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Alcatel-Lucent Application Partner Program

Inter-Working Report

Partner: AudioCodes
Application type: Session Border Controller
Application name:
Mediant 1000 SBC /Microsoft Lync 2013
Alcatel-Lucent Platform: OmniPCX Enterprise

The product and release listed have been tested with the Alcatel-Lucent Communication Platform and the release specified
hereinafter. The tests concern only the inter-working between the AAPP members product and the Alcatel-Lucent
Communication Platform. The inter-working report is valid until the AAPP members product issues a new major release of
such product (incorporating new features or functionality), or until Alcatel-Lucent issues a new major release of such
Alcatel-Lucent product (incorporating new features or functionalities), 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 AAPP MEMBERS 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 AAPP MEMBER OR ANY OTHER PARTY ARISING
FROM OR RELATED IN ANY MANNER TO THIS CERTIFICATE.

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Copyright 2014 Alcatel-Lucent, All rights reserved
Tests identification

Date of the tests March 2014

Alcatel-Lucents representative RESIDORI Florian


Partners representative LECLERCQ Serge

Alcatel-Lucent Communication OmniPCX Enterprise


Platform (OmniPCX
4400/Enterprise, OmniTouch,
OmniPCX Office, ...)
Alcatel-Lucent compatibility
R10.1 J2.501.20b
release
AudioCodes : 6.60A.231.010
Partners application version
Lync : R. 2013 5.0.8308.0
SBC
Application Category
Collaboration & UC

Author(s): LECLERCQ Serge


Reviewer(s): RESIDORI Florian, SPARD Jean Paul

Historic

Edition 1: creation of the document November 2011


Edition 2: Update with Lync 2013 October 2013
Edition 3: Configuration section update March 2014
Edition 4: Configuration section update October 2014
Edition 5: Configuration section update November 2014

Test results
Passed Refused Postponed

Passed with restrictions

Refer to the section 6 for a summary of the test results.

IWR validity extension


Validity extended to OXE R11.0 October 2013
Extended to all Audiocodes Mediant SBC range (2600, 3000 and 4000)

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Copyright 2014 Alcatel-Lucent, All rights reserved
Company Contact Information

Contact name: Serge Leclercq


Title: Field Application Engineer

Address 1: 104 Avenue Albert 1er Les Passerelles


Address 2:
City: Rueil-Malmaison
State: France
Zip: 92500
Country: France
Country code: +33

Phone: +336 42 18 15 60

Web address: http://www.AudioCodes.com


E-mail: serge.leclercq@audioCodes.com

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Copyright 2014 Alcatel-Lucent, All rights reserved
TABLE OF CONTENTS

