audiocodesmediant1000-msbg lync omnipcxenterpriser10.0 iwr ed01

Alcatel-Lucent Application Partner Program Inter-working report Edition 1 - page 1/54 Copyright 2011 Alcatel-Lucent, All rights reserved

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.


Tests identification

Date of the tests October 2011

Alcatel-Lucents representative FOUARGE Bruno RESIDORI Florian

Partners representative AHARONOV Ofer

Alcatel-Lucent Communication Platform (OmniPCX 4400/Enterprise, OmniTouch, OmniPCX Office, ...)

OmniPCX Enterprise

Alcatel-Lucent compatibility release

R9.1(i1605.21) and R10 (j1410.34c)

Partners application version AudioCodes : 6.20A.034.004 Lync : R. 2010 4.0.7577.0

Environment (if it has a sense)

Author(s): FOUARGE Bruno, RESIDORI Florian Reviewer(s): 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.

ACD Business


Company Contact Information

Contact name: Ofer Aharonov Title: Interoperability Engineer Address 1: 1 Hayarden St. AirPort City Address 2: City: Lod State: Israel Zip: 70151 Country: Israel Country code: Phone: +972-3-9764210 Fax: +972-3-9764040 Web address: http://www.AudioCodes.com E-mail: [email protected]


TABLE OF CONTENTS 1 Introduction ............................................................................................ 6 1.1 GLOSSARY ............................................................................................... 6

2 Application information .............................................................................. 7 3 Tests environment .................................................................................... 8 3.1 GENERAL ARCHITECTURE ................................................................................. 8 3.2 HARDWARE CONFIGURATION .............................................................................. 9 3.3 SOFTWARE CONFIGURATION ............................................................................. 10

4 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 5 Test Scenarios ........................................................................................ 12 5.1 TEST PROCEDURE ....................................................................................... 12 5.2 RESULT TEMPLATE ...................................................................................... 12

6 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

7 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


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 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

Attended transfer or transfer on conversation

The transferor waits to be in conversation state with the target before completing the transfer

CLIP 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 SIP GW XXX Call Manager

CVP 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 OK_but Conference/Cancel Voice Mail

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:

+-------------------------------------------------------------------+

| Cr | cpl| cpl type | hw type | cpl state | coupler ID | |----|----|------------|-----------|--------------|-----------------| | 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 :

+-------------------------------------------------------------------+

| Cr | cpl| cpl type | hw type | cpl state | coupler ID | |----|----|------------|-----------|--------------|-----------------| | 0 | 6 | CPU_CS|---------- | IN SERVICE | BAD PCMS CODE | | 0 | 10 | CPU_CS|---------- | REG NOT INIT | BAD PCMS CODE | +-------------------------------------------------------------------+

Crystal 1 :

+-------------------------------------------------------------------+

| Cr | cpl| cpl type | hw type | cpl state | coupler ID | |----|----|------------|-----------|--------------|-----------------| | 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 Action Result Comment

1 . action 1 OK

2 . action 2

OK

The application waits for PBX timer or phone set hangs

up

3 . action 3 N/A

4 . action 4

Relevant only if the CTI interface is a direct CSTA link

5 . 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 Action Result

Comment

1 Call to wrong number OK 2 Call to free user

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.

3 Call in conversation; DTMF reception ; Calling / caller line identity ; display

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.

4 Call to busy user (mono-line / multi-line)

N/A Lync user is never in busy mode

5 Call to user in Out of Service state

OK no answer display on caller: Temporary Unavailable in the trace

6 Call to user in Do not Disturb state

OK no answer display on caller: Temporary Unavailble in the trace

7 Call to Data terminal N/A 8 Call to forwarded user (locally)

(immediate forward) OXE user1 call Lync user1 who is in immediate forward to Lync user2.

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.

9 External call (T2) to Lync user. Check that external call back number is shown correctly.

OK


10 Call back feature : OXE user1 call Lync user1 and hung up. Lync user1 can press call back to call OXE user1

NYT

11 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

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.

12 Call from UA/TDM to Lync user 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 Result

Comment

1 Call to wrong number OK

Display Unavailable or maybe offline on Lync user

2 Call to free user (check Calling / caller line identity ; display, DTMF)

OK_but Lync doesnt update the OXE username

3 Call to busy user (mono-line / multi-line)(Local/Network)

OK Display Is in another call on Lync user

4 Call to user in Out of Service state (Local/Network)

OK Display Unavailable or maybe offline on Lync user

5 Call to user in Do not Disturb state OK Display Did not answer

6 Call to Data terminal N/A 7 Call to forwarded user (immediate

forward) Lync user1 call OXE user1 who is in immediate forward to OXE user2. (Local/Network)

OK_but

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).

8 Call Back feature OK

10 Lync user Calls to external number (via T2 loopback) (Check ring back tone, called party display)

OK

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.

