AudiocodesMediant1000-MSBG Lync OmniPCXEnterpriseR10.0 IWR Ed01

Alcatel-Lucent Application Partner Program
Inter-Working Report
Partner: AudioCodes
Application type: Media Gateway
Application name: Mediant 1000 MSBG /Microsoft Lync
Alcatel-Lucent Platform: OmniPCX Enterprise
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.
Alcatel-Lucent Application Partner Program Inter-working report

Copyright 2011 Alcatel-Lucent, All rights reserved
Edition 1 - page 1/54
Tests identification
Date of the tests
October 2011
Alcatel-Lucents representative
Partners representative
FOUARGE Bruno
RESIDORI Florian
AHARONOV Ofer
Alcatel-Lucent Communication OmniPCX Enterprise

Platform (OmniPCX
4400/Enterprise, OmniTouch,
OmniPCX Office, ...)
Alcatel-Lucent compatibility
R9.1(i1605.21) and R10
release
(j1410.34c)
AudioCodes : 6.20A.034.004
Partners application version
Lync : R. 2010 4.0.7577.0
Environment (if it has a sense)
Author(s):
Reviewer(s):
ACD
Business
FOUARGE Bruno, RESIDORI Florian

Denis LIENHART, Bernard ROTH
Historic
Edition 1: creation of the document January 2011
Test results
Passed
Refused
Postponed
Passed with restrictions
Refer to the section 4 for a summary of the test results.

Company Contact Information
Contact name:
Title:
Ofer Aharonov
Interoperability Engineer
Address 1:
Address 2:
City:
State:
Zip:
Country:
Country code:
1 Hayarden St. AirPort City

Lod
Israel
70151
Israel
Phone:
Fax:
+972-3-9764210
+972-3-9764040
Web address:
E-mail:
http://www.AudioCodes.com
Ofer.Aharonov@AudioCodes.com

TABLE OF CONTENTS
1
Introduction ............................................................................................ 6
GLOSSARY ............................................................................................... 6
Application information .............................................................................. 7
Tests environment .................................................................................... 8
3.1
GENERAL ARCHITECTURE ................................................................................. 8
3.2
HARDWARE CONFIGURATION.............................................................................. 9
3.3
SOFTWARE CONFIGURATION ............................................................................. 10
Summary of test results ............................................................................ 11
4.1
SUMMARY OF MAIN FUNCTIONS SUPPORTED ............................................................... 11
4.2
SUMMARY OF PROBLEMS ................................................................................. 11
4.2.1
OXE ........................................................................................................................................... 11
4.2.2
Audicodes/Lync ...................................................................................................................... 11
4.3
SUMMARY OF LIMITATIONS ............................................................................... 11
4.3.1
OXE ........................................................................................................................................... 11
4.3.2
AudioCodes/Lync .................................................................................................................... 11
4.4
NOTES, REMARKS ....................................................................................... 11
Test Scenarios ........................................................................................ 12
5.1
TEST PROCEDURE ....................................................................................... 12
5.2
RESULT TEMPLATE ...................................................................................... 12
Testing .................................................................................................. 13
6.1
BASICS OUTGOING CALLS: OXE TO LYNC ................................................................ 13
6.1.1
Calls from an OXE user set to a Lync user......................................................................... 13
6.2
INCOMING CALLS: LYNC => OXE ........................................................................ 14
6.2.1
Reception of calls .................................................................................................................. 14
6.3
FEATURES DURING CONVERSATION ....................................................................... 15
6.3.1
Hold, Consultation call and broker call ............................................................................. 15
6.3.2
Transfer call ........................................................................................................................... 17
6.3.3
Conference .............................................................................................................................. 20
6.4
VOICE MAIL............................................................................................. 21
6.4.1
Test Objectives....................................................................................................................... 21
6.4.2
Test Results ............................................................................................................................. 21
6.5
ATTENDANT ............................................................................................ 21
6.5.1
Test Objectives....................................................................................................................... 21
6.5.2
Test Results ............................................................................................................................. 21
6.6
DEFENSE / RECOVERY................................................................................... 22
6.6.1
Test Objectives....................................................................................................................... 22
6.6.2
Test Results ............................................................................................................................. 22
Appendix A : AudioCodes Mediant 1000 MSBG Configuration ................................ 23
7.1
GETTING STARTED ...................................................................................... 23
7.2
CONFIGURATION PROCEDURE............................................................................ 24
7.2.1
Configure IP Address ............................................................................................................. 24
7.2.2
VoIP DNS Settings ................................................................................................................... 24
7.2.3
Enable Applications ............................................................................................................... 24
7.2.4
SIP General Parameters ........................................................................................................ 25
7.2.5
DTMF & Dialing ....................................................................................................................... 27
7.2.6
Coders ...................................................................................................................................... 27
7.2.7
Configure IP Profile ............................................................................................................... 28
7.2.8
Configure IP Group Tables.................................................................................................... 29
7.2.9
Configure Proxy Sets Tables ................................................................................................ 31
7.2.10 Configure Proxy & Registration ........................................................................................... 33
7.2.11 Configure IP Media................................................................................................................. 34
7.2.12 Configure Routing .................................................................................................................. 35
7.2.13 Configure Manipulation ........................................................................................................ 36
7.2.14 Configure Reasons for Alternative Routing....................................................................... 36
7.2.15 Header Manipulation for Voice Mail ................................................................................... 37
7.2.16 Admin Page Parameters ........................................................................................................ 38
7.2.17 Reset the Gateway................................................................................................................. 39
1.1
2
3

7.3
AUDIOCODES INI FILES .................................................................................. 40
8
Appendix B: Alcatel-Lucent Communication Platform: configuration requirements .... 41
9
Appendix C: Partner escalation process ......................................................... 48
10
Appendix D: AAPP program ......................................................................... 49
10.1
ALCATEL-LUCENT APPLICATION PARTNER PROGRAM (AAPP) .......................................... 49
10.2
ALCATEL-LUCENT.COM ............................................................................... 49
11
Appendix E: AAPP Escalation process............................................................. 50
11.1
INTRODUCTION ....................................................................................... 50
11.2
ESCALATION IN CASE OF CERTIFIED APPLICATION/PRODUCTS ........................................... 51
11.3
ESCALATION IN CASE OF NON-CERTIFIED APPLICATION/PRODUCT ....................................... 52
11.4
TECHNICAL SUPPORT ACCESS ......................................................................... 53