1 Introduction .................................................................................................. 6
1.1 GLOSSARY ................................................................................................... 6
2 Validity of the InterWorking Report ..................................................................... 7
3 Limits of the Technical support .......................................................................... 8
3.1 CASE OF ADDITIONAL THIRD PARTY APPLICATIONS ........................................................... 8
4 Application information.................................................................................... 9
5 Tests environment ........................................................................................ 10
5.1 GENERAL ARCHITECTURE ...................................................................................10
5.2 SOFTWARE CONFIGURATION ................................................................................11
6 Summary of test results ................................................................................. 13
6.1 SUMMARY OF MAIN FUNCTIONS SUPPORTED .................................................................13
6.2 SUMMARY OF PROBLEMS ....................................................................................13
6.2.1 OXE ........................................................................................................................................... 13
6.2.2 Audicodes/Lync ...................................................................................................................... 13
6.3 SUMMARY OF LIMITATIONS ..................................................................................13
6.3.1 OXE ........................................................................................................................................... 13
6.3.2 AudioCodes/Lync .................................................................................................................... 13
6.4 NOTES, REMARKS...........................................................................................14
7 Test Scenarios ............................................................................................. 15
7.1 TEST PROCEDURE ..........................................................................................15
7.2 RESULT TEMPLATE..........................................................................................15
8 Testing ....................................................................................................... 16
8.1 BASICS OUTGOING CALLS: OXE TO LYNC...................................................................16
8.1.1 Calls from an OXE user set to a Lync user ......................................................................... 16
8.2 INCOMING CALLS: LYNC => OXE ...........................................................................18
8.2.1 Reception of calls .................................................................................................................. 18
8.3 FEATURES DURING CONVERSATION ..........................................................................20
8.3.1 Hold, Consultation call and broker call ............................................................................. 20
8.3.2 Transfer call ........................................................................................................................... 22
8.3.3 Conference .............................................................................................................................. 24
8.4 VOICE MAIL ................................................................................................25
8.4.1 Test Objectives....................................................................................................................... 25
8.4.2 Test Results ............................................................................................................................. 25
8.5 ATTENDANT ................................................................................................26
8.5.1 Test Objectives....................................................................................................................... 26
8.5.2 Test Results ............................................................................................................................. 26
8.6 DEFENSE / RECOVERY ......................................................................................27
8.6.1 Test Objectives....................................................................................................................... 27
8.6.2 Test Results ............................................................................................................................. 27
9 Appendix A : AudioCodes Mediant 1000 MSBG Configuration ................................... 28
9.1 GETTING STARTED .........................................................................................28
9.2 CONFIGURATION PROCEDURE...............................................................................29
9.2.1 Configure IP Address ............................................................................................................. 29
9.2.2 Enable the SBC Application .................................................................................................. 29
9.2.3 Add a SIP Interface ................................................................................................................ 29
9.2.4 Add IP groups .......................................................................................................................... 30
9.2.5 Configure IP Profiles ............................................................................................................. 33
9.2.6 Configure the Proxy Set table ............................................................................................. 36
9.2.7 Configure Routing .................................................................................................................. 38
9.2.8 Configure Reasons for Alternative Routing ....................................................................... 38
9.2.9 Sip Header Manipulations ..................................................................................................... 39
9.2.10 Reset the Gateway................................................................................................................. 41
10 Appendix B: Alcatel-Lucent Communication Platform: configuration requirements .... 42
11 Appendix C: Partner escalation process ........................................................... 49
12 Appendix D: AAPP program .......................................................................... 50

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12.1 ALCATEL-LUCENT APPLICATION PARTNER PROGRAM (AAPP) ...........................................50
12.2 ALCATEL-LUCENT.COM ..................................................................................51
13 Appendix E: AAPP Escalation process .............................................................. 52
13.1 INTRODUCTION ..........................................................................................52
13.2 ESCALATION IN CASE OF A VALID INTER-WORKING REPORT ..............................................53
13.3 ESCALATION IN ALL OTHER CASES ........................................................................54
13.4 TECHNICAL SUPPORT ACCESS ............................................................................55

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1 Introduction
This document is the result of the certification tests performed between the AAPP members
application and Alcatel-Lucents platform.

It certifies proper inter-working with the AAPP members application.

Information contained in this document is believed to be accurate and reliable at the time of printing.
However, due to ongoing product improvements and revisions, Alcatel-Lucent cannot guarantee
accuracy of printed material after the date of certification nor can it accept responsibility for errors or
omissions.
Updates to this document can be viewed on:

- the Technical Support page of the Enterprise Business Portal


(https://businessportal.alcatel-lucent.com) in the Application Partner Interworking Reports
corner (restricted to Business Partners)

- the Application Partner portal (https://applicationpartner.alcatel-lucent.com) with free


access.

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 The transferor waits to be in conversation state with the target before
conversation 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
SBC Session Border Controller

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2 Validity of the InterWorking Report
This InterWorking report specifies the products and releases which have been certified.

This inter-working report is valid unless specified until the AAPP member issues a new major
release of such product (incorporating new features or functionalities), or until Alcatel-Lucent
issues a new major release of such Alcatel-Lucent product (incorporating new features or
functionalities), whichever first occurs.

A new release is identified as following:


a Major Release is any x. enumerated release. Example Product 1.0 is a major product
release.
a Minor Release is any x.y enumerated release. Example Product 1.1 is a minor
product release

The validity of the InterWorking report can be extended to upper major releases, if for example the
interface didnt evolve, or to other products of the same family range. Please refer to the IWR
validity extension chapter at the beginning of the report.

Note: The InterWorking report becomes automatically obsolete when the mentioned product
releases are end of life.

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3 Limits of the Technical support
Technical support will be provided only in case of a valid InterWorking Report (see chapter 2
Validity of the InterWorking Report) and in the scope of the features which have been certified.
That scope is defined by the InterWorking report via the tests cases which have been performed,
the conditions and the perimeter of the testing as well as the observed limitations. All this being
documented in the IWR. The certification does not verify the functional achievement of the AAPP
members application as well as it does not cover load capacity checks, race conditions and
generally speaking any real customer's site conditions.

