audiocodes-mp202-v2.6.4_omnipcxenterprise-r9.0_iwr_ed02.pdf

Alliance & Application Partner Program Inter-Working Report

Partner: AudioCodes Application : Telephone Adapter / VoIP gateway

Application name: MP-202

The product and version listed have been tested with the Alcatel-Lucent Communication Server and the version specified hereinafter. The tests concern only the inter-working between the Application Partner product and the Alcatel-Lucent Communication platforms. The inter-working report is valid until the Application Partner issues a new version of such product (incorporating new features or functionality), or until Alcatel-Lucent issues a new version of such Alcatel-Lucent product (incorporating new features or functionality), whichever first occurs.

ALCATEL-LUCENT MAKES NO REPRESENTATIONS, WARRANTIES OR CONDITIONS WITH RESPECT TO THE APPLICATION PARTNER PRODUCT. WITHOUT LIMITING THE GENERALITY OF THE FOREGOING, ALCATEL-LUCENT HEREBY EXPRESSLY DISCLAIMS ANY AND ALL REPRESENTATIONS, WARRANTIES OR CONDITIONS OF ANY NATURE WHATSOEVER AS TO THE APPLICATION PARTNER PRODUCT INCLUDING WITHOUT LIMITATION THE IMPLIED WARRANTIES OF MERCHANTABILITY, NON INFRINGEMENT OR FITNESS FOR A PARTICULAR PURPOSE AND ALCATEL-LUCENT FURTHER SHALL HAVE NO LIABILITY TO APPLICATION PARTNER OR ANY OTHER PARTY ARISING FROM OR RELATED IN ANY MANNER TO THIS CERTIFICATE.

Alliance & Application Partner Program Inter-working report Edition 2 - page 2/41 Copyright 2008 Alcatel-Lucent, All rights reserved

Tests identification

Date of the tests March 2009

Alcatels representative Bruno Fouarge

Lars Ruoff

Partners representative Christophe Hubaut

Alcatel Communication Platform OmniPCX Enterprise Alcatel compatibility release R9.0 (H1.301.27) Partners application version 2.6.4

Environment (if it has a sense)

Author(s): Bruno Fouarge, Lars Ruoff Reviewer(s): Rachid Himmi, Denis Lienhart, Bernard Roth

Historic

Edition 1: Creation of the document 2009-03-23 Edition 2: Added note about voice use restriction 2010-08-17

Test results

Passed

Refused Postponed

Passed with restrictions

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

ACD Business


Company Contact Information (Information provided by the partner)

Contact name: Bertrand Goldfarb Title: Sales Director

Address 1: 54, route des Sartrouville City: LE PECQ Zip: 78230 Country: France

Phone: +33 1 74 90 96 00 Fax: +33 1 39 76 89 61

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 .................................................................................................................... 9

4 SUMMARY OF TEST RESULTS ............................................................................................................ 11 4.1 SUMMARY OF MAIN FUNCTIONS SUPPORTED .......................................................................................... 11 4.2 SUMMARY OF PROBLEMS ...................................................................................................................... 11

4.2.1 AudioCodes side ........................................................................................................................ 11 4.2.2 Alcatel-Lucent side ..................................................................................................................... 11 4.2.3 Other problems (origin not identified) ...................................................................................... 11

4.3 SUMMARY OF LIMITATIONS.................................................................................................................... 11 4.4 NOTES, REMARKS ................................................................................................................................. 11

5 TEST SCENARIOS.................................................................................................................................... 13 5.1 TEST PROCEDURE ................................................................................................................................... 13 5.2 RESULT TEMPLATE................................................................................................................................. 13

6 TESTING..................................................................................................................................................... 14 6.1 SIP SET COMMISSIONING...................................................................................................................... 14

6.1.1 Test objectives ............................................................................................................................ 14 6.1.2 Test procedure............................................................................................................................ 14

6.2 OUTGOING CALLS .................................................................................................................................. 15 6.2.1 Test Objectives .............................................................................................................................. 15 6.2.2 Test Results ................................................................................................................................... 15

6.3 INCOMING CALLS................................................................................................................................... 16 6.3.1 Test Objectives .............................................................................................................................. 16 6.3.2 Test Results ................................................................................................................................... 17

6.4 FEATURES DURING CONVERSATION ....................................................................................................... 19 6.4.1 Test Objectives .............................................................................................................................. 19 6.4.2 Test Results ................................................................................................................................... 19

6.5 CALL TRANSFER .................................................................................................................................... 19 6.5.1 Test Objectives .............................................................................................................................. 19 6.5.2 Test Results ................................................................................................................................... 20

6.6 ATTENDANT........................................................................................................................................... 21 6.6.1 Test Objectives .............................................................................................................................. 21 6.6.2 Test Results ................................................................................................................................... 21

6.7 VOICE MAIL........................................................................................................................................... 21 6.7.1 Test Objectives .............................................................................................................................. 21 6.7.2 Test Results ................................................................................................................................... 21