1 Introduction
The goal of these tests is to qualify the interworking between the OmniPCX Enterprise and
Microsoft Lync via the SIP Gateway Mediant 1000 MSBG.
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.
1.1 Glossary
Acronym
Meaning
OXE
OmniPCX Enterprise
Transferee
The party being transferred to the transfer target
Transferor
The party initiating the transfer
Transfer target
The new party being introduced into a call with the transferee
Blind or semi-attended transfer
Attended transfer or transfer on

conversation
CLIP
The transferor having a session in hold state with the transferee and
initiating the transfer by a consultation call to the target performs the
transfer while the target is in ringing state
The transferor waits to be in conversation state with the target before
completing the transfer
Calling Line Identification Presentation
CNIP
Calling Name Identification Presentation
CLIR
Calling Line Identification Restriction
CNIR
Calling Name Identification Restriction
COLP
Connected Line Identification Presentation
CONP
Connected Name Identification Presentation
COLR
Connected Line Identification Restriction
CONR
Connected Name Identification Restriction
CRC
Customer Relationship Centre
PG
SIP GW XXX Peripheral Gateway
ICM
SIP GW XXX Intelligent Call Manager
CCM
CVP
SIP GW XXX Call Manager

Customer voice Portal
BC
ALU Business Contact
CTI
Computer Telephony Integration
PSTN(analogique card on the OXE)
Public Switched Telephone Network

2 Application information
Application type: SIP GW Mediant 1000 MSBG Customer Relationship Center includes SIP GW
Mediant 1000 MSBG Call Manager (CCM) and SIP GW Mediant 1000 MSBG Customer Voice Portal
(CVP)
Application commercial name: Mediant 1000 MSBG
Application version:
6.20A.034.004
Interface type:
SIP/IP
Brief application description:

Office
Basic outgoing call
Basic incoming call
Hold/Retrieve
Consultation call
Transfer
Conference/Cancel
Voice Mail

OK_but

Figure 1 Global architecture
The Microsoft Lync Server and Mediation Server connect to the Sip Gateway Mediant 1000 MSBG
which is connected to the OmniPCX Enterprise via a private SIP Trunk.

3 Tests environment
3.1 General architecture
The tests are performed on OXE platform hosted in ALU site and AudioCodes site.
Figure 2 Tests environnent

3.2 Hardware configuration
OmniPCX Enterprise node1:

OmniPCX Enterprise common hardware
Duplicated call servers
Spatial redundancy (Different IP subnetworks)
Media gateways + T2 card*2 (local Loopback)
IP Phones 4038,4039 and 4068
UA, Analog phones
Crystal 0:
+-------------------------------------------------------------------+
| Cr | cpl| cpl type
| hw type
| cpl state | coupler ID
|
|----|----|------------|-----------|--------------|-----------------|
| 0 | 6 |
CPU_CS|---------- |
IN SERVICE |
BAD PCMS CODE |
| 0 | 10 |
CPU_CS|---------- |
IN SERVICE |
BAD PCMS CODE |
+-------------------------------------------------------------------+
Crystal 1:
+-------------------------------------------------------------------+
| hw type
|
|----|----|------------|-----------|--------------|-----------------|
| 1 | 0 |
GD|---------- |
IN SERVICE |
BAD PCMS CODE |
| 1 | 1 |
MIX244|---------- |
IN SERVICE |
BAD PCMS CODE |
| 1 | 2 |
PRA T2|---------- |
IN SERVICE |
NO PCMS CODE |
| 1 | 3 |
PRA T2|---------- |
IN SERVICE |
BAD PCMS CODE |
| 1 | 4 |
GA|---------- |
IN SERVICE |
BAD PCMS CODE |
| 1 | 5 |
APPLI|---------- |
LANX 16 |
BAD PCMS CODE |
| 1 | 6 |
PRA T2|---------- |
IN SERVICE |
BAD PCMS CODE |
| 1 | 8 |
CS|---------- |
CS+4645 |
BAD PCMS CODE |
+-------------------------------------------------------------------+
OmniPCX Enterprise Node2:

OmniPCX Enterprise common hardware
Single call server
Crystal 0 :
+-------------------------------------------------------------------+
| hw type
|
|----|----|------------|-----------|--------------|-----------------|
| 0 | 6 |
CPU_CS|---------- |
IN SERVICE |
BAD PCMS CODE |
| 0 | 10 |
CPU_CS|---------- | REG NOT INIT |
BAD PCMS CODE |
+-------------------------------------------------------------------+
Crystal 1 :
+-------------------------------------------------------------------+
| hw type
|
|----|----|------------|-----------|--------------|-----------------|
| 1 | 0 |
GD|---------- |
IN SERVICE |
BAD PCMS CODE |
| 1 | 2 |
PRA T2|---------- |
IN SERVICE |
NO PCMS CODE |
| 1 | 3 |
MIX484|---------- |
IN SERVICE |
BAD PCMS CODE |
| 1 | 4 |
UAI 4|---------- |
IN SERVICE |
BAD PCMS CODE |
| 1 | 6 |
UAI 8|---------- | REG NOT INIT |
BAD PCMS CODE |
| 1 | 7 |
PRA T2|---------- |
IN SERVICE |
NO PCMS CODE |
+-------------------------------------------------------------------+

Mediant 1000 MSBG plateform:
.
See Appendix A: Application description for details
Lync Server 2010:
.
Media Bypass activated
Refer activated or not
3.3 Software configuration
Alcatel Communication Platform: OmniPCX Enterprise R9.1 i1.605.21 or R10

j1.410.34c
Mediant 1000 MSBG: M1000_MSBG_SIP_6.20A.034.004
Microsoft Lynk RTM: Release 2010 4.0.7577.0

4 Summary of test results

4.1 Summary of main functions supported
Basic calls internal/PSTN from/to Lync set (other)/OXE sets (NOE, IP, Analog).
Voice compression codec: G.711/A, G.729A (see note on call limitation).
Call server spatial redundancy support with DNS mechanism
AudioCodes redundancy
Three party conferences initiated from Lync set and OXE set.
4.2 Summary of problems

4.2.1 OXE
None
4.2.2 Audicodes/Lync
None
4.3 Summary of limitations

4.3.1 OXE
None
4.3.2 AudioCodes/Lync
Call from an OXE set to a Lync client: Lync doesnt update the calling name.
Call from an OXE set to a Lync client: The trunk name is displayed on the OXE set instead
of the Lync client name
Transfers from Lync: Lync doesnt update transferee information after a transfer
(unattended or attended transfer).
Semi attended or attended transfers from OXE: OXE sends the user information to
AudioCodes gateway (in REFER or REINVITE messages). However Lync display is not
updated. Transfer is OK.
4.4 Notes, remarks
In all Mediant GW and Lync configuration, each Mediant 1000 MSBG has to be coupled to a
Mediation Server.
Microsoft Lync user does not call a number with a character: *, #,$
E-911 is out of the scope of this certification.
AudioCodes gateway Median 1000 supports 300 simultaneous sessions per CPU or 120
simultaneous sessions in case of audio transcoding (for example law change : G711a>G711, codec change : G729->G711. Session equal for one lag, e.g., 1 IP two IP calls
equal to two lags)

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.
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.
Comment: 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 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
. action 1
. action 2
Result
OK
OK
3
4
. action 3
. action 4
Comment
The application
waits for PBX timer
or phone set hangs
up
N/A
Relevant only if the
CTI interface is a
direct CSTA link
. action 5
NOK
No indication, no
error message

6 Testing
6.1 Basics Outgoing calls: OXE to Lync
6.1.1 Calls from an OXE user set to a Lync user
6.1.1.1 Test objectives
The calls are generated to several numbers corresponding to users on the Lync platform.
Called party can be in different states: free, busy, Out of service, do not disturb.
Points to be checked: tones, voice during the conversation, display (on caller and called party),
hang-up phase.
Note: dialling will be based on direct dialling number but also using programming numbers on the
phone.
6.1.1.2 Test procedure

Step
1
2
Action
Call to wrong number
Call to free user
Result
OK
OK_but
Call in conversation; DTMF

reception ; Calling / caller line
identity ; display
OK_but
4
5
6
7
8
Call to busy user (mono-line /

multi-line)
Call to user in Out of Service
state
Call to user in Do not Disturb
state
Call to Data terminal
Call to forwarded user (locally)
(immediate forward)
OXE user1 call Lync user1 who is in
immediate forward to Lync user2.
External call (T2) to Lync user.