Any possible issue will require first to be addressed and analyzed by the AAPP member before
being escalated to Alcatel-Lucent.

For any request outside the scope of this IWR, Alcatel-Lucent offers the On Demand Diagnostic
service where assistance will be provided against payment.

For more details, please refer to Appendix F AAPP Escalation Process.

3.1 Case of additional Third party applications


In case at a customer site an additional third party application NOT provided by Alcatel-Lucent is
included in the solution between the certified Alcatel-Lucent and AAPP member products such as a
Session Border Controller or a firewall for example, Alcatel-Lucent will consider that situation as to
that where no IWR exists. Alcatel-Lucent will handle this situation accordingly (for more details,
please refer to Appendix F AAPP Escalation Process).

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4 Application information

Application type: SIP SBC Mediant 1000 MSBG Customer Relationship Center includes SIP SBC
Mediant 1000 MSBG

Application commercial name: SBC Mediant 1000 MSBG

Application version: 6.60A.231.010

Interface type: SIP/IP

Brief application description:

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

Figure 1 Global architecture

Microsoft
Lync 2013
server

OXE
Lync
Phones
users
AudioCodes Mediant 1000 : 1
Microsoft Microsoft
Domain Mediation
Controller server Alcatel-Lucent
OmniPCX Enterprise

The Microsoft Lync 2013 Server and Mediation Server connect to the Sip SBC Mediant 1000 MSBG
which is connected to the OmniPCX Enterprise via a private SIP Trunk.

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5 Tests environment

5.1 General architecture


All test environments, including Lync 2013 PBX/endpoints, OXE R10 with IPTouch endpoints and
Audiocodes SBC are hosted in Audiocodes laboratory.

Figure 2 Tests environnent

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Mediant 1000 MSBG platform:
.
See Appendix A: Application description for details

Lync Server 2013:

5.2 Software configuration


Alcatel Communication Platform: OmniPCX Enterprise R10.1 J2.501.20b

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Copyright 2014 Alcatel-Lucent, All rights reserved
o Spatial redundancy
o OXE network with two different nodes connected with an ABC/IP link
Mediant 1000 : M1KB_SIP_F6.60A.231.010
Microsoft Lync RTM: Release 2013 5.0.8308.0

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6 Summary of test results

6.1 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).
Three party conferences initiated from Lync set and OXE set.

6.2 Summary of problems

6.2.1 OXE
None

6.2.2 Audicodes/Lync
None

6.3 Summary of limitations

6.3.1 OXE
None

6.3.2 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.

After an OXE CPU swichover to a standby CPU, an existing call cannot evolve (stet
changed to on hold, transferred)

After an OXE CPU swichover to a standby CPU resolved by spatial redundancy mechanism,
a call cannot be established with the new main call server before the next DNS request.
These DNS request are configured to be sent every 10 seconds on the Median 1000
(parameter PROXYIPLISTREFRESHTIME=10).

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6.4 Notes, remarks
In all Mediant GW and Lync configuration, each Mediant 1000 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 Mediant 1000 supports 150 simultaneous sessions per CPU or 60
simultaneous sessions in case of audio transcoding (for example law change : G711a-
>G711, codec change : G729->G711).

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7 Test Scenarios

7.1 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.

7.2 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 The application
waits for PBX timer
OK
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
No indication, no
NOK
error message

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8 Testing

8.1 Basics Outgoing calls: OXE to Lync

8.1.1 Calls from an OXE user set to a Lync user


8.1.1.1 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.

