK
Check order of codecs in SDP list.

Check chosen codec. We expect G.729
See note (2)
OK
OK
Ring back tone ok but display not

available on analog device
Call is maintained by re- INVITE
Notes:
(1) For test cases with call to forwarded user: User is forwarded to another local user. Special case
of forward to Voice Mail is tested in another section.
(2) For IP domain tests, the following setup is used:
(output from command cnx dom)
| number |
0 |
1 |
2 |
|
type | NIPR | IP_R | IP_R |
| allowed | ffff |
1 |
10 |
|
used |
0 |
0 |
0 |
| IPP Intr| G711 | G711 | G711 |
| IPP Extr| G711 | G729 | G711 |
Partner SIP set is in domain 1.
6.3 Incoming Calls

Calls will be generated using the numbers or the name of the SIP user.
SIP terminal will be called in different states: free, busy, out of service, forward
The states are to be set by the appropriate system prefixes unless otherwise noted.
Points to be checked: tones, voice during the conversation, display (on caller and called party),
hang-up phase.
6.3.2 Test Results

Test Action
1 Local /network call to free SIP
terminal
/ Check ring back tone, called party
display
2 Local/network call to busy SIP
terminal
/ Call a set which is already in
conversation
3 Local/network call to unplugged SIP
terminal
4 Local/network call to SIP terminal in
Do Not Disturb (DND) mode
4A By local feature on MP202, using *73
prefix
4B By system feature (SEPLOS)
Dial 42 + user password wait for
acknowledgement ring back tone
from OXE
5 Local/network/SIP call to SIP
terminal in immediate forward (CFU)
to local user:
5A By local feature of MP202, using the
*72 prefix
Dial 51 + target extension wait for
from OXE / Dial 41 for forward
cancellation
to external number
6A By local feature if applicable
6B By system feature (SEPLOS) /

cancellation
to another SIP user
Result
OK
Comment
OK
OK
As long as the SIP device is registered, we

get a ring back tone. If is is not
registered, we are prompted to go on
hook.
OK
OK
OK
OK
NOK
NOK
Call not established but SIP device sends

the temporary moved with the right
external number. => OXE problem.
crms00168027
SIP device still gets INVITEd and rings. =>
OXE problem.
crms00168030

7A By local feature if applicable

cancellation
8 Local call to SIP terminal in forward
on busy (CFB) state:
8A By local feature if applicable *72

cancellation
9 Local call to SIP terminal in forward
on no reply (CFNR)
9A By local feature if applicable *72
cancellation
10 Call to busy user, Call waiting.
(Camp-on)(local feature)
11 External call to SIP terminal.
Check that external call back number
is shown correctly. / Dial external
number 03881234XY =>1XY
12 CLIR: Local call to SIP terminal.
/ Check that caller id is not shown.
13 Display: Call to free SIP terminal
from user with a name containing
non-ASCII characters. Check caller
display.
14 Display: Call to free SIP terminal
from user with a UTF-8 name
containing non-ASCII characters.
Check caller display.
15 SIP set is part of a sequential hunt
group. Call to hunt group. Check
call/release.
16 SIP set is part of a cyclic hunt group.
Call to hunt group. Check
call/release.
18 SIP set is declared as a twin set
(tandem). Call to main set and see if
twin set rings. Take call with twin
set.
OK
OK
NOK
NOK
Call not established but SIP device sends

the temporary moved with the right
external number. => OXE problem.
crms00168027
The SIP device sends 486 Busy here.
It behaves like a mono-line set, but is
declared as multi-line. OXE Problem: For
the moment it is not possible to remove
the 2nd line on a SIP extension.
OK
OK
N/A
Feature unavailable.
N/A
Feature unavailable.
N/A
Analogue device does not allow display of

the calling id
the calling id
N/A
N/A

the calling id
OK
OK
OK

6.4 Features during Conversation

Features during conversation between local user and SIP user must be checked.
Check that right tones are generated on the SIP phone.
6.4.2 Test Results