Check that external call back
number is shown correctly.
Comment
N/A
OK
OK
N/A
OK_but
No called name display on OXE phone,

because Lync does not send the user
name to the AudioCodes (only the
number). The trunk name with the
called number is displayed on OXE
phone. On Lync user, the display name
is OK.
No called name display on OXE phone,
because Lync does not send the user
name to the AudioCodes (only the
number). The trunk name with the
called number is displayed on OXE
phone. On Lync user, the display name
is OK.
Lync user is never in busy mode
no answer display on caller:
Temporary Unavailable in the trace
no answer display on caller:
Temporary Unavailble in the trace
OXE phone displays the forwarded Lync
user information, not final Lync
destination.
AudioCodes gateway doesnt receive
the final destination information from
Microsoft Lync.
OK

10
11
12
Call back feature :

OXE user1 call Lync user1 and hung
up. Lync user1 can press call back
to call OXE user1
No answer of the called party
(Forward no reply)
OXE phone call Lync user and Lync
user does not take the call and
after some minute the call is
forwarded to another Lync user
Call from UA/TDM to Lync user
NYT
OK_but
OXE phone displays the forwarded Lync

user information, not final Lync
destination.
AudioCodes gateway doesnt receive
the final destination information from
Microsoft Lync.
OK
6.2 Incoming calls: Lync => OXE

6.2.1 Reception of calls
Call to unknown numbers must be rejected.
Calls will be generated using the numbers or the name of the SIP user.
SIP terminal will be called in different state: free, busy, Out of service.
Points to be checked: tones, voice during the conversation, display (on caller and called party),
hang-up phase.

Step
Action
Call to wrong number
Call to free user (check Calling / caller

line identity ; display, DTMF)
Call to busy user (mono-line / multiline)(Local/Network)
Call to user in Out of Service state
(Local/Network)
Call to user in Do not Disturb state
3
4
5
6
7
8
10
12
Call to Data terminal

Call to forwarded user (immediate
forward)
Lync user1 call OXE user1 who is in
immediate forward to OXE user2.
(Local/Network)
Call Back feature

Lync user Calls to external number (via
T2 loopback)
(Check ring back tone, called party
display)
No answer of the called party
Result
OK
OK_but
OK
OK
OK
N/A
OK_but
Display Unavailable or maybe

offline on Lync user
Lync doesnt update the OXE
username
Display Is in another call on
Lync user
Display Unavailable or maybe
offline on Lync user
Display Did not answer
Lync phone displays the
forwarded OXE user information,
not final OXE destination.
AudioCodes gateway doesnt
send the final destination
information to Microsoft Lync (it
changes the call flow).
OK
OK
OK

Comment
Ring back tone and audio Ok

Lync user plays a local ring back
tone first; and update this ring
back tone by the OXE one.
Configuration from OXE to
release after a few minutes
Lync user 1 calls IP Phone which is on

forward on busy to Lync user 2 (prefix
52) (Local/Network)
13
Lync user 1 calIs IP Phone which is on

forward on no reply to Lync user 2
(prefix 53) (Local/Network)
14
Lync user1 calIs IP Phone which is in

immediate forward mode to Lync user2
(51) (Local/Network)
15
Oxe Phone call Lync user then Oxe

phone display secret identity
Lync user put OXE phone on hold
16
17

information to Microsoft Lync.
Identity secret display on OXE
phone
Local music on hold played on
Lync client
OK_but
OK_but
OK_but
OK
OK_but
6.3 Features during conversation

6.3.1 Hold, Consultation call and broker call
During conversation, waiting and consultation call are provided and must be checked.
In addition, a second call must be generated in order to check that right tones are generated on
Lync user.
Step
Action
A=Lync user, B=IPPhone1, C=IPPhone2

(Local/Network)
1.1 Hold state request
Lync user->IPPhone1, Lync user on hold
1.2
Consultation call request
IPPhone1->IPPhone2
1.3
Broker request
IPPhone1->Lync user, IPPhone2 on hold
2 A=PSTN, B=IPPhone1, C=Lync user
(Local/Network)
2.1 Hold state request
PSTN->IPPhone1, PSTN on hold
2.2
IPPhone1->Lync user
Result
Comment
OK_but
Local music on hold played

on Lync client
OK
OK
OK
OK

2.3
3
3.1
3.2
3.3
4
4.1
4.2
4. 3
5
5.1
5.2
5.3
6
6.1
6.2
6.3
7
7.1
7.2
7.3
8
8.1
8.2
8.3
9
9.1
Broker request
IPPhone1->PSTN, Lync user on hold
A=PSTN, B= Lync user , C= IPPhone1
(Local/Network)
Hold state request
PSTN-> Lync user,
PSTN on hold
Lync user-> IPPhone1
Broker request
Lync user->PSTN, IPPhone1 on hold
A=IPPhone1, B=IPPhone2, C=Lync user
(Local/Network)
Hold state request
IPPhone1->IPPhone2, IPPhone1 on hold
IPPhone2->Lync user
Broker request
IPPhone2->IPPhone1, Lync user on hold
A=IPPhone1, B=Lync user, C=IPPhone2
(Local/Network)
Hold state request
IPPhone1->Lync user, IPPhone1 on hold
Lync user->IPPhone2
Broker request
Lync user->IPPhone1, IPPhone2 on hold
A=Lync user1, B=IPPhone, C= Lync user2
(Local/Network)
Hold state request
Lync user1->IPPhone, Lync user1 on hold
IPPhone->Lync user2
Broker request
IPPhone->Lync user1, Lync user2 on hold
A=Lync user1, B=IPPhone, C=UA/TDM set
(Local/Network)
Hold state request
IPPhone->Lync user1, IPPhone on hold
Lync user1->UA/TDM set
Broker request
Lync user1->IPPhone, UA/TDM set on hold
A=Lync user1, B=UA/TDM set 1 C=UA/TDM
set2 (Local/Network)
Hold state request
IUA/TDM set1 ->Lync user1, UA/TDM set1 on
hold
Lync user1 > UA/TDM set2
Broker request
Lync user1->UA/TDM set 1
UA/TDM set2 on hold
A=IPPhone, B=Lync user1, C=Lync user2
(Local/Network)
Hold state request
IPPhone->Lync user1, IPPhone on hold
OK_but
OK
OK
Ok
OK
OK
OK_but

on Lync client
OK
OK
OK
OK_but

on Lync client
OK
OK_but

on Lync client
OK_but

on Lync client
OK
Ok
OK
OK
OK
OK


on Lync client
9.2
9.3

Lync user1->Lync user2
Broker request
Lync user1->IPPhone, Lync user2 on hold
OK
OK_but

on Lync client
6.3.2 Transfer call

During the consultation call step, the transfer service can be requested and must be tested.
Several transfer services exist: blind transfer, supervised transfer and busy transfer.
Audio, tones and display must be checked.
Tests are performed using all possible combinations of legacy (IPPHONE) and Lync sets.