12 No answer of the called party

OK Configuration from OXE to release after a few minutes


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

OK_but

Lync phone displays the forwarded OXE user information, not final OXE destination. AudioCodes gateway doesnt send the final destination information to Microsoft Lync.

14 Lync user 1 calIs IP Phone which is on forward on no reply to Lync user 2 (prefix 53) (Local/Network)

OK_but


15 Lync user1 calIs IP Phone which is in immediate forward mode to Lync user2 (51) (Local/Network)

OK_but


16 Oxe Phone call Lync user then Oxe phone display secret identity

OK Identity secret display on OXE phone

17 Lync user put OXE phone on hold OK_but

Local music on hold played on Lync client

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 Result

Comment

1 A=Lync user, B=IPPhone1, C=IPPhone2 (Local/Network)

1.1 Hold state request Lync user->IPPhone1, Lync user on hold

OK_but Local music on hold played on Lync client

1.2 Consultation call request IPPhone1->IPPhone2

OK

1.3 Broker request IPPhone1->Lync user, IPPhone2 on hold

OK

2 A=PSTN, B=IPPhone1, C=Lync user (Local/Network)

2.1 Hold state request PSTN->IPPhone1, PSTN on hold

OK

2.2 Consultation call request IPPhone1->Lync user

OK


2.3 Broker request IPPhone1->PSTN, Lync user on hold


3 A=PSTN, B= Lync user , C= IPPhone1 (Local/Network)

3.1 Hold state request PSTN-> Lync user, PSTN on hold

OK

3.2 Consultation call request Lync user-> IPPhone1

OK

3.3 Broker request Lync user->PSTN, IPPhone1 on hold

Ok

4 A=IPPhone1, B=IPPhone2, C=Lync user (Local/Network)

4.1 Hold state request IPPhone1->IPPhone2, IPPhone1 on hold

OK

4.2 Consultation call request IPPhone2->Lync user

OK

4. 3 Broker request IPPhone2->IPPhone1, Lync user on hold


5 A=IPPhone1, B=Lync user, C=IPPhone2 (Local/Network)

5.1

Hold state request IPPhone1->Lync user, IPPhone1 on hold

OK

5.2 Consultation call request Lync user->IPPhone2

OK

5.3 Broker request Lync user->IPPhone1, IPPhone2 on hold

OK

6 A=Lync user1, B=IPPhone, C= Lync user2 (Local/Network)

6.1

Hold state request Lync user1->IPPhone, Lync user1 on hold


6.2 Consultation call request IPPhone->Lync user2

OK

6.3 Broker request IPPhone->Lync user1, Lync user2 on hold


7 A=Lync user1, B=IPPhone, C=UA/TDM set (Local/Network)

7.1 Hold state request IPPhone->Lync user1, IPPhone on hold


7.2

Consultation call request Lync user1->UA/TDM set

OK

7.3

Broker request Lync user1->IPPhone, UA/TDM set on hold

Ok

8 A=Lync user1, B=UA/TDM set 1 C=UA/TDM set2 (Local/Network)

8.1 Hold state request IUA/TDM set1 ->Lync user1, UA/TDM set1 on hold

OK

8.2 Consultation call request Lync user1 > UA/TDM set2

OK

8.3 Broker request Lync user1->UA/TDM set 1 UA/TDM set2 on hold

OK

9 A=IPPhone, B=Lync user1, C=Lync user2 (Local/Network)

9.1 Hold state request IPPhone->Lync user1, IPPhone on hold

OK


9.2 Consultation call request Lync user1->Lync user2

OK

9.3 Broker request Lync user1->IPPhone, Lync user2 on hold


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 Comment

A Transferee

B Transferor

C Transfer Target

1 LYNC OXE OXE NA

Unattended transfers not available from an OXE user.

2

OXE LYNC OXE OK_but

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.

3 OXE OXE LYNC NA


4

OXE LYNC LYNC OK_but

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.

5 LYNC OXE LYNC NA


6

LYNC LYNC OXE OK_but

No display update on A. AudioCodes gateway doesnt receive the final destination information from Microsoft Lync. 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.

Semi attended Transfer (On Ringing)


A Transferee

B Transferor

C Transfer Target

1 LYNC OXE OXE OK_but

No display update on A. AudioCodes gateway receives the REFER from OXE, but doesnt send to Lync.

2 OXE LYNC OXE NA

Semi-attended transfers not available from a Lync user.

3 OXE OXE LYNC OK 4

OXE LYNC LYNC NA Semi-attended transfers not available from a Lync user.


5 LYNC OXE LYNC OK 6

LYNC LYNC OXE NA Semi-attended transfers not available from a Lync user.

Attended Transfer (in conversation)


A Transferee

B Transferor

C Transfer Target

1 LYNC OXE OXE OK 2

OXE LYNC OXE OK_but No display update on A. AudioCodes gateway doesnt receive the final destination information from Microsoft Lync.