Test Action
1 Hold and resume (both directions)
using R key on sip device (Check
tones)
2 Second call to another local user.
Distant user is put on hold.
3 Broker request
(toggle back and forth between both
lines, local feature if applicable),
using the 1 suffix
4 Release first call. Keep second call.
Result
OK
OK
OK
NOK
5 Call park:
- Call between SIP set and OXE set.
- Put your call on hold.
- New call: Dial the prefix for call
parking (402+number).
Now call can be hung up.
Later call can be retrieved by calling
prefix+number again.
6 Send/receive DTMF
7 Three party conference initiated
from OXE set. Released by OXE set.
/ Dial extension target + suffix 3.
8 Three party conference initiated
from SIP set (local feature).
Released by SIP set.
Comment
Using the R + <number> on the Z phone
After the second call, if you press the R

key, you go in conference. If you press
again the R key, you release call the
conference.=> Not possible to keep the
second call
OK
OK
DTMF are sent as RFC2833
OK
OK
6.5 Call Transfer

During the consultation call step, the transfer service can be requested and must be tested.
Several transfer services exist: Unattended Transfer, Semi-Attended Transfer and Attended
Transfer.
Audio, tones and display must be checked.
We use the following scenario, terminology and notation:

There are three actors in a given transfer event:
A Transferee : the party being transferred to the Transfer Target.
B Transferor : the party doing the transfer.
C Transfer Target : the new party being introduced into a call with the Transferee.
The scenario is:
1. A (Transferee) calls B (Transferor).
2. B (Transferor) calls C (Transfer Target). A is on hold during this phase.
3. B (Transferor) executes the transfer. B drops out of the communication. A is now in
communication with C.
There are three sorts of transfers in the SIP world:
Unattended Transfer or Basic Transfer: Unattended Transfer consists of the Transferor
providing the Transfer Target's contact to the Transferee. The Transferee attempts to
establish a session using that contact and reports the results of that attempt to the
Transferor.
Semi-Attended Transfer or Early Attended Transfer or Transfer on ringing : Semi-Attended
Transfer consists of the Transferor having a session in hold state with the Transferee and
initiating the Transfer when the consultation call to the Target Transfer is in ringing state.
Attended Transfer or Transfer in conversation: Attended Transfer consists of the Transferor
having a session in hold state with the Transferee and initiating the Transfer when the
session between the Transferor and Transfer target is established.
Note: Unattended Transfer is not provided by the OXE.
6.5.2 Test Results

In the below table, SIP means a partner SIP set, OXE means a proprietary OXE (Z/UA/IP) set.
Att means attended transfer (transfer in conversation), Semi means Semi-attended transfer
(transfer on ringing) and Unatt means unattended transfer.
Test
Action
A
B
C
Transfe Transfe Transfe
ree
ror
r
Target
1
OXE
MP 202
OXE
2
OXE
MP 202
OXE
3
MP202
OXE
OXE
4
MP 202
OXE
OXE
5
OXE
6
OXE
7
SIP
8
SIP
9
OXE
10
OXE
11 MP 202
OXE
OXE
OXE
OXE
SIP
SIP
SIP
MP 202
MP 202
MP 202
MP 202
MP 202
MP 202
OXE
Type
Result
Att
Semi
Att
OK
OK
NOK
Semi
NOK
Att
Semi
Att
Semi
Att
Semi
Att
OK
OK
OK
OK
OK
OK
OK
Comment
User C doesnt hear user A. MP202

limitation. See ALu Ref 1-106975741 and
AC Ref #84185
User C doesnt hear user A. MP202
limitation. See ALu Ref 1-106975741 and
AC Ref #84185
See note (1)
See note (1)
See note (1)

12 MP 202
13 MP 202
14 MP 202
SIP
SIP
SIP
OXE
SIP
SIP
Semi
Att
Semi
OK
OK
OK
See note (1)
Note (1): We define two different semi-attended transfers. The transfer on ringing (when target set
rings at least one time) and immediate transfer (when the transferor goes on hook immediately
after having transferred the initial caller without waiting for the target ring).
6.6 Attendant
An attendant console is defined on the system. Call going to and coming from the attendant console
are tested.
6.6.2 Test Results
Test Action
1 call to attendant (using attendant
call prefix 9)
call prefix 9), attendant transfers
to OXE set, unattended.
call prefix 9), attendant transfers
to OXE set, attended.
4 OXE set calls to attendant (using
attendant call prefix 9), attendant
transfers to SIP set, attended.
Result
OK
Comment
OK
OK
OK
6.7 Voice Mail