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.
There are three sorts of transfers in the SIP world:
Unattended Transfer or Blind Transfer: The Transferor provides 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.
Note: Unattended Transfer is not provided by the OXE, but is supported from external
parties.
Semi-Attended Transfer or Early Attended Transfer or Transfer on ringing:

o A (Transferee) calls B (Transferor).
o B (Transferor) calls C (Transfer Target). A is on hold during this phase. C is in ringing
state (does not pick up the call).
o B executes the transfer. B drops out of the communication. A is now in contact with
C, in ringing state. When C picks up the call it is in conversation with A.
Attended Transfer or Consultative Transfer or Transfer in conversation:

1. A (Transferee) calls B (Transferor).
2. B (Transferor) calls C (Transfer Target). A is on hold during this phase. C picks up the
call and goes in conversation with B.
3. B executes the transfer. B drops out of the communication. A is now in conversation
with C.
In the below tables, OXE means a proprietary OXE (Z/UA/IP) set. The calls can be local/network
calls.

Check the transfer for two configuration possibilities on Lync (with or without REFER). For blind
transfer check that the transferred call can be taken back from the transferee in case of no
answer or wrong number dialed.
Unattended Transfer (Blind)
Test Action
Result
A
B
Transf Transf
eree
eror
1
C
Transfer
Target
LYNC
OXE
OXE
NA
OXE
LYNC
OXE
OK_but
OXE
OXE
LYNC
NA
OXE
LYNC
LYNC
OK_but
LYNC
OXE
LYNC
NA
LYNC
LYNC
OXE
OK_but
3
4
Comment
Unattended transfers not available from an

OXE user.
No display update on A.
AudioCodes gateway doesnt receive the final
destination information from Microsoft Lync.
Music on hold is played by OXE.
If the transfer target doesnt answer the call
can not be taken back from the Lync client.
If the transfer target doesnt no exist the call
can be taken back from the Lync client.
OXE user.
Music on hold is played by OXE.
OXE user.
Semi attended Transfer (On Ringing)
Test Action
Result
A
B
Transf Transf
eree
eror
C
Transfer
Target
2
3
4
Comment
LYNC
OXE
OXE
OK_but
OXE
LYNC
OXE
NA
OXE
OXE
LYNC
OK
OXE
LYNC
LYNC
NA
AudioCodes gateway receives the REFER from
OXE, but doesnt send to Lync.
Semi-attended transfers not available from a
Lync user.
Lync user.

5
6
LYNC
OXE
LYNC
OK
LYNC
LYNC
OXE
NA

Lync user.
Attended Transfer (in conversation)
Test Action
Result
A
B
Transf Transf
eree
eror
1
2
Comment
C
Transfer
Target
LYNC
OXE
OXE
OK
OXE
LYNC
OXE
OK_but
OXE
OXE
LYNC
OK_but
OXE
LYNC
LYNC
OK_but
LYNC
OXE
LYNC
OK_but
LYNC
LYNC
OXE
OK_but
AudioCodes gateway receives the REINVITE
from OXE, but doesnt send to Lync.
AudioCodes gateway receives the REINVITE
from OXE, but doesnt send to Lync.

6.3.3 Conference
During the consultation call step, the conference is provided and must be tested.
Programmed conference and 3 steps conferences have to be checked by analyzing the audio and
display on each user.

We use the following scenario, terminology and notation:
We start with A in conversation with B. (A->B)
A places B on hold. B should hear hold tone.
A calls C while B is on hold. C rings and goes off-hook.
A activates conference.
A, B, C should be in communication now.
Step
Action
3 steps conference request

PSTN->IPPHone->Lync user
Step1 then IPPhone leaves the
conference
Step 1 then Lync user leaves the
conference
Step 1 then PSTN leaves the
conference
3 steps conference request
PSTN->Lync user-> IPPHone
Step 5 then IPPhone leaves the
conference
Step 5 then Lync user leaves the
conference
Step 5 then PSTN leaves the
conference
Stay in a conference for long period to
check keep alive mechanisms.
2
3
4
5
6
7
8
9
Result
OK
OK
OK
OK
OK
OK
OK
OK
OK

Comment
6.4 Voice Mail

6.4.1 Test Objectives
Note: It has to be defined which Voicemail system is used for the interoperability tests:
Option 1: Voice mail is Exchange 2010 for all users: Not tested (Should be OK from
Option 2: Each system is served by its own Voice Mail: OK
Voice Mail notification, consultation and password modification must be checked.
MWI (Message Waiting Indication) has to be checked.
6.4.2 Test Results

Test Action
1 IPTouch phone leave as a VoiceMail
message for the Lync user
Check that MWI is OK on Lync user
2 Message consultation by Lync user
3 Password modification by Lync user
4 Lync user call to a OXE user
forwarded to Voice Mail (4645)
5 Message consultation by OXE user
Result
Comment
Not
tested
Not
tested
Not
tested
OK
OK
6.5 Attendant
An attendant console is defined on the system. Call going to and coming from the attendant
console are tested.
6.5.2 Test Results

Test Action
1 Lync user calls attendant (using
attendant calls prefix 8)
2 Attendant calls Lync user
3 Attendant calls Lync user then
Attendant calls OXE user
Attendant transfer Lync user To OXE
user
attendant calls prefix 8),
attendant transfers on ringing to OXE
set.
Result
Comment
OK
OK
OK
OK


attendant calls prefix 8),
attendant transfers in conversation
to OXE set,
6 OXE set calls to attendant (using
attendant calsl prefix **8),
attendant transfers in conversation
to Lync user.
7 External user(T2) calls Attendant
Attendant calls Lync user
Attendant transfer on ringing
External user(T2) to Lync user
8 Lync user calls Attendant
Attendant calls External user(T2)
Attendant transfer on conversation
Lync user to External user(T2)
OK
OK
OK_but
But the display on Lync user is not

updated.
OK_but
But the display on Lync user is not

updated.
6.6 Defense / Recovery

Check the reaction in case of a PBX reboot, switch-over or link failure.
AudioCodes gateway redundancy is also tested.
6.6.2 Test Results

Step
1
2
3
Action
Temporary Data Network Link down with
the PBX and Mediant 1000 MSBG
Spatial redundancy IP Method : CPU
switchover with SIP communication
Spatial redundancy DNS method
(delegation on a third party DNS server) :
CPU switchover without SIP communication
Spatial redundancy DNS method : CPU
switchover with SIP communication
Switchover to Passive Call Server (PCS). (IP
link to main/stdby OXE call servers down)
Switchover to a backup AudioCodes

gateway. Stop the main AudioCodes
gateway; verify that a call is possible with
the backup AudioCodes gateway.
Result
Comment
OK
Existing calls are stopped.

