configuring the audiocodes mediant ... - devconnect program

DS; Reviewed:

SPOC 03/25/2012

Solution & Interoperability Test Lab Application Notes

©2012 Avaya Inc. All Rights Reserved.

1 of 46

AC3K_MX52_T1

Avaya Solution & Interoperability Test Lab

Configuring the AudioCodes Mediant 3000 Media Gateway

to provide Connectivity between Public Switched Telephone

Network (PSTN/T1 ISDN) and Avaya Meeting Exchange®

R5.2 Enterprise Edition using TCP - Issue 1.0

Abstract

These Application Notes present a sample configuration for a network consisting of

AudioCodes Mediant 3000 Media Gateway and Avaya Meeting Exchange® R5.2 Enterprise

Edition. The two systems are utilizing a SIP trunk (using TCP) between each other with

PSTN/T1 ISDN connectivity to Avaya Aura® Communication Manager R6.0.1.

Testing was conducted via the Internal Interoperability Program at the Avaya Solution and

Interoperability Test Lab.

DS; Reviewed:

SPOC 03/25/2012



2 of 46

AC3K_MX52_T1

1. Introduction The purpose of this interoperability Application Notes is to validate AudioCodes Mediant 3000

with Avaya Meeting Exchange® R5.2 Enterprise Edition (MX). The sample network used is

shown in Figure 1, where the Avaya Aura® Communications Manager, Avaya Meeting

Exchange® Enterprise Edition and AudioCodes Mediant 3000 Media Gateway all have a

presence on the IP Network. Additionally a PSTN/T1 ISDN line is connected between Avaya

Aura® Communications Manager and AudioCodes Mediant 3000 Media Gateway to represent

the PSTN/T1 ISDN trunk line. SIP devices are registered with Avaya Aura® Session Manager;

however the configuration of the relationship between Avaya Aura® Session Manager and

Avaya Aura® Communication Manager is not part of this document except where it impacts

directly with the purpose of this Application Notes. Initially, calls to Avaya Meeting Exchange®

R5.2 Enterprise Edition being dialed from SIP/H323/Digital handsets reach the Avaya Aura®

Communication Manger (using SIP/TCP). The Avaya Aura® Communication Manager checks

the number dialed and determines the call should be routed out over the PSTN/T1 ISDN line to

AudioCodes Mediant 3000. The configuration on AudioCodes converts the call back to SIP and

transmits to the Avaya Meeting Exchange® Enterprise Edition using TCP. Avaya Meeting

Exchange® Enterprise Edition receives the call and handles it accordingly.

Figure 1: Connection of AudioCodes Mediant 3000 and Avaya Meeting Exchange®

Enterprise Edition via PSTN, using Avaya Aura® Communications Manager.

DS; Reviewed:

SPOC 03/25/2012



3 of 46

AC3K_MX52_T1

1.1. Verified Scenarios

The following scenarios have been verified for the configuration described in these Application

Notes.

• Conference calls including various telephone types (see Figure 1) can be made using

G.711mu/A-law, G.729 and G726-32k utilizing PSTN T1 ISDN and AudioCodes

Mediant 3000

• Scan, Flex, and Direct Conference modes

• Name Recording, Conference recording and Playback (NRP)

• RFC 2833 DTMF support for all moderator and conferee commands

• In band DTMF support for all moderator and conferee commands

• Dialout via DTMF to conference participants

• Manual and automatic blast dial-out to conference participants

• Network outage failure and recovery

• Bridge process (“softms”) failure and recovery

• Session timers on Meeting Exchange

• Operator Audio Path establishment

• Line and Conference transfer by BridgeTalk Operator

• Endpoint blind transfer while in conference

• EC500 operation in conference

1.2. Test Results

As illustrated in these Application Notes, AudioCodes Mediant 3000 Media Gateway can

interoperate with Avaya Meeting Exchange® Enterprise Edition using SIP and PSTN trunks. The

Avaya Meeting Exchange® Enterprise Edition functioned well when using TCP in conjunction

with the AudioCodes Mediant 3000 with a PSTN/T1 ISDN connection to Avaya Aura®

Communication Manager. The following was observed during testing:

• The SIP trunk between Avaya Aura® Session Manager and Avaya Aura®

Communications Manager must be configured to send DTMF Out Of Band, some SIP

endpoints experience DTMF recognition issues in Avaya Meeting Exchange Enterprise

Edition when sent In-band.