Voice Mail notification, consultation and password modification must be checked.
MWI (Message Waiting Indication) has to be checked.
6.7.2 Test Results

Test Action
1 A Voice Mail message for the SIP
subscriber is generated.
Check that MWI is activated.
/ Dial prefix 570 + target extension
or target extension + suffix 8
2 Message consultation
3 Password modification
Result
(OK)
Comment
The message led on the Z phone is not
blinking. Probably a poroblem of Z phone
compatibility with voltages.
OK
OK

4 MP202 call to a OXE user forwarded

to Voice Mail
/ Dial 51 + VoiceMail extension,
wait for acknowledgement ring back
tone from OXE / Dial 41 for
forward cancellation
5 OXE call to a MP202 user forwarded
to Voice Mail
/ Dial 51 + VoiceMail extension,
wait for acknowledgement ring back
tone from OXE / Dial 41 for
forward cancellation
OK
OK
6.8 Duplication and Robustness

Check how the system will react in case of a CPU reboot, switchover or link failure etc.
The test system is configured with spatial redundancy (duplicate call servers on two different IP
subnetworks).
Spatial redundancy can be configured in two ways:
Alternate Proxy method: Specify both CS MAIN addresses as primary and alternative proxy
respectively. Requires that on non-availability of primary proxy, secondary proxy is used. Requires
ability to accept incoming calls from secondary proxy.
DNS method: Do not specify a proxy address, only SIP domain. Specify the CS MAIN addresses as
first and second DNS server, respectively. Requires that (at least on non availability of current
proxy) a new DNS request is issued for every message. Only MAIN CS will respond. Requires ability to
accept incoming calls from secondary CS when it becomes new MAIN.
For each configuration, check:
Can new outgoing calls be made immediately after switchover?
Are existing calls maintained after switchover?
Are incoming calls (from new MAIN CS) accepted immediately after switchover?
Can existing call be modified (transfer, hang-up, etc.) after switchover?
Check if a session that has been started before switchover is maintained after switchover, i.e. does
the new MAIN CS send session updates and is this accepted by the client?
6.8.2 Test Results

Test Action
1 Spatial redundancy, using Alternate
Proxy method: Switchover to
standby call server.
Result
(OK)
Comment
Works with restrictions:
Still need to have DNS configured,
because when no DNS is configured, MP202 cannot resolve the Contact FQDN.
Can only call after re-registered on new
proxy. => requires frequent registration to
minimize downtime after switchover.
Not recommended.

2 Spatial redundancy, using DNS

method: Switchover to standby call
server.
3 Switchover to Passive Call Server
(PCS). (IP link to main/stdby call
servers down)
4 SIP device reboot.
Check that calls are possible as soon
as device has come back to service.
5 Temporary Link down with the PBX
OK
N/A
Works very well, seamless switchover.

DNS requests are sent simultaneously to
CPU A and B. => no delay.
Recommended method.
Not tested, not available in R9.0
OK
OK

6.9 FAX
6.9.1 Test objectives
Perform fax transmission in T.38 mode

Note: For these tests, the user is declared in SIP device (trunk) mode
6.9.2 Test procedure

Test
1
2
3
4
5
6
Action
single fax (local)
Fax local forwarding
Fax sending to busy fax machine
single fax through PSTN
single fax (remote)
long and complex fax
Result
OK
OK
OK
OK
OK
OK

Comment
Appendix B : AudioCodes Configuration

MP202 configuration
(Spatial redundancy, with DNS method)
CS A MAIN
CS B MAIN
IP addresses setting (WAN, DNS method)

Node+domain name
SIP domain declaration
Node+domain name
SIP proxy declaration (DNS method, recommended)

CS A MAIN
Redundant proxy method

CS B MAIN
SIP proxy declaration (redundant proxy method, NOT recommended!)