Establishing new call is possible
when the link is reestablished.
OK
OK
OK
N/A
OK

The gateway status is checked

every 10 seconds. After this
period, you can establish a new
call.
7 Appendix A : AudioCodes Mediant 1000 MSBG

Configuration
7.1 Getting Started
This section describes how to navigate in the Mediant 1000 MSBG Web server
navigation tree.
When navigating in the Navigation tree, you can view listed menus and submenus in
either an expanded or contracted view. This is relevant when using the configuration
tabs (Configuration, Maintenance, and Status & Diagnostics) on the Navigation bar.
The Navigation tree menu can be displayed in one of two views:
Basic: displays only commonly used menus
Full: displays all the menus pertaining to a configuration tab.
The advantage of the Basic view is that it prevents "cluttering" the Navigation tree with
menus that may not be required. Therefore, a Basic view allows you to easily locate the
required menus.
To toggle between Full and Basic view:

Select the Basic option (located below the Navigation bar) to display a reduced
menu tree; select the Full option to display all the menus. By default, the Basic
option is selected.
Figure 7-1: Navigation Tree in Basic and Full View
For more information, see the Mediant 1000 MSBG Users manual.

7.2 Configuration Procedure

This section describes the Mediant 1000 MSBG Configuration procedure.
7.2.1 Configure IP Address

To configure IP-Address:
1.
Open the 'IP Settings' page (Configuration tab > VoIP menu > Network > IP
Settings).
Figure 7-2: IP Settings
2.
Set the following parameters:

IP-Address: <Gateway IP-Address> (e.g., 10.15.7.131).
Prefix Length: The Subnet Mask in bits (e.g., 16 for 255.255.0.0).
Gateway: <Gateway Default Gateway> (e.g., 10.15.7.130).
7.2.2 VoIP DNS Settings

To configure VoIP DNS settings:
1.
Open the 'DNS Settings' page (Configuration tab > VoIP menu > DNS > DNS
Settings).
Figure 7-3: VoIP DNS Settings
2.

DNS Primary Server IP: <Primary DNS IP-Address> (e.g., 10.1.2.21).
DNS Secondary Server IP: <Secondary DNS IP-Address>.
7.2.3 Enable Applications

To configure Firewall Settings:
1.
Open the 'Applications Enabling' page (Configuration tab > VoIP menu >
Applications Enabling > Applications Enabling).
Figure 7-4: Applications Enabling

2.
Enable IP2IP Application: Enable.
Note: Reset with BURN to FLASH is Required.
7.2.4 SIP General Parameters

To configure SIP General Parameters:
1.
Open the 'Applications Enabling' page (Configuration tab > VoIP menu > SIP
Definitions > General Parameters).
Figure 7-5: General Parameters
2.
Enable Early Media: Enable.
SIP Transport Type: UDP.

Figure 7-6: General Parameters (Cont.)
3.
Set SIP Destination Port: 5060.
4.
Set Play Ringback Tone to Tel: Play Local Until Remote Media Arriving.
Figure 7-7: General Parameters (Cont.)
5.
Set Forking Handling Mode: Sequential handling.

7.2.5 DTMF & Dialing

To configure DTMF & Dialing:
1.
Open the 'DTMF & Dialing' page (Configuration tab > VoIP menu > GW and IP
to IP > DTMF and Supplementary > DTMF & Dialing).
Figure 7-8: DTMF & Dialing
2.
Set RFC 2833 Payload Type: 101.
7.2.6 Coders
To configure Coders:
1.
Open the 'Coders' page (Configuration tab > VoIP menu > Coders And
Profiles > Coders).
Figure 7-9: Coders
2.
Set the Coders G.711A-law and G.711U-law.
To configure Coders Group 1:

1.
Open the 'Coders Group Settings' page (Configuration tab > VoIP menu >
Coders And Profiles > Coders Group Settings).

Figure 7-10: Coders Group Settings 1
2.
In Coder Group ID 1 (for OXE side) set the Coders G.711A-law, G.711U-law
and G.729
7.2.7 Configure IP Profile

To configure IP Profile Settings 1:
1.
Open the 'IP Profile Settings' page (Configuration tab > VoIP menu > Coders
And Profiles > IP Profile Settings).

Figure 7-11: IP Profile Settings
2.
Set Copy Destination Number to Redirect Number: After Manipulation.
3.
Set Coder Group: Coder Group 1.
7.2.8 Configure IP Group Tables

To configure IP Group Table 1:
1.
Open the 'IP Group Table' page (Configuration tab > VoIP menu > Control
Network> IP Group Table).

Figure 7-12: IP Group Table 1
2.
Set Type: Server.
3.
Set Proxy Set ID: 1.
To configure IP Group Table 2:

1.
Open the 'IP Group Table' page (Configuration tab > VoIP menu > Control
Network> IP Group Table).

Figure 7-13: IP Group Table 2
2.
Set Type: Server.
3.
Set Proxy Set ID: 2.
4.
Set SIP Group Name: etesting.XmlForum.com.
5.
Set Outbound Message Manipulation Set: 1.
7.2.9 Configure Proxy Sets Tables

To configure Proxy Sets Table 1:
1.
Open the 'Proxy Sets Table' page (Configuration tab > VoIP menu > Control
Network> Proxy Sets Table).

Figure 7-14: Proxy Sets Table 1
2.
Configure Microsoft Lync IP-Address or FQDN and Destination Port (e.g.,

10.15.9.11:5068).
3.
Transport Type: TCP or TLS (e.g., TCP).
Enable Proxy Keep Alive: Using Options.
Proxy Load Balancing Method: Round Robin.
Is Proxy Hot Swap: Yes.
To configure Proxy Sets Table 2:

1.
Open the 'Proxy Sets Table' page (Configuration tab > VoIP menu > Control
Network> Proxy Sets Table).

Figure 7-15: Proxy Sets Table 2
2.
Configure Alcatel Omni IP-Address or FQDN and Destination Port (e.g.,

etesting.XmlForum.com).
3.
Enable Proxy Keep Alive: Using Options.
Proxy Load Balancing Method: Disable.
Is Proxy Hot Swap: Yes.
7.2.10 Configure Proxy & Registration

To configure Proxy & Registration:
1.
Open the 'Proxy & Registration' page (Configuration tab > VoIP menu > SIP
Definitions > Proxy & Registration).