8.1.1.2 Test procedure

Step Action Result Comment

1 Call to wrong number OK


2 Call to free user OK
3 Call in conversation; DTMF No called name display on OXE phone,
reception ; Calling / caller line because Lync does not send the user
identity ; display name to the AudioCodes (only the
number). The trunk name with the
OK_but
called number is displayed on OXE
phone. On Lync user, the display name
is OK.
DTMF sent using RFC2833.
4 Call to busy user (mono-line /
N/A Lync user is never in busy mode
multi-line)
5 Call to user in Out of Service no answer display on caller:
OK
state Temporary Unavailable in the trace
6 Call to user in Do not Disturb no answer display on caller:
OK
state Temporary Unavailble in the trace
7 Call to Data terminal N/A
8 Call to forwarded user (locally) OK Same display issue as step 3
(immediate forward)
OXE user1 call Lync user1 who is in
immediate forward to Lync user2.
9 External call (T2) to Lync user. No PSTN trunk connected to the OXE
Check that external call back N/A
number is shown correctly.
10 Call back feature :
OXE user1 call Lync user1 and hung Test done using Missed call list on
OK
up. Lync user1 can press call back Audiocodes HD420.
to call OXE user1
11 No answer of the called party
(Forward no reply)
OXE phone call Lync user and Lync Impossible to configure a call forward
N/A
user does not take the call and on no reply on Lync 2013
after some minute the call is
forwarded to another Lync user

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12 Call from UA/TDM to Lync user Test done with a digital phone
connected to OXE Digital card as no
OK PSTN trunk connected to OXE
Display from digital phone is OK

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8.2 Incoming calls: Lync => OXE

8.2.1 Reception of calls


8.2.1.1 Test objectives

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.

8.2.1.2 Test procedure

Step Action Result Comment

1 Call to wrong number OK


2 Call to free user (check Calling / caller
OK
line identity ; display, DTMF)
3 Call to busy user (mono-line / multi- Display Extension Busy on Lync
OK
line)(Local/Network) phone
4 Call to user in Out of Service state
OK
(Local/Network)
5 Call to user in Do not Disturb state OK
6 Call to Data terminal N/A
7 Call to forwarded user (immediate Lync phone displays the
forward) forwarded OXE user information,
Lync user1 call OXE user1 who is in not final OXE destination.
immediate forward to OXE user2. OK_but AudioCodes let the updated PAI
(Local/Network) (with final destination
information) in 200OK/SDP but
not interpreted by Lync phone
8 Call Back feature OK
10 Lync user Calls to external number (via
T2 loopback)
N/A NO T2 link connected to OXE
(Check ring back tone, called party
display)
12 No answer of the called party Call released by the OXE using a
OK 480 temporary unavailable after
30 seconds
13 Lync user 1 calls IP Phone which is on Lync phone displays the
forward on busy to Lync user 2 (prefix forwarded OXE user information,
52) (Local/Network) not final OXE destination.
OXE provide diversion
OK_but information in history-info
header and Audiocodes SBC adds
a Diversion header but Lync
client dont display final
destination number

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14 Lync user 1 calIs IP Phone which is on Lync phone displays the
forward on no reply to Lync user 2 forwarded OXE user information,
(Local/Network) not final OXE destination.
OXE provide diversion
OK_but information in history-info
header and Audiocodes SBC adds
a Diversion header but Lync
client dont display final
destination number
15 Lync user1 calIs IP Phone which is in Lync phone displays the
immediate forward mode to Lync user2 forwarded OXE user information,
(51) (Local/Network) not final OXE destination.
OXE provide diversion
OK_but information in history-info
header and Audiocodes SBC adds
a Diversion header but Lync
client dont display final
destination number
16 Oxe Phone call Lync user then Oxe Anonymous is displayed on the
OK
phone display secret identity Lync phone
17 Lync user put OXE phone on hold Local music on hold played from
OK
Lync client

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8.3 Features during conversation

8.3.1 Hold, Consultation call and broker call

8.3.1.1 Test objectives

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.