6.8 DUPLICATION AND ROBUSTNESS ........................................................................................................... 22 6.8.1 Test Objectives .............................................................................................................................. 22 6.8.2 Test Results ................................................................................................................................... 22

6.9 FAX....................................................................................................................................................... 24 6.9.1 Test objectives ............................................................................................................................... 24 6.9.2 Test procedure............................................................................................................................... 24

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

APPENDIX D: ALCATEL-LUCENT COMMUNICATION PLATFORM CONFIGURATION REQUIREMENTS.............................................................................................................................................. 31


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

APPENDIX F: AAPP PROGRAM, DOCUMENTATION AND TECHNICAL ASSISTANCE................ 35

APPENDIX G: ALCATEL-LUCENT ESCALATION PROCESS IN CASE OF PROBLEM WITH A CERTIFIED EXTERNAL APPLICATION (REFERENCED IN THE AAPP)........................................... 39


1 Introduction

The goal of these tests is to qualify an external application as an Alliance & Application Partner Program solution for the Alcatel-Lucent Communication Platform.

The scope of the tests is the interoperability of the application with the Alcatel-Lucent Communication Platform. It covers a basic or complex inter-working to ensure that services requested by the application and provided by the Communication Platform (and/or conversely) are properly completed. These tests do not verify the functional achievement of the application as well as they do not cover load capacity checks, race conditions and generally speaking any real customer's site conditions.

This document is the interworking report between the AudioCodes MP-202 FXS working in SIP mode - and an OmniPCX Enterprise (OXE) Call Server.

1.1 Glossary

ABC-F : Alcatel Business Communication telephone Feature

ARS : Automatic Route Selection

CS : Call Server (Alize or Appliance Communication server) CSTA : Computer Supported Telecommunications Applications

CPU : Call Processing Unit (4400 Communication Server) DTMF : Dual-Tone Multi-Frequency

GD : Gateway Driver

IP : Internet Protocol

ISDN : Integrated Services Digital Network

MG : Media Gateway

MLA : Multi Line Appearance

OXE : OmniPCX Enterprise

SIP : Session Initialization Protocol

UA : Universal Alcatel

UM : Unified Messaging

Z : Alcatel-Lucent Analogical Set


2 Application information

Application type: Telephone Adapter / VoIP Gateway for Analog equipments (Sets, Faxes).

Application commercial name: MP-202-B

Application version: 2.6.4_p7_build_9

Interface type: SIP

Brief application description:

AudioCodes MP202 is a telephone adapter that allows connecting ordinary analog telephones or fax machines to a Voice over Broadband service. It is typically adapted for Branch Offices. The MP202 connects to a Service Provider by using its IP uplink connection. It proposes up to 2 VoIP ports for connecting up to 2 analog sets or faxes. MP202 also supports the SIP protocol, used in the present case for communicating with OXE. The equipments connected on the MP202 ports will therefore be declared as SIP terminals (extern station) and will register on OXE.

MP202 also supports additional features such as routing, NAT, DHCP, PPPoE functions as well as a built-in firewall. These features are not covered by the present test campaign.


3 Tests environment

3.1 General architecture The tests are performed on the Alcatel-Lucent TSS Applications International platform in the following environment:

Figure 1 - Test System Architecture

IP Network

Node 1 OXE/Aliz hardware Duplicated call servers, spatial redundancy 4645 Voice Mail

Node 2 4400/crystal hardware Single CPU

PSTN Network Simulated by loopback

Analog set

FAX

AudioCodes MP-202-B

FAX

ABCF/IP hybrid link

SIP phone

IP phones


3.2 Hardware configuration

Alcatel-Lucent Communication Platform: Node1:

o OmniPCX Enterprise common hardware (Aliz) o Duplicated call servers o Spatial redundancy (Different IP subnetworks) o Two media gateways:

Cristal 0 : +-------------------------------------------------------------------+ | Cr | cpl| cpl type | hw type | cpl state | coupler ID | |----|----|------------|-----------|--------------|-----------------| | 0 | 6 | CPU_CS|---------- | IN SERVICE | NO PCMS CODE | | 0 | 10 | CPU_CS|---------- | IN SERVICE | NO PCMS CODE | +-------------------------------------------------------------------+

Cristal 1 : +-------------------------------------------------------------------+ | Cr | cpl| cpl type | hw type | cpl state | coupler ID | |----|----|------------|-----------|--------------|-----------------| | 1 | 0 | GD|---------- | IN SERVICE | NO PCMS CODE | | 1 | 1 | UAI 8|---------- | IN SERVICE | NO PCMS CODE | | 1 | 3 | PRA T2|---------- | IN SERVICE | NO PCMS CODE | | 1 | 6 | PRA T2|---------- | IN SERVICE | NO PCMS CODE | +-------------------------------------------------------------------+