Codecs setting

Services setting

Analog set configuration

Appendix D: Alcatel-Lucent Communication Platform

Configuration Requirements
Handsets are declared as SIP extensions for analog phones, and as SIP device for FAX users:
Review/Modify: Users
Node Number (reserved) : 1
Directory Number : 160
Directory name : SIP_160
Directory First Name : Anna
UTF-8 Directory Name : ---------------------------------------
UTF-8 Directory First Name : ---------------------------------------
Location Node : 1
Shelf Address : 255
Board Address : 255
Equipment Address : 255
Set Type + SIP extension
Entity Number : 1
Set Function + Default
Profile Name : --------------
Key Profiles + None
Domain Identifier : 0
Language ID : 1
Secret Code : ****
SIP Gateway configuration as used in tests:

Review/Modify: SIP Gateway
Instance (reserved) : 1
SIP Subnetwork : 4
SIP Trunk Group : 4
IP Address : 172.25.32.252
Machine name - Host : vega
SIP Proxy Port Number : 5060
SIP Subscribe Min Duration : 1800
SIP Subscribe Max Duration : 86400
Session Timer : 180
Min Session Timer : 180
Session Timer Method + RE_INVITE
DNS local domain name : lab.ts
DNS type + DNS A
SIP DNS1 IP Address : ---------------------------------------
SIP DNS2 IP Address : ---------------------------------------
SDP IN 180 + True
Cac SIP-SIP + True

SIP Registrar configuration as used in tests:

Review/Modify: SIP Registrar
SIP Min Expiration Date : 60
SIP Max Expiration Date : 86400
List of prefixes and suffixes defined on OmniPCX TSS lab system (default):
+-----------------+----------------------------------------------------------+
|dir
|mean
|
+-----------------+----------------------------------------------------------+
|400
|Set_In/Out_of_service
|
|401
|Recordable_Voice_Guides
|
|402
|Park_Call/Retrieve
|
|403
|Charging_meter_readout
|
|404
|Associated_Set_No_Modif
|
|405
|Password_modification
|
|406
|Redial_last_number
|
|407
|Night_service_answering
|
|408
|Contrast_programmation
|
|409
|Secret/Identity
|
|41
|Forward_cancellation
|
|42
|Do_not_disturb
|
|43
|Voice_Mail
|
|44
|Canc_auto_call_back_on_busy
|
|45
|PadLock
|
|46
|Consult_Call_back_list
|
|470
|Waiting_call_consultation
|
|471
|Business_account_code
|
|472
|Consult_Messages
|
|473
|Paging_call_answer
|
|474
|Language
|
|480
|Set_group_entry
|
|481
|Set_group_exit
|
|482
|Switch_off_Message_LED
|
|483
|Mask_Remote_Calling_Identity
|
|484
|Cancel_Remote_forward
|
|485
|Overfl_busy_to_assoc_set
|
|486
|Overf_busy/no_repl_assoc_set
|
|487
|Recording_Conversation
|
|490
|Ubiquity_Mobile_Programming
|
|491:493
|Ubiquity_Services_Pfx
|
|495
|Ubiquity_Assistant
|
|500
|Last_Caller_Call_back
|
|501
|Remote_forward
|
|502
|Overflow_on_associated_set
|
|503
|Cancel_Overfl_on_assoc_set
|
|504
|Protection_against_beeps
|
|505
|Substitution
|
|506
|Wake_up/appointment_remind
|
|507
|Cancel_Wake_up
|
|508
|Forward_cancel_by_destinat
|
|509
|Meet_me_Conference
|
|51
|Immediate_forward
|
|52
|Immediate_forward_on_busy
|
|53
|Forward_on_no_reply
|
|54
|Forward_on_busy_or_no_reply
|
|55
|Direct_call_pick_up
|
|56
|Group_call_pick_up
|
|570
|Voice_Mail_Deposit
|
|580
|Tone_test
|
|581
|Personal_directory_Progr
|
|582
|Personal_Directory_Use
|

|583
|584
|585
|586
|587
|588
|589
|591
|599
|899
|9
|*
|#
|Force_type_identification_pfx
|Suite_Wakeup
|Suite_Wakeup_Cancel
|Suite_Dont_Disturb
|Room_status_management
|Mini_bar
|Direct_Paging_Call
|Pabx_address_in_DPNSS
|Professional_trunk_seize
|Pabx_address_in_DPNSS
|Attendant_Call
|DTMF_End_to_End_Dialling
|Speed_call_to_associated_set

|
|
|
|
|
|
|
|
|
|
|
|
|
Appendix E: Partner escalation process

In case you would need technical assistance, please contact the reseller/distributor where you
purchased your AudioCodes products. They have been trained on the products to give you 1st and
2nd levels of support. They are in plus in direct relation with 3rd level AudioCodes support in case an
escalation would be needed.