8.3.1.2 Test procedure

Step Action Result Comment

1 A=Lync user, B=IPPhone1, C=IPPhone2


(Local/Network)
1.1 Hold state request
OK
Lync user->IPPhone1, Lync user on hold
1.2 Consultation call request
OK
IPPhone1->IPPhone2
1.3 Broker request
OK
IPPhone1->Lync user, IPPhone2 on hold
2 A=PSTN, B=IPPhone1, C=Lync user
N/A No PSTN phone
(Local/Network)
2.1 Hold state request
N/A
PSTN->IPPhone1, PSTN on hold
2.2 Consultation call request
N/A
IPPhone1->Lync user
2.3 Broker request
N/A
IPPhone1->PSTN, Lync user on hold
3 A=PSTN, B= Lync user , C= IPPhone1
N/A No PSTN phone
(Local/Network)
3.1 Hold state request
PSTN-> Lync user, N/A
PSTN on hold
3.2 Consultation call request
N/A
Lync user-> IPPhone1
3.3 Broker request
N/A
Lync user->PSTN, IPPhone1 on hold
4 A=IPPhone1, B=IPPhone2, C=Lync user
(Local/Network)
4.1 Hold state request
OK
IPPhone1->IPPhone2, IPPhone1 on hold
4.2 Consultation call request
OK
IPPhone2->Lync user
4. 3 Broker request
OK
IPPhone2->IPPhone1, Lync user on hold
5 A=IPPhone1, B=Lync user, C=IPPhone2
(Local/Network)
Hold state request
OK
5.1 IPPhone1->Lync user, IPPhone1 on hold
5.2 Consultation call request
OK
Lync user->IPPhone2

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5.3 Broker request
OK
Lync user->IPPhone1, IPPhone2 on hold
6 A=Lync user1, B=IPPhone, C= Lync user2
OK
(Local/Network)
Hold state request Lync User 1 heard OXE
OK
6.1 Lync user1->IPPhone, Lync user1 on hold MoH
6.2 Consultation call request Lync User 2 heard OXE
OK
IPPhone->Lync user2 MoH
6.3 Broker request
OK
IPPhone->Lync user1, Lync user2 on hold
7 A=Lync user1, B=IPPhone, C=UA/TDM set
(Local/Network)
7.1 Hold state request
OK
IPPhone->Lync user1, IPPhone on hold
Consultation call request
OK
7.2 Lync user1->UA/TDM set
Broker request
OK
7.3 Lync user1->IPPhone, UA/TDM set on hold
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 OK
hold
8.2 Consultation call request
OK
Lync user1 > UA/TDM set2
8.3 Broker request
Lync user1->UA/TDM set 1 OK
UA/TDM set2 on hold
9 A=IPPhone, B=Lync user1, C=Lync user2
(Local/Network)
9.1 Hold state request
OK
IPPhone->Lync user1, IPPhone on hold
9.2 Consultation call request
OK
Lync user1->Lync user2
9.3 Broker request
OK
Lync user1->IPPhone, Lync user2 on hold

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8.3.2 Transfer call

8.3.2.1 Test objectives

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.

8.3.2.2 Test procedure

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)

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Test Action Result Comment
A B C
Transf Transf Transfer
eree eror Target
1 Unattended transfers not available from an
LYNC OXE OXE N/A
OXE user.
2 Unattended transfers not available from the
OXE LYNC OXE N/A
Lync client.
3 Unattended transfers not available from an
OXE OXE LYNC N/A
OXE user.
4 Unattended transfers not available from the
OXE LYNC LYNC N/A
Lync client.
5 Unattended transfers not available from an
LYNC OXE LYNC N/A
OXE user.
6 Unattended transfers not available from the
LYNC LYNC OXE N/A
Lync client.

Semi attended Transfer (On Ringing)

Test Action Result Comment


A B C
Transf Transf Transfer
eree eror Target
1 LYNC OXE OXE OK
2 OXE LYNC OXE OK
3 OXE OXE LYNC OK
4 OXE LYNC LYNC OK
5 LYNC OXE LYNC OK
6 LYNC LYNC OXE OK

Attended Transfer (in conversation)

Test Action Result Comment


A B C
Transf Transf Transfer
eree eror Target
1 LYNC OXE OXE OK
2 No display update on A.
OXE LYNC OXE OK
3 No display update on A.
OXE OXE LYNC OK
4 No display update on A.
OXE LYNC LYNC OK
5 No display update on A.
LYNC OXE LYNC OK
6 No display update on A.
LYNC LYNC OXE OK

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8.3.3 Conference
8.3.3.1 Test objectives

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.

8.3.3.2 Test procedure

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


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

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8.4 Voice Mail

8.4.1 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.

8.4.2 Test Results

Test Action Result Comment


1 IPTouch phone leave as a VoiceMail
Not
message for the Lync user No exchange voicemail unavailable
tested
Check that MWI is OK on Lync user
2 Message consultation by Lync user Not No exchange voicemail unavailable
tested
3 Password modification by Lync user Not No exchange voicemail unavailable
tested
4 Lync user call to a OXE user
OK
forwarded to Voice Mail (4645)
5 Message consultation by OXE user OK

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8.5 Attendant

8.5.1 Test Objectives


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

8.5.2 Test Results

Test Action Result Comment


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

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8.6 Defense / Recovery

8.6.1 Test Objectives


Check the reaction in case of a PBX reboot, switch-over or link failure.