Cristal 2 : +-------------------------------------------------------------------+ | Cr | cpl| cpl type | hw type | cpl state | coupler ID | |----|----|------------|-----------|--------------|-----------------| | 2 | 0 | GD|---------- | IN SERVICE | NO PCMS CODE | | 2 | 1 | MIX448|---------- | IN SERVICE | NO PCMS CODE | +-------------------------------------------------------------------+

Node2:

OmniPCX Enterprise crystal hardware (4400) voice hub Single CPU One INTIP:

Cristal 0 : +-------------------------------------------------------------------+ | Cr | cpl| cpl type | hw type | cpl state | coupler ID | |----|----|------------|-----------|--------------|-----------------| | 0 | 1 | CPU7|---------- | IN SERVICE | BAD PCMS CODE | | 0 | 3 | UA32|---------- | IN SERVICE | BAD PCMS CODE | | 0 | 4 | INTIPA| INT-IP | IN SERVICE | BAD PCMS CODE | +-------------------------------------------------------------------+

Application platform:

AudioCodes MP-202-B

3.3 Software configuration

Alcatel-Lucent Communication Platform: OmniPCX Enterprise R9.0 h1.301.25


Note: Handsets are declared as SIP extensions (SIP Endpoint Level of Service - SEPLOS). This means that all telephone features must be tested once by prefixes and once locally on the phone, if available. Recommended configuration:

OXE > SIP > SIP Registrar > SIP Min Expiration Date = 60. This is to allow for a 60 secs registration time for the IPBS, see IPBS recommended configuration below.

Application platform:

AC MP-202 2.6.4


4 Summary of test results

4.1 Summary of main functions supported

The main supported functions implied in the test campaign are: Installation and configuration of the AudioCodes gateway Configuration of SIP users connected to the AudioCodes gateway on the OXE

registrar server Registration and call authentication* Basic communications (internal and external, intra-node, inter-nodes) Supports telephonic features (Broker call, Transfer*, Conference, Call back, Waiting*,

Forwards*, Do Not Disturb, in intra-node and inter-nodes communications).. Access to the OXE Voice mail boxes and MF dialing for navigation Multi Frequency sending in conversation/connected state Framing, VAD, Codec impact* Fax (T38) transmission in intra and inter node Simple and spatial call server redundancy

4.2 Summary of problems

4.2.1 AudioCodes side Some transfer cases not working. See section 6.3. Note: Due to this limitation we do not recommend to use AudioCodes MP202 for use as a voice gateway (with analog voice phones). FAX operations are OK. The restriction is present for both SIP device and SIP extension mode.

4.2.2 Alcatel-Lucent side Forward to external number not possible. See section 6.3.

4.2.3 Other problems (origin not identified) Music on hold : in case of remote calls, when the called set put the calling (MP202)

on hold, no music on hold is heard. Local call to SIP terminal in forward on busy (CFB) state NOK.

4.3 Summary of limitations

Overlap dialing not supported. After a second call, if you press the R key, you go in conference. If you press again

the R key, you release call the conference.=> Not possible to keep the second call. Not possible to call sip device with attendant. Message Waiting Indication (MWI): The message led on the analog phone is not

blinking.

4.4 Notes, remarks


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

extension, the other one as SIP device.


5 Test Scenarios

5.1 Test procedure

Step Action N/A Result Origin of the problem Comment

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

it is not intended during this test session to debug and fix problems.

5.2 Result template

The results table must be formatted as indicated in the example below:

Step Action N/A Result Origin of the problem

Comment

1 . action 1 OK 2 . action 2

OK The application waits for PBX timer or phone set hangs up

3 . action 3 OK 4 . action 4

X

Relevant only if the CTI

interface is a direct CSTA

link 5 . action 5

NOK No indication,

no error message


6 Testing

6.1 SIP set commissioning

6.1.1 Test objectives

These tests shall verify that the analog devices ( phones and fax machine ) connected to the MP-202 can be declared on OmniPCX Enterprise as SIP extension (SEPLOS) and are able to register properly on the OXE server registrar.

6.1.2 Test procedure Test Action Result Comment

1 Provisioning OK 2 DHCP registration

(with OXE internal DHCP server) Not

tested 3 NTP registration

/ The SIP phone is configured to retrieve the date and time from the OXE internal NTP server Check the phone retrieves the information and displays it.

OK The NTP server of the MP202 is set as the MAIN 1 CPU address (not the DNS name)

SIP set registration to OXE in static IP addressing mode / The phone is configured to use IP static parameters

OK 4A SIP registration, using OXE MAIN IP

addresse(s) (without authentication)

OK

4B SIP registration, using DNS (without authentication) / The phone is configured to use a domain name as registrar / proxy server address. The DNS IP address is the OXE CPU main address

OK The MP202 DNS is specified as Main1 and the secondary DNS as MAIN 2.