3 OXE OXE LYNC OK_but

No display update on A. AudioCodes gateway receives the REINVITE from OXE, but doesnt send to Lync.

4 OXE LYNC LYNC OK_but

No display update on A. AudioCodes gateway doesnt receive the final destination information from Microsoft Lync.

5 LYNC OXE LYNC OK_but

No display update on A. AudioCodes gateway receives the REINVITE from OXE, but doesnt send to Lync.

6 LYNC LYNC OXE OK_but

No display update on A. AudioCodes gateway doesnt receive the final destination information from Microsoft 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 Result

Comment

1 3 steps conference request PSTN->IPPHone->Lync user

OK

2 Step1 then IPPhone leaves the conference

OK

3 Step 1 then Lync user leaves the conference

OK

4 Step 1 then PSTN leaves the conference

OK

5 3 steps conference request PSTN->Lync user-> IPPHone

OK

6 Step 5 then IPPhone leaves the conference

OK

7 Step 5 then Lync user leaves the conference

OK

8 Step 5 then PSTN leaves the conference

OK

9 Stay in a conference for long period to check keep alive mechanisms.

OK


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


1 IPTouch phone leave as a VoiceMail message for the Lync user Check that MWI is OK on Lync user

Not tested

2 Message consultation by Lync user Not tested

3 Password modification by Lync user Not tested

4 Lync user call to a OXE user forwarded to Voice Mail (4645)

OK

5 Message consultation by OXE user OK

6.5 Attendant

6.5.1 Test Objectives An attendant console is defined on the system. Call going to and coming from the attendant console are tested.

6.5.2 Test Results


1 Lync user calls attendant (using attendant calls prefix 8)

OK

2 Attendant calls Lync user OK 3 Attendant calls Lync user then

Attendant calls OXE user Attendant transfer Lync user To OXE user

OK

4 Lync user calls attendant (using attendant calls prefix 8), attendant transfers on ringing to OXE set.

OK


5 Lync user calls attendant (using attendant calls prefix 8), attendant transfers in conversation to OXE set,

OK

6 OXE set calls to attendant (using attendant calsl prefix **8), attendant transfers in conversation to Lync user.

OK

7 External user(T2) calls Attendant Attendant calls Lync user Attendant transfer on ringing External user(T2) to Lync user

OK_but But the display on Lync user is not updated.

8 Lync user calls Attendant Attendant calls External user(T2) Attendant transfer on conversation Lync user to External user(T2)

OK_but But the display on Lync user is not updated.

6.6 Defense / Recovery

6.6.1 Test Objectives 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 Action Result

Comment

1 Temporary Data Network Link down with the PBX and Mediant 1000 MSBG OK

Existing calls are stopped. Establishing new call is possible when the link is reestablished.

2 Spatial redundancy IP Method : CPU switchover with SIP communication

OK

3

Spatial redundancy DNS method (delegation on a third party DNS server) : CPU switchover without SIP communication

OK

4

Spatial redundancy DNS method : CPU switchover with SIP communication OK

5

Switchover to Passive Call Server (PCS). (IP link to main/stdby OXE call servers down)

N/A

6

Switchover to a backup AudioCodes gateway. Stop the main AudioCodes gateway; verify that a call is possible with the backup AudioCodes gateway.

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: (e.g., 10.15.7.131). Prefix Length: The Subnet Mask in bits (e.g., 16 for 255.255.0.0). 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. Set the following parameters: DNS Primary Server IP: (e.g., 10.1.2.21). DNS Secondary Server IP: .

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. Set the following parameters: 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. Set the following parameters: 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. Set the following parameters: 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. Set the following parameters: 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:

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.

2. Open the 'Configuration File' page (Configuration tab > SBC menu > Manipulations SBC menu > Message).

7.2.16 Admin Page Parameters

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

[ 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 ]