AudioCodes gateway redundancy is also tested.

8.6.2 Test Results

Step Action Result Comment

Temporary Data Network Link down with Existing calls are stopped.
1 the PBX and Mediant 1000 MSBG OK Establishing new call is possible
when the link is reestablished.
Spatial redundancy IP Method : CPU An existing call cannot evolve
2 switchover with SIP communication OK,but (stet changed to on hold,
transferred)
Spatial redundancy DNS method A call cannot be established after
(delegation on a third party DNS server) : the switchover with the new main
CPU switchover without SIP communication call server before the next DNS
request. These DNS request are
3 OK,but
configured to be sent every 10
seconds on the Median 1000
(parameter
PROXYIPLISTREFRESHTIME).
Spatial redundancy DNS method : CPU A second call cannot be
switchover with SIP communication established after the switchover
with the new main call server
before the next DNS request.
4 OK,but
These DNS request are configured
to be sent every 10 seconds on
the Median 1000 (parameter
PROXYIPLISTREFRESHTIME).
Switchover to Passive Call Server (PCS). (IP
link to main/stdby OXE call servers down)
5 N/A

Switchover to a backup AudioCodes The gateway status is checked


gateway. Stop the main AudioCodes every 10 seconds. After this
6 gateway; verify that a call is possible with OK
period, you can establish a new
the backup AudioCodes gateway. call.

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9 Appendix A : AudioCodes Mediant 1000 MSBG
Configuration
9.1 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 9-1: Navigation Tree in Basic and Full View

For more information, see the Mediant 1000 MSBG Users manual.

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9.2 Configuration Procedure
This section describes the Mediant 1000 MSBG Configuration procedure.

9.2.1 Configure IP Address


To configure IP-Address:
1. Open the 'IP Settings' page (Configuration tab > VoIP menu > Network > IP
Settings).
Figure 9-2: IP Settings

2. Set the following parameters:


IP-Address: <Gateway IP-Address> (e.g., 172.17.165.3).
Prefix Length: The Subnet Mask in bits (e.g., 24 for 255.255.255.0).
Gateway: <Gateway Default Gateway> (e.g., 172.17.165.1).
DNS server : <Primary DNS Server IP Address> (e.g 172.17.165.1)

9.2.2 Enable the SBC Application


To configure applications:
1. Open the 'Applications Enabling' page (Configuration tab > VoIP menu >
Applications Enabling > Applications Enabling).

Figure 9-3: Applications Enabling

2. SBC Application: Enable.

Note: Reset with BURN to FLASH is Required.

9.2.3 Add a SIP Interface


To configure a SIP interface:
1. Open the Sip Interface Table page (Configuration tab > VoIP menu > Control
Network >SIP Interface Table).

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Figure 9-4: SIP Interface Table

2. Set the following parameters:


Network Interface: refer to the network interface name given in 7.2.1 (e.g
Voice)
Application Type: SBC.
UDP Port : 5060
TCP Port : 5068
TLS Port : 5067

9.2.4 Add IP groups


To configure IP group 1 (Lync 2013):
1. Open the 'IP Group Table' page (Configuration tab > VoIP menu > Control
Network> IP Group Table).

Figure 9-5: IP Group 1

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2. Set the ProxySet ID to 1
3. Set the IP Profile ID to 1

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To configure IP group 2 (OXE) :
4. Open the 'IP Group Table' page (Configuration tab > VoIP menu > Control
Network> IP Group Table).