5 Support of 423 Interval Too Brief (1)

OK By default, OXE>SIP>SIP Registrar>Min Expires=1800s

6 SIP registration with authentication: Turn on SIP Digest authentication, specify realm on OXE, and specify user name and password on SIP client. / SIP digest authentication is activated on OXE and phone side. Check also that outgoing call is authenticated.

OK

It works for Registration authentication. Not for Outgoing call with authentication. Hence the OXE parameter in the SIP proxy must be set as : Only authenticated incoming call == false


Notes:

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

6.2 Outgoing Calls

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

6.2.2 Test Results Test Action Result Comment

1 Call to a local user OK 2 Call to a local user with overlap

dialing: Dial a part of the number, wait and continue.

NOK Overlap dialing not supported

3 Call to a local user with overlap dialing, timeout: Dial a part of the number, wait and stop.


4 Call to a local user with overlap dialing, release: Dial a part of the number, wait and release the call.


5 Call to local user with no answer. Check timeout. OK

Cancel sent by SIP device after few seconds

6 Call to another SIP set / The SIP phone under test calls another SIP phone. Check the display and audio during all steps (dialing, ring back tone, conversation, release).

OK

7 Call to wrong number (SIP: 404 Not Found) OK

8 Call rejected by call handling (SIP: 183 Progress/487 Request Terminated) e.g. max number of calls reached etc.

9 Call to busy user (SIP: 486 Busy Here) Check busy tone.

OK


10 Call to user in Out of Service state (SIP: 480 Temporarily Unavailable) OK

11 Call to user in Do not Disturb state (SIP: 603 Decline) OK

12 Call to local user, immediate forward (CFU). (SIP: 302 Moved Temporarily)(1)

OK

13 Call to local user, forward on no reply (CFNR). (1) OK

14 Call to local user, forward on busy (CFB). (1 OK

15 Call to a local user with proxy Authentication OK

16 Call within same IP domain. SIP set in domain A (intra-domain=without compression). Call to OXE set in domain A (intra-domain=without compression).

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

17 Call to another IP domain. SIP set in domain A (extra-domain=with compression). Call to OXE set in domain B (extra-domain=with compression).

OK Check order of codecs in SDP list. Check chosen codec. We expect G.729 See note (2)

18 Call to external number (via T2 loopback) (Check ring back tone, called party display) / Dial external number 03881234XY =>1XY

OK

Ring back tone ok but display not available on analog device

19 SIP session timer expiration: Check if call is maintained or released after the session timer has expired.

OK Call is maintained by re- INVITE

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

6.3 Incoming Calls

6.3.1 Test Objectives


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


1 Local /network call to free SIP terminal / Check ring back tone, called party display

OK

2 Local/network call to busy SIP terminal / Call a set which is already in conversation

OK

3 Local/network call to unplugged SIP terminal

OK As long as the SIP device is registered, we get a ring back tone. If is is not registered, we are prompted to go on hook.

4 Local/network call to SIP terminal in Do Not Disturb (DND) mode

4A By local feature on MP202, using *73 prefix

OK

4B By system feature (SEPLOS) Dial 42 + user password wait for acknowledgement ring back tone from OXE

OK

5 Local/network/SIP call to SIP terminal in immediate forward (CFU) to local user:

5A By local feature of MP202, using the *72 prefix

OK

5B By system feature (SEPLOS) Dial 51 + target extension wait for acknowledgement ring back tone from OXE / Dial 41 for forward cancellation

OK

6 Local/network/SIP call to SIP terminal in immediate forward (CFU) to external number

6A By local feature if applicable NOK Call not established but SIP device sends the temporary moved with the right external number. => OXE problem. crms00168027

6B By system feature (SEPLOS) / Dial 51 + target extension wait for acknowledgement ring back tone from OXE / Dial 41 for forward cancellation

NOK SIP device still gets INVITEd and rings. => OXE problem. crms00168030

7 Local/network/SIP call to SIP terminal in immediate forward (CFU) to another SIP user


7A By local feature if applicable OK 7B By system feature (SEPLOS)

Dial 51 + target extension wait for acknowledgement ring back tone from OXE / Dial 41 for forward cancellation

OK

8 Local call to SIP terminal in forward on busy (CFB) state:

8A By local feature if applicable *72 NOK Call not established but SIP device sends the temporary moved with the right external number. => OXE problem. crms00168027

8B By system feature (SEPLOS) Dial 52 + target extension wait for acknowledgement ring back tone from OXE / Dial 41 for forward cancellation

NOK The SIP device sends 486 Busy here. It behaves like a mono-line set, but is declared as multi-line. OXE Problem: For the moment it is not possible to remove the 2nd line on a SIP extension.

9 Local call to SIP terminal in forward on no reply (CFNR)

9A By local feature if applicable *72 OK 9B By system feature (SEPLOS)

Dial 53 + target extension wait for acknowledgement ring back tone from OXE / Dial 41 for forward cancellation

OK

10 Call to busy user, Call waiting. (Camp-on)(local feature)

N/A Feature unavailable.