Figure 7-16: Proxy & Registration
2.
Use Default Proxy: No.
Redundant Routing Mode: Proxy.
Gateway Name: Gateway FQDN Name (e.g., ALE-GW.Lync.local).
7.2.11 Configure IP Media

To configure IP Media Settings:
1.
Open the 'IP Media Settings' page (Configuration tab > VoIP menu > IP Media
> IP Media Settings).

Figure 7-17: IP Media Settings
2.
Set Number of Media Channels: 120.
7.2.12 Configure Routing

To configure IP to Trunk Group Routing:
1.
Open the 'IP to Trunk Group Routing' page (Configuration tab > VoIP menu >
GW and IP to IP > Routing > IP to Trunk Group Routing).
Figure 7-18: IP to Trunk Group Routing (example)
2.
Configure according to the routing topology.
3.
In the example above, all the calls from IP 10.1.2.16 (the Lync server) marks as
IP2IP calls (-1) with Source IPGroup ID = 1). All the other calls marks as IP2IP
calls (-1) with the default Source IPGroup ID.
To configure Tel to IP Routing:

1.
Open the 'Tel to IP Routing' page (Configuration tab > VoIP menu > GW and
IP to IP > Routing > Tel to IP Routing).
Figure 7-19: Tel to IP Routing (example)

2.
Configure according to the organization routing topology.
3.
In the example above, all the calls from Source IPGroup ID = 1 route to IP Group
2 using IP Profile 1 (Lync call sent to the OmniPCX Soft Switch). All the other
calls route to the Lync server.
7.2.13 Configure Manipulation

To configure Manipulations:
1.
Open the 'Dest Number IP->Tel' page (Configuration tab > VoIP menu > GW
and IP to IP > Manipulations > Dest Number IP->Tel).
Figure 7-20: Dest Number IP->Tel (example)
2.
Configure according to the organization topology.
3.
In the example above:

Line 1 If the calls come from the Lync server IP-Address and the number
starting with +333906, we stripped the first 7 digits.
Line 2 and Line 3 If the calls come from Alcatel OmniPCX and the number
exactly 5 digits that starting with 7 add +333906 to prefix.
4.
Open the 'Source Number IP->Tel' page (Configuration tab > VoIP menu > GW
and IP to IP > Manipulations > Source Number IP->Tel).
Figure 7-21: Source Number IP->Tel (example)
5.
Configure according to the organization topology.
6.
In the example above:

Line 1 and Line 2 If the calls come from Alcatel OmniPCX and the number
exactly 5 digits that starting with 7 add +333906 to prefix.
Line 3 If the calls come from the Lync server IP-Address and the number
starting with +333906, we stripped the first 7 digits.
7.2.14 Configure Reasons for Alternative Routing

To configure Reasons for Alternative Routing:
1.
Open the 'Dest Number IP->Tel' page (Configuration tab > VoIP menu > GW
and IP to IP > Routing > Reasons for Alternative Routing).

Figure 7-22: Reasons for Alternative Routing
2.
Set Tel to IP Reasons 1: 503.
7.2.15 Header Manipulation for Voice Mail

To configure Header Manipulation for Voice Mail you need add it in the ini and
reload the GW:
When call forward from Lync sever to Alcatel Voice Mail the Invite should be contain
the following SIP Headers:
URI = Voice Mail extension number.
TO = The Redirect Lync Extension.
1.
Open the 'Configuration File' page (maintenance tab > Software Update menu
> Configuration File).
Figure 7-23: Configuration File
2.
Save the INI file.

3.
Add the following line:
[ MessageManipulations ]
MessageManipulations 1 = 1, invite, "Header.To.url.user == '31300'", Header.TO.url.user, 2, Header.Diversion.url.user, 0;
MessageManipulations 2 = 1, Invite, "Header.To.url.user == '31301'", Header.TO.url.user, 2, Header.Diversion.url.user, 0;
[ \MessageManipulations ]
4.
Replace 31300 and 31301 with the Voice Mail your Voice mail extensions.
5.
After you add the Voice Mail lines, save the INI file and reload it to the GW
(require restart).
Note: This example contains only two Voice Mail extensions, if required,
you can add more extension lines.
.
Header manipulation available on the Gateway web GUI, only if the SBC
application is enabled
1.
Enable SBC Application = Enable.
Open the 'Configuration File' page (Configuration tab > SBC

menu > Manipulations SBC menu > Message).
2.
7.2.16 Admin Page Parameters

To configure Parameters in the AdminPage:
1.
Open the 'AdminPage' page (HTTP://<GW IP>/AdminPage).

Figure 7-24: AdminPage
2.
Configure the following parameters:
ENABLEEARLY183 = 1.
FAKERETRYAFTER = 60.
GWINBOUNDMANIPULATIONSET = 1
GWOUTBOUNDMANIPULATIONSET = 2
7.2.17 Reset the Gateway

After you have completed the gateway configuration as described in the steps above,
burn the configuration to the gateways flash memory and reset the gateway.
Click the Reset button to burn the configuration to flash and reset the gateway (ensure
that the Burn to FLASH field is set to Yes).
Figure 7-25: Reset the GW
Note: Reset with BURN to FLASH is Required.

7.3 AudioCodes ini files
MSBG_BOARD.ini

8 Appendix B: Alcatel-Lucent Communication

Platform: configuration requirements
SIP trunking management via ARS routing
Node1 Configuration:
Field
Value
Private SIP Trunk Group which will be used for outgoing and incoming (or not) calls
Trunk group
Trunk Groups
Trunk Group ID
Trunk Group Type
Trunk Group name
Remote Network
Node number
Q931 signal variant
(2)
37
T2
SIPTRK
15
1
ABC-F (private sip trunk)
T2 Specification
Overlap dialing
Virtual accesses
Trunk Group Id
Number of SIP Access
Trunk group
SIP
No
Trunk Groups/Trunk Group/Virtual access for SIP
37
2 (ie 2*62 accesses)
Trunk Groups
Trunk Group ID
Trunk Group Type
Trunk Group name
Remote Network
Node number
Q931 signal variant
(2)
38
T2
SIPTRK2
15
1
ABC-F (private sip trunk)
T2 Specification
Overlap dialing
Virtual accesses
Trunk Group Id
Number of SIP Access
SIP
No
Trunk Groups/Trunk Group/Virtual access for SIP
38
2 (ie 2*62 accesses)
SIP management for incoming calls (from remote SIP domain)

SIP External Gateway configuration used to reach Operator SIP GW
SIP Ext Gateway
SIP/SIP Ext Gateway
SIP External Gateway Id
1
SIP Remote domain
10.200.56.2
SIP Port Number
5060
SIP Transport Type
TCP
Trunk group number
37
Supervision timer
10 (used for AudioCodes gateway redundancy)
RFC 3325 supported by the distant True
SIP Ext Gateway
SIP/SIP Ext Gateway
SIP External Gateway Id

SIP Remote domain
SIP Port Number
SIP Transport Type
Trunk group number
Supervision timer
RFC 3325 supported by the distant
2
10.200.56.3
5060
TCP
38
10 (used for AudioCodes gateway redundancy)
True
Management for outgoing calls via SIP Trunk Group