Figure 9-6: IP Group 2

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5. Set the ProxySet ID to 2
6. Set the Sip Group Name to the ip address of the OXE (e.g 172.17.165.12)
7. Set the IP Profile ID to 2

9.2.5 Configure IP Profiles


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

Figure 9-7: IP Profile 1 Settings

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This table resume parameters that are mandatory to interconnect with Lync 2013:
Parameter Setting
Media Security Behavior SRTP: If RTP flow encryption is activated on
Lync
RTP: if no media encryption is configured on
Lync server
SBC Remote Early Media RTP Delayed (mandatory because the Lync Server
2013 does not immediately send RTP to the
remote side if it sends a SIP 18x response)

RFC 2833 Behavior Extend (in case the OXE does not send RFC
2833 in SDP)

SBC Remote Update Support Supported Only After Connect

SBC Remote Re-Invite Support Supported Only With SDP

SBC Remote Refer Behavior Handle Locally (mandatory because Lync


Server 2013 does not support receive Refer)

SBC Remote 3xx Behavior Handle Locally (mandatory because Lync


Server 2013 does not support receive 3xx)

SBC Remote Delayed Offer Support Not Supported

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To configure IP Profile Settings 2 :
2. Open the 'IP Profile Settings' page (Configuration tab > VoIP menu > Coders
And Profiles > IP Profile Settings).

Figure 9-8: IP Profile 2 Settings

This table resume parameters that are mandatory to interconnect with Lync 2013:
Parameter Setting
SBC Remote Refer Behavior Handle Locally (mandatory because Lync
Server 2013 does not support receive Refer)

Media Security Behavior RTP

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9.2.6 Configure the Proxy Set table
To configure Proxy Set 1 :
1. Open the Proxy Set Table page (Configuration tab > VoIP menu > Control
Network> ProxySet Table).
Figure 9-9: ProxySet 1

To configure Proxyset 2 :
2. Open the Proxy Set Table page (Configuration tab > VoIP menu > Control
Network> ProxySet Table).

Figure 9-10: ProxySet 2

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To configure Proxy & Registration:

1. Open the 'Proxy & Registration' page (Configuration tab > VoIP menu > SIP
Definitions > Proxy & Registration).
2.
Figure 9-11: Proxy & Registration

3. Set the following parameters:


Use Default Proxy: No.
Redundant Routing Mode: Proxy.
Gateway Name: Gateway FQDN Name (e.g., ALE-GW.tr.local).

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9.2.7 Configure Routing
To configure IP to IP Routing:
1. Open the 'IP to Trunk Group Routing' page (Configuration tab > VoIP menu >
SBC > Routing SBC > IP to IP Routing Table).
2.
Figure 9-12: IP to IP Routing (example)

3. Configure according to the routing topology.


4. In the example above, all the calls from IP Group 1 (the Lync server) are routed
to the destination IP Group 2 (OXE) in the line index 1.
5. Line index 2 is the reverse route, where all calls from IP Group 2 (OXE) are
routed to IP Group 1 (Lync)
6. Line index 0 is used to acknowledge locally the sip options received from the
Lync server

9.2.8 Configure Reasons for Alternative Routing


To configure Reasons for Alternative Routing:
1. Open the 'Dest Number IP->Tel' page (Configuration tab > VoIP menu > SBC>
Routing SBC > Reasons for Alternative Routing).
Figure 9-13: Reasons for Alternative Routing

2. Set reason 1: 503.

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9.2.9 Sip Header Manipulations
To configure Sip Headers manipulations :

7. Open the 'IP to Trunk Group Routing' page (Configuration tab > VoIP menu >
Sip Definitions > Msg Policy & Manipulation > Messages Manipulation).
Figure 9-13: Messages Manipulations for OXE

Parameter Setting
Action Subject header.from.url.host
Action Type Modify
Action Value 172.17.165.3 : AudioCodes SBC IP address

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Parameter Setting
Action Subject header.p-preferred-identity.url.host
Action Type Modify
Action Value header.from.url.host

Parameter Setting
Action Subject header.p-asserted-identity.url.host
Action Type Modify
Action Value header.from.url.host

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9.2.10 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 9-3: Reset the GW

Note: Reset with BURN to FLASH is Required.

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10 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
(2)
Trunk Group ID 37
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)
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 172.17.165.3
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
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
(4)
Table Id 1
Carrier Reference 0
Command I
Associated Ext SIP gateway 1
Dialing command table (DCT) Translator/Automatic Route Selection/Dialing Command Table
(4)
Table Id 2
Carrier Reference 0
Command I
Associated Ext SIP gateway 2
ARS route list Translator/Automatic Route Selection/ARS Route list
(5)
ARS Route list 4