11 External call to SIP terminal. Check that external call back number is shown correctly. / Dial external number 03881234XY =>1XY

N/A Feature unavailable.

12 CLIR: Local call to SIP terminal. / Check that caller id is not shown.

N/A Analogue device does not allow display of the calling id

13 Display: Call to free SIP terminal from user with a name containing non-ASCII characters. Check caller display.


14 Display: Call to free SIP terminal from user with a UTF-8 name containing non-ASCII characters. Check caller display.


15 SIP set is part of a sequential hunt group. Call to hunt group. Check call/release.

OK

16 SIP set is part of a cyclic hunt group. Call to hunt group. Check call/release.

OK

18 SIP set is declared as a twin set (tandem). Call to main set and see if twin set rings. Take call with twin set.

OK


6.4 Features during Conversation

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


1 Hold and resume (both directions) using R key on sip device (Check tones)

OK

2 Second call to another local user. Distant user is put on hold.

OK

3 Broker request (toggle back and forth between both lines, local feature if applicable), using the 1 suffix

OK Using the R + on the Z phone

4 Release first call. Keep second call. NOK

After the second call, if you press the R key, you go in conference. If you press again the R key, you release call the conference.=> Not possible to keep the second call

5 Call park: - Call between SIP set and OXE set. - Put your call on hold. - New call: Dial the prefix for call parking (402+number). Now call can be hung up. Later call can be retrieved by calling prefix+number again.

OK

6 Send/receive DTMF OK DTMF are sent as RFC2833 7 Three party conference initiated

from OXE set. Released by OXE set. / Dial extension target + suffix 3.

OK

8 Three party conference initiated from SIP set (local feature). Released by SIP set.

OK

6.5 Call Transfer

6.5.1 Test Objectives During the consultation call step, the transfer service can be requested and must be tested. Several transfer services exist: Unattended Transfer, Semi-Attended Transfer and Attended Transfer. Audio, tones and display must be checked.


We use the following scenario, terminology and notation: There are three actors in a given transfer event:

A Transferee : the party being transferred to the Transfer Target. B Transferor : the party doing the transfer. C Transfer Target : the new party being introduced into a call with the Transferee.

The scenario is:

1. A (Transferee) calls B (Transferor). 2. B (Transferor) calls C (Transfer Target). A is on hold during this phase. 3. B (Transferor) executes the transfer. B drops out of the communication. A is now in

communication with C. There are three sorts of transfers in the SIP world:

Unattended Transfer or Basic Transfer: Unattended Transfer consists of the Transferor providing the Transfer Target's contact to the Transferee. The Transferee attempts to establish a session using that contact and reports the results of that attempt to the Transferor.

Semi-Attended Transfer or Early Attended Transfer or Transfer on ringing : Semi-Attended Transfer consists of the Transferor having a session in hold state with the Transferee and initiating the Transfer when the consultation call to the Target Transfer is in ringing state.

Attended Transfer or Transfer in conversation: Attended Transfer consists of the Transferor having a session in hold state with the Transferee and initiating the Transfer when the session between the Transferor and Transfer target is established.

Note: Unattended Transfer is not provided by the OXE.

6.5.2 Test Results In the below table, SIP means a partner SIP set, OXE means a proprietary OXE (Z/UA/IP) set. Att means attended transfer (transfer in conversation), Semi means Semi-attended transfer (transfer on ringing) and Unatt means unattended transfer. Test Action Result Comment A

Transferee

B Transferor

C Transfe

r Target

Type

1 OXE MP 202 OXE Att OK 2 OXE MP 202 OXE Semi OK 3

MP202 OXE OXE Att NOK User C doesnt hear user A. MP202

limitation. See ALu Ref 1-106975741 and AC Ref #84185

4 MP 202 OXE OXE

Semi NOK User C doesnt hear user A. MP202 limitation. See ALu Ref 1-106975741 and AC Ref #84185

5 OXE OXE MP 202 Att OK 6 OXE OXE MP 202 Semi OK See note (1) 7 SIP OXE MP 202 Att OK 8 SIP OXE MP 202 Semi OK See note (1) 9 OXE SIP MP 202 Att OK 10 OXE SIP MP 202 Semi OK See note (1) 11 MP 202 SIP OXE Att OK


12 MP 202 SIP OXE Semi OK 13 MP 202 SIP SIP Att OK 14 MP 202 SIP SIP Semi OK See note (1)

Note (1): We define two different semi-attended transfers. The transfer on ringing (when target set rings at least one time) and immediate transfer (when the transferor goes on hook immediately after having transferred the initial caller without waiting for the target ring).

6.6 Attendant

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


1 call to attendant (using attendant call prefix 9) OK

2 call to attendant (using attendant call prefix 9), attendant transfers to OXE set, unattended.