ARS routing used to rout outgoing calls to external SIP GW through Public SIP TG
Dialing command table (DCT)
Translator/Automatic Route Selection/Dialing Command Table
Table Id
Carrier Reference
Command
Associated Ext SIP gateway
1
0
I
1
Table Id
Carrier Reference
Command
ARS route list
2(4)
0
I
2
Translator/Automatic Route Selection/ARS Route list
(4)
4 (5)
SIP ARS
Translator/Automatic Route Selection/ARS Route list/ARS Route
4
(6)
1
SIP ARS route 1
37
0
97239764
1
Dependant on Trunk Group Type
Speech
4
(6)
2
SIP ARS route 2
38
0
97239764
2
Speech
Translator/Automatic Route Selection/ARS Route list/Time-based
Time-based route list
route list
ARS Route list
4
Time-based Route List ID
1
Route Number
1 and 2
Barring / Discrimination (call restrictions): used to give to caller the right to call the external
dialled number
Translator/External Dialing Plan/Dialing (Numbering)
Real Discriminator
Discriminator
ARS Route list
Name
ARS route
ARS Route list
Route
Name
Trunk Group
No.Digits To Be Removed
Digits To Add
Numbering Command Tabl. ID
Protocol Type
Quality
ARS route
ARS Route list
Route
Name
Trunk Group
Digits To Add
Protocol Type
Quality

Discriminator n
Name
Discriminator rule
Discriminator n
Call number
Area number
ARS route list number
Number of digits
Logical discriminator
Entity Number
Discriminator 0x (9)
ARS prefix
Number
Prefix Meaning
Discriminator n
Public Network COS
Public Network COS
Area identifier
Public Access Rights
Night
Day
Mode 1
Mode 2
(7)
2
SIP
Discriminator/Discriminator rule
2
2
(8)
1
4
3
Entities/Discriminator Selector
1
2
Translator/Prefix plan
4982
ARS Prof.Trg Grp Seizure
2
Classes of Service / Access COS / Public Access COS
1
1
1
1
To get the display name of Lync user on OXE user when the OXE user calls the Lync user or when
Lync user calls the OXE user, You must make the configuration bellow:
1. System -> Other System Parameter ->Descend Hierarchy -> External signaling Parameter > review Modify -> calling name presentation -> put it in True.

2. Check the user Lync Phone feature Cos then Go to:

Classes of Service -> Phone Feature Cos -> Calling name display (CNIP/I CNAM) put it in 1

For ABC-IP Network Test, you must configure the ARS route in Node2.
Node2 Configuration:
Field
Value
Management for ABC-F Network calls via SIP Trunk Group

ARS routing used to rout outgoing calls to external SIP GW through Public SIP TG
Table Id
Carrier Reference
Command
1 (4)
0
I
1
Table Id
Carrier Reference
Command
ARS route list
1 (4)
0
I
2
Translator/Automatic Route Selection/ARS Route list
ARS Route list

Name
ARS route
ARS Route list
Route
Name
Trunk Group
Digits To Add
Protocol Type
Quality
ARS route
ARS Route list
Route
Name
Trunk Group
Digits To Add
Protocol Type
Quality
(5)
4
SIP ARS
4
1 (6)
SIP ARS route 1
37
0
97239764
1
Speech
4
2 (6)
SIP ARS route 2
32
0
97239764
2
Speech
Translator/Automatic Route Selection/ARS Route list/Time-based
Time-based route list
route list
ARS Route list
4
Time-based Route List ID
1
Route Number
1 and 2
Barring / Discrimination (call restrictions): used to give to caller the right to call the external
dialled number
Real Discriminator
Translator/External Dialing Plan/Dialing (Numbering) Discriminator

Discriminator n
Name
Discriminator rule
Discriminator n
Call number
Area number
ARS route list number
Number of digits
Logical discriminator
Entity Number
Discriminator 0x (9)
ARS prefix
Number
Prefix Meaning
Discriminator n
Public Network COS
Public Network COS
Area identifier
Public Access Rights
Night
Day
Mode 1
Mode 2
2 (7)
SIP
Discriminator/Discriminator rule
2
2
(8)
1
4
3
Entities/Discriminator Selector
1
2
Translator/Prefix plan
4982
ARS Prof.Trg Grp Seizure
2
Classes of Service / Access COS / Public Access COS
1
1
1
1
Note: All the IP Parameter codec in node 2 must be in G711.

To make a Call to ARS via T2 Loopback, You must make the configuration bellow, to allow OXE to
make a transit calls. In this case: we are getting call from:
One trunk (T2) and routed to another trunk (SIP).
mgr-->Class of service-->Connection COS-->5
By default COS 5 is 0 we need to modify this as 1

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

9 Appendix C: 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.

10 Appendix D: AAPP program

10.1 Alcatel-Lucent Application Partner Program (AAPP)
Complete e-business solutions at your disposal
The 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 thirdparty applications that complement Alcatel-Lucent's Omni-based products. Alcatel-Lucent
facilitates market access for compliant applications.
The Alcatel-Lucent 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 labelled
Alcatel-Lucent Compliant Application, come from every area of voice and data
communications.
The Allcatel-Lucent 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 Application Partner, you can access the AAPP website at this URL:
http://applicationpartner.alcatel-lucent.com
10.2 Alcatel-Lucent.com
You can access the Alcatel-Lucent website at this URL: http://www.Alcatel-Lucent.com/

11 Appendix E: AAPP Escalation process

11.1 Introduction
The purpose of this appendix is to define the split of responsibilities and the escalation process to
be applied by the Alcatel-Lucent Business Partners when facing a problem with a solution
involving an Alcatel-Lucent platform and a Third-Party application with or without a valid
Alcatel-Lucent Inter-Working Report.
If a problem occurs on an installation involving Alcatel-Lucent platforms and a certified product
or application, both parties, Alcatel-Lucent and the Application Partner, are engaged as follows:
(*) The Application Partner Business Partner can be a Third-Party company or the AlcatelLucent Business Partner itself

11.2 Escalation in case of certified application/products

The Alcatel-Lucent support will be limited to applications with a valid Inter-Working Report
(IWR). Known problems or remarks mentioned in the IWR will not be taken into account.
A valid IWR means an official IWR exists which is posted on the Alcatel-Lucent Enterprise
Business Portal and mentions the same release/version of the software of both parties as those
of the current customer installation (Or an official agreement between Alcatel-Lucent and the
Third-Party exists to support the customer installation if the release/version doesnt match
those mentioned in the latest IWR ).
If there is an interworking issue, both 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 ALU Business Partner to the AlcatelLucent Support Center using the standard process: open a ticket (eService Request eSR)
Case 2: the responsibility can be established 100% on Application Partner side.
In that case, the problem must be escalated directly to the Application Partner by
opening a ticket through the Partner Hotline. In general, the process to be applied for the
Application Partner is described in the IWR.
Case 3: the responsibility can not be established.
In that case the following process applies:
The Application Partner shall be contacted first by the Business Partner (responsible
for the application, see figure in previous page) for an analysis of the problem.
The Alcatel-Lucent Business Partner will escalate the problem to the Alcatel-Lucent
Support Center only if the Application Partner has demonstrated with traces a
problem on the Alcatel-Lucent side or if the Application Partner (not the Business
Partner) needs the involvement of Alcatel-Lucent.
In that case, the Alcatel-Lucent Business Partner must provide the reference of the Case
Number on the Application Partner side. The Application Partner must provide to AlcatelLucent 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 the Application Partner side are insufficient or do no exist.
IMPORTANT NOTE 1: 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 the AAPP (Url:
https://private.applicationpartner.alcatel-lucent.com) or Enterprise Business Portal (Url:
Enterprise Business Portal) web sites.
IMPORTANT NOTE 2: Involvement of the Alcatel-Lucent Business Partner is mandatory, the
access to the Alcatel-Lucent platform (remote access, login/password) being the Business Partner
responsibility.