Appendix F: AAPP program, documentation and technical

assistance
Alliance & Application Partner Program (AAPP)

Complete e-business solutions at your disposal
The Alliance & Application Partner Program is designed to support companies that develop
communication applications for the enterprise market, based on Alcatel-Lucent's Omni product family.
The program provides tools and support for developing, verifying and promoting compliant third-party
applications that complement Alcatel-Lucent's Omni-based products. Alcatel-Lucent facilitates market
access for compliant applications.
The Alliance & Application Partner Program (AAPP) has two main objectives:
Provide
easy
interfacing
for
Alcatel-Lucent
communication
products:
Alcatel-Lucent's communication products for the enterprise market include infrastructure
elements, platforms and software suites. To ensure easy integration, the AAPP provides a full
array of standards-based application programming interfaces and fully documented
proprietary interfaces. Together, these enable third-party applications to benefit fully from the
potential of Alcatel-Lucent products.
Test
and
verify
a
comprehensive
range
of
third-party
applications:
to ensure proper inter-working, Alcatel-Lucent tests and verifies selected third-party
applications that complement its portfolio. Successful candidates, which are labeled AlcatelLucent Compliant Application, come from every area of voice and data communications.
The Alliance & Application Partner Program covers a wide array of third-party applications/products
designed for voice-centric and data-centric networks in the enterprise market, including terminals,
communication applications, mobility, management, security,
Web site
If registered Alliance & Application Partner, you can access the AAPP website at this URL:
http://www.applicationpartner.alcatel-lucent.com
Alcatel-Lucent.com
You can access the Alcatel-Lucent website at this URL: http://www.Alcatel-Lucent.com/

Alcatel-Lucent documentation
Alliance & Application Partner Program (AAPP)
If registered Alliance & Application Partner, you can access the current AAPP documentation at this
URL:
http://www.applicationpartner.alcatel-lucent.com and then click the Partner Center link.
Alcatel-Lucent Business Partner Program (ABPP)

The Alcatel-Lucent Business Partner Program is designed to empower and maximize the business of
the Partners. In addition, it enables them to help their customers successfully maximize their telecom
investment through optimum deployment and proper configuration of Alcatel-Lucent's solutions.
Alcatel-Lucent Partners also receive the added benefit of rapid, highly qualified service and support as
well as world-class training. Alcatel-Lucent will work closely with Business Partners to provide top
quality design, delivery, and support of the very best solutions for your customers. The Business
Partner Program is designed around a flexible and scalable framework so each Partner can identify
the exact support they need. So, depending on your specific requirements you can quickly become a
'Certified', 'Expert' or 'Premium' Business Partner with one of the world leaders in the communications
industry.
If registered Alcatel-Lucent Business Partners, you can access to an exciting on-line resource centre
with a wealth of information on all product lines at this URL:
http://www.businesspartner.Alcatel-Lucent.com

Technical assistance
In order to guide you in your purchasing decisions and provide you with assistance for updating our
Communication Server and Networking Infrastructure products and for commercial development,
Alcatel-Lucent has created the SUPPORT CENTER. The SUPPORT CENTER is responsible for the
management and routing of all your requests. It includes e-Support and a Contact Centre reserved
for registered Alliance & Application Partner and Alcatel-Lucent Business Partners.
The Contact Centre is open 24 hours a day; 7 days a week and is available in 5 languages. This Call
Centre has a team of 15 people and handles 10; 000 requests per month.
e-Support from the Alliance & Application Partner Web site (if registered Alliance & Application
Partner): http://www.applicationpartner.alcatel-lucent.com click the Partner Center link and
then Support
e-Support from the Alcatel-Lucent Business Partners Web site (if registered Alcatel-Lucent
Business Partners): http://www.businesspartner.Alcatel-Lucent.com click the e-Support link
and then e-Service Request
e-mail: Support.Center@Alcatel-Lucent.fr
Fax number: +33 (0) 3 90 67 73 45
Telephone numbers:
Alcatel-Lucent Business Partner Contact Center:
France :
Austria :
Denmark :
Germany :
Ireland :
Italy :
UK :
Spain :
Switzerland :
0 811 900 110