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 37 (2) Trunk Group Type T2 Trunk Group name SIPTRK Remote Network 15 Node number 1 Q931 signal variant ABC-F (private sip trunk) T2 Specification SIP Overlap dialing No Virtual accesses Trunk Groups/Trunk Group/Virtual access for SIP Trunk Group Id 37 Number of SIP Access 2 (ie 2*62 accesses) Trunk group Trunk Groups Trunk Group ID 38 (2) Trunk Group Type T2 Trunk Group name SIPTRK2 Remote Network 15 Node number 1 Q931 signal variant ABC-F (private sip trunk) T2 Specification SIP Overlap dialing No Virtual accesses Trunk Groups/Trunk Group/Virtual access for SIP Trunk Group Id 38 Number of SIP Access 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 2 SIP Remote domain 10.200.56.3 SIP Port Number 5060 SIP Transport Type TCP Trunk group number 38 Supervision timer 10 (used for AudioCodes gateway redundancy) RFC 3325 supported by the distant 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 1 (4) Carrier Reference 0 Command I Associated Ext SIP gateway 1 Dialing command table (DCT) Translator/Automatic Route Selection/Dialing Command Table Table Id 2(4) Carrier Reference 0 Command I Associated Ext SIP gateway 2 ARS route list Translator/Automatic Route Selection/ARS Route list ARS Route list 4 (5) Name SIP ARS ARS route Translator/Automatic Route Selection/ARS Route list/ARS Route ARS Route list 4 Route 1 (6) Name SIP ARS route 1 Trunk Group 37 No.Digits To Be Removed 0 Digits To Add 97239764 Numbering Command Tabl. ID 1 Protocol Type Dependant on Trunk Group Type Quality Speech ARS route Translator/Automatic Route Selection/ARS Route list/ARS Route ARS Route list 4 Route 2 (6) Name SIP ARS route 2 Trunk Group 38 No.Digits To Be Removed 0 Digits To Add 97239764 Numbering Command Tabl. ID 2 Protocol Type Dependant on Trunk Group Type Quality Speech

Time-based route list Translator/Automatic Route Selection/ARS Route list/Time-based 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 2 (7) Name SIP

Discriminator rule Translator/External Dialing Plan/Dialing (Numbering) Discriminator/Discriminator rule

Discriminator n 2 Call number 2 Area number 1 (8) ARS route list number 4 Number of digits 3 Logical discriminator Entities/Discriminator Selector Entity Number 1 Discriminator 0x (9) 2 ARS prefix Translator/Prefix plan Number 4982 Prefix Meaning ARS Prof.Trg Grp Seizure Discriminator n 2 Public Network COS Classes of Service / Access COS / Public Access COS Public Network COS Area identifier Public Access Rights Night 1 Day 1 Mode 1 1 Mode 2 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 Dialing command table (DCT) Translator/Automatic Route Selection/Dialing Command Table Table Id 1 (4) Carrier Reference 0 Command I Associated Ext SIP gateway 1 Dialing command table (DCT) Translator/Automatic Route Selection/Dialing Command Table Table Id 1 (4) Carrier Reference 0 Command I Associated Ext SIP gateway 2 ARS route list Translator/Automatic Route Selection/ARS Route list ARS Route list 4 (5) Name SIP ARS ARS route Translator/Automatic Route Selection/ARS Route list/ARS Route ARS Route list 4 Route 1 (6) Name SIP ARS route 1 Trunk Group 37 No.Digits To Be Removed 0 Digits To Add 97239764 Numbering Command Tabl. ID 1 Protocol Type Dependant on Trunk Group Type Quality Speech ARS route Translator/Automatic Route Selection/ARS Route list/ARS Route ARS Route list 4 Route 2 (6) Name SIP ARS route 2 Trunk Group 32 No.Digits To Be Removed 0 Digits To Add 97239764 Numbering Command Tabl. ID 2 Protocol Type Dependant on Trunk Group Type Quality Speech

Time-based route list Translator/Automatic Route Selection/ARS Route list/Time-based 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 2 (7) Name SIP

Discriminator rule Translator/External Dialing Plan/Dialing (Numbering) Discriminator/Discriminator rule

Discriminator n 2 Call number 2 Area number 1 (8) ARS route list number 4 Number of digits 3 Logical discriminator Entities/Discriminator Selector Entity Number 1 Discriminator 0x (9) 2 ARS prefix Translator/Prefix plan Number 4982 Prefix Meaning ARS Prof.Trg Grp Seizure Discriminator n 2 Public Network COS Classes of Service / Access COS / Public Access COS Public Network COS Area identifier Public Access Rights Night 1 Day 1 Mode 1 1 Mode 2 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 third-

party 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 Alcatel-Lucent 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 Alcatel-Lucent 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 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 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 (Alcatel-

Lucent 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 Alcatel-

Lucent Business Partners): https://businessportal.alcatel-lucent.com click under Let us help you the eService Request link

e-mail: [email protected] 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

French

+800-00200100

Belgium

Luxembourg

Germany

German Austria

Switzerland

United Kingdom

English

Italy

Australia

Denmark

Ireland

Netherlands

South Africa

Norway

Poland

Sweden

Czech Republic

Estonia

Finland

Greece

Slovakia

Portugal

Spain Spanish

For other countries:

English answer : + 1 650 385 2193 French answer : + 1 650 385 2196 German answer : + 1 650 385 2197 Spanish answer : + 1 650 385 2198


END OF DOCUMENT

audiocodesmediant1000-msbg lync omnipcxenterpriser10.0 iwr ed01

Documents

alcatellucent product

alcatellucent issues

application partner

application partner

audiocodes application

media gateway application

new version

rights reserved company