OK

3 call to attendant (using attendant call prefix 9), attendant transfers to OXE set, attended.

OK

4 OXE set calls to attendant (using attendant call prefix 9), attendant transfers to SIP set, attended.

OK

6.7 Voice Mail

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


1 A Voice Mail message for the SIP subscriber is generated. Check that MWI is activated. / Dial prefix 570 + target extension or target extension + suffix 8

(OK) The message led on the Z phone is not blinking. Probably a poroblem of Z phone compatibility with voltages.

2 Message consultation OK 3 Password modification OK


4 MP202 call to a OXE user forwarded to Voice Mail / Dial 51 + VoiceMail extension, wait for acknowledgement ring back tone from OXE / Dial 41 for forward cancellation

OK

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

OK

6.8 Duplication and Robustness

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


1 Spatial redundancy, using Alternate Proxy method: Switchover to standby call server.

(OK)

Works with restrictions: Still need to have DNS configured, because when no DNS is configured, MP-202 cannot resolve the Contact FQDN. Can only call after re-registered on new proxy. => requires frequent registration to minimize downtime after switchover. Not recommended.


2 Spatial redundancy, using DNS method: Switchover to standby call server. OK

Works very well, seamless switchover. DNS requests are sent simultaneously to CPU A and B. => no delay. Recommended method.

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

N/A Not tested, not available in R9.0

4 SIP device reboot. Check that calls are possible as soon as device has come back to service.

OK

5 Temporary Link down with the PBX OK


6.9 FAX

6.9.1 Test objectives Perform fax transmission in T.38 mode Note: For these tests, the user is declared in SIP device (trunk) mode

6.9.2 Test procedure

Test Action Result Comment 1 single fax (local)

OK

2 Fax local forwarding OK 3 Fax sending to busy fax machine

OK

4 single fax through PSTN

OK

5 single fax (remote)

OK

6 long and complex fax

OK


Appendix B : AudioCodes Configuration

MP202 configuration (Spatial redundancy, with DNS method)

IP addresses setting (WAN, DNS method)

CS A MAIN CS B MAIN


SIP domain declaration

SIP proxy declaration (DNS method, recommended)

Node+domain name

Node+domain name


SIP proxy declaration (redundant proxy method, NOT recommended!)

CS A MAIN

CS B MAIN Redundant proxy method


Codecs setting


Services setting


Analog set configuration


Appendix D: Alcatel-Lucent Communication Platform Configuration Requirements

Handsets are declared as SIP extensions for analog phones, and as SIP device for FAX users: Review/Modify: Users Node Number (reserved) : 1 Directory Number : 160 Directory name : SIP_160 Directory First Name : Anna UTF-8 Directory Name : --------------------------------------- UTF-8 Directory First Name : --------------------------------------- Location Node : 1 Shelf Address : 255 Board Address : 255 Equipment Address : 255 Set Type + SIP extension Entity Number : 1 Set Function + Default Profile Name : --------------- Key Profiles + None Domain Identifier : 0 Language ID : 1 Secret Code : **** SIP Gateway configuration as used in tests: Review/Modify: SIP Gateway Node Number (reserved) : 1 Instance (reserved) : 1 Instance (reserved) : 1 SIP Subnetwork : 4 SIP Trunk Group : 4 IP Address : 172.25.32.252 Machine name - Host : vega SIP Proxy Port Number : 5060 SIP Subscribe Min Duration : 1800 SIP Subscribe Max Duration : 86400 Session Timer : 180 Min Session Timer : 180 Session Timer Method + RE_INVITE DNS local domain name : lab.ts DNS type + DNS A SIP DNS1 IP Address : --------------------------------------- SIP DNS2 IP Address : --------------------------------------- SDP IN 180 + True Cac SIP-SIP + True