0 810 810 012
70 11 21 09
0 1 803 000 680
1 890 925 039
848 800 389
0 845 601 4101
901 120 085
0 844 850 588
French agent
German agent
English agent
German agent
English agent
Italian agent
English agent
Spanish agent
German agent
For other countries:

English answer :
French answer :
+ 33 (0)3 88 55 69 04
+ 33 (0)3 88 55 69 02

Alcatel-Lucent training
Technological innovative cycles are quickening and your customers are more and more demanding
regarding the quality of services. In order to meet these requirements, you have to invest in skills: a
key success factor for services.
If registered Alcatel-Lucent Business Partners, you can access to the training part at this URL:
http://www.businesspartner.Alcatel-Lucent.com and then click the Training link.
Our vision of learning services is described in the Services Portfolio section. The Certification
section gives you some statistics and details on how training curricula are designed to match
certification levels.
All updated training curricula and assessment tools are available in the Curricula & Catalogues
section.
The Schedule section is regularly updated to show forthcoming training sessions over the world.
The How to Enrol section provides you with the registration procedure and the Alcatel-Lucent
University Customer Service list of contacts world wide.
Last but not least, find statistics and reports of what you think about our training services in the
Customer Satisfaction section.
The Alcatel-Lucent commitment : enabling you to optimize your training investments.

Appendix G: Alcatel-Lucent escalation process in case of

problem with a certified external application (referenced in the
AAPP)
Introduction
The purpose of this document is to define the split of responsibilities and the escalation process to be
applied by the Business Partners when facing a problem with a solution involving an
Alcatel-Lucent platform and an external application with a valid Alcatel-Lucent Inter-working
report.
As for other Alcatel-Lucent equipment, the Alcatel-Lucent business partner is the only one facing the
end-customer for support or maintenance. The Business partner will open cases (service request)
either on Alcatel-Lucent side or on Application Partner side depending on the nature of the issue.
Expert from both companies will collaborate to provide the best and quickest correction.
Alcatel Technical Support
Business Partner
Customer
Answer
SR
Experts
Exchanges
Technical
Issues
In cases of the problem has

not been sent to the right
organization: Please open a
SR into our partner database
by mentioning our SR number
Partners
answers
SR
Answer
Application Partner Support
SR = Service Request

General Rules
The following general rules are applied:
Only certified AAPP applications are officially supported by Alcatel-Lucent
The certification is based on tests suite passed by Alcatel-Lucent and the

Application Partner and the result is consigned into an Inter-Working Report (IWR)
validated by the two parties.
The IWR is available on the AAPP Web site.
Only the major releases of both parties are certified. Certification tests are usually not
performed for intermediate versions. Only the existence of the IWR in the AAPP Web site for
the right Alcatel-Lucent release is the guarantee that the application has been certified
with this Alcatel-Lucent release.
If the IWR for the Alcatel-Lucent release is not available, Alcatel-Lucent doesnt engage any
responsibility. In that case, please contact the central Pre-Sales team.
The existence of the IWR engages Alcatel-Lucent and the Application Partner. Both
parties are engaged, not exclusively Alcatel-Lucent (see the section escalation process).
Warning:
The possibility to configure the Alcatel-Lucent PBX with ACTIS quotation tool in order to interwork
with an external application, is not a guarantee of the availability of the solution. Please check the
availability of the Inter-Working Report on AAPP web site.