11.3 Escalation in case of non-certified application/product

If an Alcatel-Lucent Business Partner escalates an issue where a 3rd party application is involved
and the following conditions apply:
1. no IWR exist (not available on the Enterprise Business Portal for Business Partners or on
the Alcatel-Lucent Application Partner web site) ,
2. Or the 3rd party company is referenced as AAPP participant but with no existing IWR,
3. Or the existing IWR is available but the release/version of the both parties (AlcatelLucent and 3rd-party) are not the same than those currently deployed at the customer
site (see exception in Note 2).
In this case, the only responsibility of the Alcatel-Lucent Technical Support is to verify that the
Alcatel-Lucent platform is correctly installed and configured for a standard use and that the
Alcatel-Lucent equipments perform as expected. If thats the case, Alcatel-Lucent will be forced
to close the case.
If the Alcatel-Lucent Business Partner, the customer or the 3rd party company need additional and
specific involvement from Alcatel-Lucent, there are two options:
Either request a quote for specific investigation and diagnosis (with no agreement to
fix the issue),
Or the AAPP program process is followed to officially certify the 3rd party
application/product.
For both options, just send the request to the AAPP team (by opening an e-SR).
IMPORTANT NOTE 1: Even if the 3rd party company is able to demonstrate the issue is on the
Alcatel-Lucent side, there is no obligation from Alcatel-Lucent to fix it (there is no official IWR
established between the two parties).
IMPORTANT NOTE 2: For case 3, Alcatel-Lucent and the Third-Party company may decide to
provide a document specifying the possible extension of the IWR by mentioning the list of
releases/versions officially supported. (Another way is to update an existing IWR with new
release/version compatibility).

11.4 Technical Support access

The Alcatel-Lucent Support Center is open 24 hours a day; 7 days a week:
e-Support from the Application Partner Web site (if registered Alcatel-Lucent Application
Partner): http://applicationpartner.alcatel-lucent.com
e-Support from the Alcatel-Lucent Business Partners Web site (if registered AlcatelLucent Business Partners): https://businessportal.alcatel-lucent.com click under Let us
help you the eService Request link
e-mail: Ebg_Global_Supportcenter@alcatel-lucent.com
Fax number: +33(0)3 69 20 85 85
Telephone numbers:
Alcatel-Lucent Business Partners Support Center for countries:
Country
Supported language
Toll free number
France
Belgium
French
Luxembourg
Germany
Austria
German
Switzerland
United Kingdom
Italy
Australia
Denmark
Ireland
Netherlands
+800-00200100
South Africa
Norway
Poland
English
Sweden
Czech Republic
Estonia
Finland
Greece
Slovakia
Portugal
Spain
Spanish
For other countries:

English answer :
French answer :
German answer :
Spanish answer :
+ 1 650 385 2193

+ 1 650 385 2196
+ 1 650 385 2197
+ 1 650 385 2198

END OF DOCUMENT


AudiocodesMediant1000-MSBG Lync OmniPCXEnterpriseR10.0 IWR Ed01

Diunggah oleh

Informasi Dokumen

Hak Cipta

Format Tersedia

Bagikan dokumen Ini

Bagikan atau Tanam Dokumen

Opsi Berbagi

Apakah menurut Anda dokumen ini bermanfaat?

Apakah konten ini tidak pantas?

Hak Cipta:

Format Tersedia

AudiocodesMediant1000-MSBG Lync OmniPCXEnterpriseR10.0 IWR Ed01

Diunggah oleh

Hak Cipta:

Format Tersedia

Alcatel-Lucent Application Partner Program

ALCATEL-LUCENT MAKES NO REPRESENTATIONS, WARRANTIES OR CONDITIONS WITH RESPECT TO

Alcatel-Lucent Application Partner Program Inter-working report

Edition 1 - page 1/54

Alcatel-Lucent Communication OmniPCX Enterprise

FOUARGE Bruno, RESIDORI Florian

Passed with restrictions

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

Alcatel-Lucent Application Partner Program Inter-working report

Edition 1 - page 2/54

Company Contact Information

1 Hayarden St. AirPort City

Alcatel-Lucent Application Partner Program Inter-working report

Edition 1 - page 3/54

Alcatel-Lucent Application Partner Program Inter-working report

Edition 1 - page 4/54

Alcatel-Lucent Application Partner Program Inter-working report

Edition 1 - page 5/54

The party being transferred to the transfer target

The party initiating the transfer

Blind or semi-attended transfer

Attended transfer or transfer on

Calling Name Identification Presentation

Calling Line Identification Restriction

Calling Name Identification Restriction

Connected Line Identification Presentation

Connected Name Identification Presentation

Connected Line Identification Restriction

Connected Name Identification Restriction

Customer Relationship Centre

SIP GW XXX Peripheral Gateway

SIP GW XXX Intelligent Call Manager

SIP GW XXX Call Manager

ALU Business Contact

Computer Telephony Integration

PSTN(analogique card on the OXE)

Public Switched Telephone Network

Alcatel-Lucent Application Partner Program Inter-working report

Edition 1 - page 6/54

Brief application description:

Figure 1 Global architecture

Alcatel-Lucent Application Partner Program Inter-working report

Edition 1 - page 7/54

Figure 2 Tests environnent

Alcatel-Lucent Application Partner Program Inter-working report

Edition 1 - page 8/54

3.2 Hardware configuration

OmniPCX Enterprise node1:

OmniPCX Enterprise Node2:

Alcatel-Lucent Application Partner Program Inter-working report

Edition 1 - page 9/54

Mediant 1000 MSBG plateform:

Lync Server 2010:

3.3 Software configuration

Alcatel Communication Platform: OmniPCX Enterprise R9.1 i1.605.21 or R10

Alcatel-Lucent Application Partner Program Inter-working report

Edition 1 - page 10/54

4 Summary of test results

4.2 Summary of problems

4.3 Summary of limitations

4.4 Notes, remarks

E-911 is out of the scope of this certification.

Alcatel-Lucent Application Partner Program Inter-working report

Edition 1 - page 11/54