SIP Registrar configuration as used in tests: Review/Modify: SIP Registrar Node Number (reserved) : 1 Instance (reserved) : 1 Instance (reserved) : 1 SIP Min Expiration Date : 60 SIP Max Expiration Date : 86400 List of prefixes and suffixes defined on OmniPCX TSS lab system (default): +-----------------+----------------------------------------------------------+ |dir |mean | +-----------------+----------------------------------------------------------+ |400 |Set_In/Out_of_service | |401 |Recordable_Voice_Guides | |402 |Park_Call/Retrieve | |403 |Charging_meter_readout | |404 |Associated_Set_No_Modif | |405 |Password_modification | |406 |Redial_last_number | |407 |Night_service_answering | |408 |Contrast_programmation | |409 |Secret/Identity | |41 |Forward_cancellation | |42 |Do_not_disturb | |43 |Voice_Mail | |44 |Canc_auto_call_back_on_busy | |45 |PadLock | |46 |Consult_Call_back_list | |470 |Waiting_call_consultation | |471 |Business_account_code | |472 |Consult_Messages | |473 |Paging_call_answer | |474 |Language | |480 |Set_group_entry | |481 |Set_group_exit | |482 |Switch_off_Message_LED | |483 |Mask_Remote_Calling_Identity | |484 |Cancel_Remote_forward | |485 |Overfl_busy_to_assoc_set | |486 |Overf_busy/no_repl_assoc_set | |487 |Recording_Conversation | |490 |Ubiquity_Mobile_Programming | |491:493 |Ubiquity_Services_Pfx | |495 |Ubiquity_Assistant | |500 |Last_Caller_Call_back | |501 |Remote_forward | |502 |Overflow_on_associated_set | |503 |Cancel_Overfl_on_assoc_set | |504 |Protection_against_beeps | |505 |Substitution | |506 |Wake_up/appointment_remind | |507 |Cancel_Wake_up | |508 |Forward_cancel_by_destinat | |509 |Meet_me_Conference | |51 |Immediate_forward | |52 |Immediate_forward_on_busy | |53 |Forward_on_no_reply | |54 |Forward_on_busy_or_no_reply | |55 |Direct_call_pick_up | |56 |Group_call_pick_up | |570 |Voice_Mail_Deposit | |580 |Tone_test | |581 |Personal_directory_Progr | |582 |Personal_Directory_Use |


|583 |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 |


Appendix E: Partner escalation process

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


Appendix F: AAPP program, documentation and technical assistance

Alliance & Application Partner Program (AAPP)

Complete e-business solutions at your disposal

The Alliance & Application Partner Program is designed to support companies that develop communication applications for the enterprise market, based on Alcatel-Lucent's Omni product family. The program provides tools and support for developing, verifying and promoting compliant third-party applications that complement Alcatel-Lucent's Omni-based products. Alcatel-Lucent facilitates market access for compliant applications.

The Alliance & Application Partner Program (AAPP) has two main objectives: Provide easy interfacing for Alcatel-Lucent communication products:

Alcatel-Lucent's communication products for the enterprise market include infrastructure elements, platforms and software suites. To ensure easy integration, the AAPP provides a full array of standards-based application programming interfaces and fully documented proprietary interfaces. Together, these enable third-party applications to benefit fully from the potential of Alcatel-Lucent products.

Test and verify a comprehensive range of third-party applications: to ensure proper inter-working, Alcatel-Lucent tests and verifies selected third-party applications that complement its portfolio. Successful candidates, which are labeled Alcatel-Lucent Compliant Application, come from every area of voice and data communications.

The Alliance & Application Partner Program covers a wide array of third-party applications/products designed for voice-centric and data-centric networks in the enterprise market, including terminals, communication applications, mobility, management, security,

Web site

If registered Alliance & Application Partner, you can access the AAPP website at this URL: http://www.applicationpartner.alcatel-lucent.com

Alcatel-Lucent.com

You can access the Alcatel-Lucent website at this URL: http://www.Alcatel-Lucent.com/


Alcatel-Lucent documentation

Alliance & Application Partner Program (AAPP)

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

Alcatel-Lucent Business Partner Program (ABPP)

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

If registered Alcatel-Lucent Business Partners, you can access to an exciting on-line resource centre with a wealth of information on all product lines at this URL: http://www.businesspartner.Alcatel-Lucent.com


Technical assistance

In order to guide you in your purchasing decisions and provide you with assistance for updating our Communication Server and Networking Infrastructure products and for commercial development, Alcatel-Lucent has created the SUPPORT CENTER. The SUPPORT CENTER is responsible for the management and routing of all your requests. It includes e-Support and a Contact Centre reserved for registered Alliance & Application Partner and Alcatel-Lucent Business Partners.

The Contact Centre is open 24 hours a day; 7 days a week and is available in 5 languages. This Call Centre has a team of 15 people and handles 10; 000 requests per month.

e-Support from the Alliance & Application Partner Web site (if registered Alliance & Application Partner): http://www.applicationpartner.alcatel-lucent.com click the Partner Center link and then Support

e-Support from the Alcatel-Lucent Business Partners Web site (if registered Alcatel-Lucent Business Partners): http://www.businesspartner.Alcatel-Lucent.com click the e-Support link and then e-Service Request

e-mail: [email protected] Fax number: +33 (0) 3 90 67 73 45 Telephone numbers:

Alcatel-Lucent Business Partner Contact Center:

France : 0 811 900 110 French agent Austria : 0 810 810 012 German agent Denmark : 70 11 21 09 English agent Germany : 0 1 803 000 680 German agent Ireland : 1 890 925 039 English agent Italy : 848 800 389 Italian agent UK : 0 845 601 4101 English agent Spain : 901 120 085 Spanish agent Switzerland : 0 844 850 588 German agent

For other countries:

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


Alcatel-Lucent training

Technological innovative cycles are quickening and your customers are more and more demanding regarding the quality of services. In order to meet these requirements, you have to invest in skills: a key success factor for services.