The escalation process

As stated above, the Alcatel-Lucent support will be limited to applications with a valid inter-working
report. Known problems or remarks mentioned in the IWR will not be taken into account.
In case of problem, the two parties, Alcatel-Lucent and the Application Partner, are engaged:
Case 1: the responsibility can be established 100% on Alcatel-Lucent side

In that case, the problem must be escalated by the Business Partner to the Alcatel-Lucent
Hot-line via the standard process: open a ticket (Service Request SR)
Case 2 : the responsibility can be established 100% on Application Partner side

In that case, the problem must be escalated directly to the partner by opening a ticket trough
the Partner Hotline. In general, the process to be applied for the partner side is described in
the IWR.
Case 3 : the responsibility can not be established

In that case the following process applies:
1) The Application Partner shall be contacted first by the Business Partner or the party
responsible for that Application for an analysis of the problem.
Alcatel-Lucent has to be involved solely if the application partner demonstrate, with traces,
after reproduction of the problem, that the defect which has generated the end-users demand
of support is coming from the equipment provided by Alcatel-Lucent or if he needs support of
Alcatel-Lucent.
2) The Business partner will escalate the problem to the Alcatel-Lucent Hot-line if the
Application Partner has demonstrated a problem on Alcatel-Lucent side or if the Application
Partner (not the Business Partner) needs the involvement of Alcatel-Lucent.
In that case, the Business Partner must provide the reference of the Case Number on
Application Partner side. The Application Partner must provide to Alcatel-Lucent the results
of its investigations, traces, etc, related to this Case Number.
Alcatel-Lucent reserves the right to close the case opened on his side if the investigations
made on Application Partner side are insufficient or do no exist.
Note:
Involvement of the Business Partner is mandatory because the access to the Alcatel-Lucent
Platform (remote access, login/password) is under the Business Partner responsibility.


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Alliance & Application Partner Program

Alcatel Communication Platform

Environment (if it has a sense)

Bruno Fouarge, Lars Ruoff

Passed with restrictions

Refer to the section 4 for a summary of the test results.

Alliance & Application Partner Program Inter-working report

Edition 2 - page 2/41

Company Contact Information

54, route des Sartrouville

Alliance & Application Partner Program Inter-working report

Edition 2 - page 3/41

APPLICATION INFORMATION ................................................................................................................ 7

TESTS ENVIRONMENT ............................................................................................................................ 8

GENERAL ARCHITECTURE ........................................................................................................................ 8

SUMMARY OF TEST RESULTS ............................................................................................................ 11

TEST PROCEDURE ................................................................................................................................... 13

APPENDIX B : AUDIOCODES CONFIGURATION..................................................................................... 25

Edition 2 - page 4/41

APPENDIX E: PARTNER ESCALATION PROCESS.................................................................................. 34

Alliance & Application Partner Program Inter-working report

Edition 2 - page 5/41

: Alcatel Business Communication telephone Feature

: Automatic Route Selection

: Call Server (Alize or Appliance Communication server)

: Computer Supported Telecommunications Applications

: Call Processing Unit (4400 Communication Server)

: Integrated Services Digital Network

: Multi Line Appearance

: Session Initialization Protocol

: Alcatel-Lucent Analogical Set

Alliance & Application Partner Program Inter-working report

Edition 2 - page 6/41

Telephone Adapter / VoIP Gateway for Analog equipments

Application commercial name:

Brief application description:

Alliance & Application Partner Program Inter-working report

Edition 2 - page 7/41

PSTN Netw ork

Figure 1 - Test System Architecture

Alliance & Application Partner Program Inter-working report

Edition 2 - page 8/41

3.2 Hardware configuration

Alcatel-Lucent Communication Platform:

OmniPCX Enterprise common hardware (Aliz)

3.3 Software configuration

Alcatel-Lucent Communication Platform:

Alliance & Application Partner Program Inter-working report

Edition 2 - page 9/41

Alliance & Application Partner Program Inter-working report

Edition 2 - page 10/41

4 Summary of test results

4.2 Summary of problems

4.2.2 Alcatel-Lucent side

Forward to external number not possible. See section 6.3.

4.2.3 Other problems (origin not identified)

4.3 Summary of limitations

Overlap dialing not supported.

4.4 Notes, remarks

Edition 2 - page 11/41

Analog phones have been tested in SIP extension (SEPLOS) mode.

Alliance & Application Partner Program Inter-working report

Edition 2 - page 12/41

Origin of the problem