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Name SIP ARS
ARS route Translator/Automatic Route Selection/ARS Route list/ARS Route
ARS Route list 4
(6)
Route 1
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
Translator/Automatic Route Selection/ARS Route list/Time-based
Time-based route list route list
ARS Route list 4
Time-based Route List ID 1
Route Number 1
Barring / Discrimination (call restrictions): used to give to caller the right to call the external
dialled number
Translator/External Dialing Plan/Dialing (Numbering)
Real Discriminator Discriminator
(7)
Discriminator n 2
Name SIP
Translator/External Dialing Plan/Dialing (Numbering)
Discriminator rule Discriminator/Discriminator rule
Discriminator n 2
Call number 2
(8)
Area number 1
ARS route list number 4
Number of digits 3
Logical discriminator Entities/Discriminator Selector
Entity Number 1
(9)
Discriminator 0x 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.

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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

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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
(4)
Table Id 1
Carrier Reference 0
Command I
Associated Ext SIP gateway 1
Dialing command table (DCT) Translator/Automatic Route Selection/Dialing Command Table
(4)
Table Id 1
Carrier Reference 0
Command I
Associated Ext SIP gateway 2
ARS route list Translator/Automatic Route Selection/ARS Route list
(5)
ARS Route list 4
Name SIP ARS
ARS route Translator/Automatic Route Selection/ARS Route list/ARS Route
ARS Route list 4
(6)
Route 1
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
Translator/Automatic Route Selection/ARS Route list/Time-based
Time-based route list route list
ARS Route list 4
Time-based Route List ID 1
Route Number 1
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
2 (7)
Discriminator n
Name SIP
Translator/External Dialing Plan/Dialing (Numbering)
Discriminator rule Discriminator/Discriminator rule
Discriminator n 2
Call number 2
(8)
Area number 1
ARS route list number 4
Number of digits 3
Logical discriminator Entities/Discriminator Selector
Entity Number 1

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(9)
Discriminator 0x 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

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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 |

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11 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.

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12 Appendix D: AAPP program

12.1 Alcatel-Lucent Application Partner Program (AAPP)


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

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Web site
The Application Partner Portal is a website dedicated to the AAPP program and where the
InterWorking Reports can be consulted. Its access is free at
http://applicationpartner.alcatel-lucent.com

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

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13 Appendix E: AAPP Escalation process

13.1 Introduction

The purpose of this appendix is to define the escalation process to be applied by the Alcatel-Lucent
Business Partners when facing a problem with the solution certified in this document.

The principle is that Alcatel-Lucent Technical Support will be subject to the existence of a valid
InterWorking Report within the limits defined in the chapter Limits of the Technical support.

In case technical support is granted, Alcatel-Lucent and the Application Partner, are engaged as
following:

(*) The Application Partner Business Partner can be a Third-Party company or the Alcatel-
Lucent Business Partner itself

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13.2 Escalation in case of a valid Inter-Working Report
The InterWorking Report describes the test cases which have been performed, the conditions of
the testing and the observed limitations.

This defines the scope of what has been certified.

If the issue is in the scope of the IWR, 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.

Note: Known problems or remarks mentioned in the IWR will not be taken into account.

For any issue reported by a Business Partner outside the scope of the IWR, Alcatel-Lucent offers the
On Demand Diagnostic service where Alcatel-Lucent will provide 8 hours assistance against
payment.

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 the guarantee of the availability and
the support of the solution. The reference remains the existence of a valid InterWorking Report.

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.

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13.3 Escalation in all other cases
These cases can cover following situations:

1. An InterWorking Report exist but is not valid (see Chap 2 Validity of an Interworking
Report)

2. The 3rd party company is referenced as AAPP participant but there is no official
InterWorking Report (no IWR published on the Enterprise Business Portal for Business
Partners or on the Alcatel-Lucent Application Partner web site) ,

3. The 3rd party company is NOT referenced as AAPP participant

In all these cases, Alcatel-Lucent offers the On Demand Diagnostic service where Alcatel-Lucent will
provide 8 hours assistance against payment.

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13.4 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: Ebg_Global_Supportcenter@alcatel-lucent.com
Fax number: +33(0)3 69 20 85 85
Telephone numbers:

Alcatel-Lucent Business Partners Support Center for countries:

Country Supported language Toll free number


France
Belgium French
Luxembourg
Germany
Austria German
Switzerland
United Kingdom
Italy
Australia
Denmark
Ireland
Netherlands +800-00200100
South Africa
Norway
English
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

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