If registered Alcatel-Lucent Business Partners, you can access to the training part at this URL: http://www.businesspartner.Alcatel-Lucent.com and then click the Training link.

Our vision of learning services is described in the Services Portfolio section. The Certification section gives you some statistics and details on how training curricula are designed to match certification levels. All updated training curricula and assessment tools are available in the Curricula & Catalogues section. The Schedule section is regularly updated to show forthcoming training sessions over the world. The How to Enrol section provides you with the registration procedure and the Alcatel-Lucent University Customer Service list of contacts world wide. Last but not least, find statistics and reports of what you think about our training services in the Customer Satisfaction section.

The Alcatel-Lucent commitment : enabling you to optimize your training investments.


Appendix G: Alcatel-Lucent escalation process in case of problem with a certified external application (referenced in the AAPP)

Introduction

The purpose of this document is to define the split of responsibilities and the escalation process to be applied by the Business Partners when facing a problem with a solution involving an Alcatel-Lucent platform and an external application with a valid Alcatel-Lucent Inter-working report.

As for other Alcatel-Lucent equipment, the Alcatel-Lucent business partner is the only one facing the end-customer for support or maintenance. The Business partner will open cases (service request) either on Alcatel-Lucent side or on Application Partner side depending on the nature of the issue. Expert from both companies will collaborate to provide the best and quickest correction.

Partners

answers

CustomerBusiness Partner

Application Partner Support

TechnicalIssues

Alcatel Technical Support

SR

SR = Service Request

SR

Answer

Answer

Experts Exchanges

In cases of the problem has not been sent to the right organization: Please open a SR into our partner database by mentioning our SR number

Partners

answers

CustomerBusiness Partner

Application Partner SupportApplication Partner Support

TechnicalIssues

Alcatel Technical SupportAlcatel Technical Support

SR

SR = Service Request

SR

Answer

Answer

Experts Exchanges

In cases of the problem has not been sent to the right organization: Please open a SR into our partner database by mentioning our SR number


General Rules

The following general rules are applied:

Only certified AAPP applications are officially supported by Alcatel-Lucent

The certification is based on tests suite passed by Alcatel-Lucent and the Application Partner and the result is consigned into an Inter-Working Report (IWR) validated by the two parties.

The IWR is available on the AAPP Web site.

Only the major releases of both parties are certified. Certification tests are usually not performed for intermediate versions. Only the existence of the IWR in the AAPP Web site for the right Alcatel-Lucent release is the guarantee that the application has been certified with this Alcatel-Lucent release.

If the IWR for the Alcatel-Lucent release is not available, Alcatel-Lucent doesnt engage any responsibility. In that case, please contact the central Pre-Sales team.

The existence of the IWR engages Alcatel-Lucent and the Application Partner. Both parties are engaged, not exclusively Alcatel-Lucent (see the section escalation process).

Warning: The possibility to configure the Alcatel-Lucent PBX with ACTIS quotation tool in order to interwork with an external application, is not a guarantee of the availability of the solution. Please check the availability of the Inter-Working Report on AAPP web site.


The escalation process

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

Case 1: the responsibility can be established 100% on Alcatel-Lucent side In that case, the problem must be escalated by the Business Partner to the Alcatel-Lucent Hot-line via the standard process: open a ticket (Service Request SR)

Case 2 : the responsibility can be established 100% on Application Partner side In that case, the problem must be escalated directly to the partner by opening a ticket trough the Partner Hotline. In general, the process to be applied for the partner side is described in the IWR.

Case 3 : the responsibility can not be established In that case the following process applies:

1) The Application Partner shall be contacted first by the Business Partner or the party responsible for that Application for an analysis of the problem. Alcatel-Lucent has to be involved solely if the application partner demonstrate, with traces, after reproduction of the problem, that the defect which has generated the end-users demand of support is coming from the equipment provided by Alcatel-Lucent or if he needs support of Alcatel-Lucent.

2) The Business partner will escalate the problem to the Alcatel-Lucent Hot-line if the Application Partner has demonstrated a problem on Alcatel-Lucent side or if the Application Partner (not the Business Partner) needs the involvement of Alcatel-Lucent.

In that case, the Business Partner must provide the reference of the Case Number on Application Partner side. The Application Partner must provide to Alcatel-Lucent the results of its investigations, traces, etc, related to this Case Number.

Alcatel-Lucent reserves the right to close the case opened on his side if the investigations made on Application Partner side are insufficient or do no exist.

Note: Involvement of the Business Partner is mandatory because the access to the Alcatel-Lucent Platform (remote access, login/password) is under the Business Partner responsibility.

audiocodes-mp202-v2.6.4_omnipcxenterprise-r9.0_iwr_ed02.pdf

Documents

application partner

alcatellucent product

application partner

application information

alcatellucent issues

audiocodes application

partners application

partner contact