DS; Reviewed:

SPOC 03/25/2012



4 of 46

AC3K_MX52_T1

1.3. Equipment and Software Validated

The following equipment and software were used for the sample configuration provided:

Equipment Software

Avaya S8800 Server Avaya Aura® Communication Manager R6.0.1 SP7

Version R016x00.1.510.1 with patch 19528

AvayaG650 Media Gateway TN2312BP HW28 FW054 (IP Interface)

TN2602AP HW08 FW061 (IP Media Processor)

TN799DP HW016 FW040 (C-LAN)

TN2464CP HW02 FW024 (DS1 Interface)

TN2214CP HW10 FW015 (Digital Line)

Avaya S8800 Server Avaya Meeting Exchange® Enterprise Edition R5.2 SP2

MX-5.2.2.0.10 with patch 5.2.2.7.1-1, using Linux 2.6.9-

67.0.1.ELsmp

Avaya Handset 9611G SIP – Firmware S96x1_SALBR6_0_3_V470

Avaya Desktop Video Device,

A175 D01A-003

SIP – Firmware V1.0.3

Avaya Handset 9640G SIP – Firmware 96xx_2_6_6_0.bin

Avaya Softphone one-X®

Communicator

Avaya Softphone one-X® Communicator

SIP – Software 6.1.2.0.6-SP2-33739

Avaya Handset 9650 H.323 – Firmware ha96xxua3_1_03_S.bin

Avaya Handset 2420 Digital - Firmware V6

AudioCodes Mediant 3000

Media Gateway

Firmware : 6.20A.027.005

DS; Reviewed:

SPOC 03/25/2012



5 of 46

AC3K_MX52_T1

2. Configure Avaya Aura® Communications Manager This document makes the following assumptions:

• Installation and basic configuration of Communication Manager has taken place,

including dial plans, signaling groups and trunk groups for SIP using TCP.

• Configuration of Session Manager and System Manager with SIP Entity links and

relevant settings to Communication Manager for TCP.

• Configuration of handsets (Digital, H.323 and SIP) has already taken place. SIP handsets

should have the SIP Proxy set to Transport Type: TCP and SIP Port:5060

For further details of these processes, please refer to Section 8. The rest of this section will cover

configuration of the PSTN circuit on Communications Manager.

2.1. Configure Customer Options

Using the command change system-parameters customer-options configure the

Communication Manager for PSTN. The necessary changes within the customer options are

made across several of the screens. Navigate to Page 2, check the field Maximum

Administered H.323 has a value greater than 0. If the value is 0, please contact the Account

Manager to discuss license options.

change system-parameters customer-options Page 2 of 11 OPTIONAL FEATURES IP PORT CAPACITIES USED Maximum Administered H.323 Trunks: 12000 0 Maximum Concurrently Registered IP Stations: 18000 2 Maximum Administered Remote Office Trunks: 12000 0 Maximum Concurrently Registered Remote Office Stations: 18000 0 Maximum Concurrently Registered IP eCons: 414 0 Max Concur Registered Unauthenticated H.323 Stations: 100 0 Maximum Video Capable Stations: 18000 1 Maximum Video Capable IP Softphones: 18000 4 Maximum Administered SIP Trunks: 24000 15 Maximum Administered Ad-hoc Video Conferencing Ports: 24000 0 Maximum Number of DS1 Boards with Echo Cancellation: 522 0 Maximum TN2501 VAL Boards: 128 0 Maximum Media Gateway VAL Sources: 250 0 Maximum TN2602 Boards with 80 VoIP Channels: 128 0 Maximum TN2602 Boards with 320 VoIP Channels: 128 1 Maximum Number of Expanded Meet-me Conference Ports: 300 0

DS; Reviewed:

SPOC 03/25/2012



6 of 46

AC3K_MX52_T1

Navigate to Page 3, verify the field ARS/AAR Dialing without FAC? is set to y.

change system-parameters customer-options Page 3 of 11 OPTIONAL FEATURES Abbreviated Dialing Enhanced List? y Audible Message Waiting? y Access Security Gateway (ASG)? n Authorization Codes? y Analog Trunk Incoming Call ID? y CAS Branch? n A/D Grp/Sys List Dialing Start at 01? y CAS Main? n Answer Supervision by Call Classifier? y Change COR by FAC? n ARS? y Computer Telephony Adjunct Links? y ARS/AAR Partitioning? y Cvg Of Calls Redirected Off-net? y ARS/AAR Dialing without FAC? y DCS (Basic)? y ASAI Link Core Capabilities? n DCS Call Coverage? y ASAI Link Plus Capabilities? n DCS with Rerouting? y Async. Transfer Mode (ATM) PNC? n Async. Transfer Mode (ATM) Trunking? n Digital Loss Plan Modification? y ATM WAN Spare Processor? n DS1 MSP? y ATMS? y DS1 Echo Cancellation? y Attendant Vectoring? y

Navigate to Page 4 and verify the following:

• Check ISDN-BRI Trunks is set to y

• Check ISDN-PRI is set to y

change system-parameters customer-options Page 4 of 11 OPTIONAL FEATURES Emergency Access to Attendant? y IP Stations? y Enable 'dadmin' Login? y Enhanced Conferencing? y ISDN Feature Plus? n Enhanced EC500? y ISDN/SIP Network Call Redirection? y Enterprise Survivable Server? n ISDN-BRI Trunks? y Enterprise Wide Licensing? n ISDN-PRI? y ESS Administration? y Local Survivable Processor? n Extended Cvg/Fwd Admin? y Malicious Call Trace? y External Device Alarm Admin? y Media Encryption Over IP? n Five Port Networks Max Per MCC? n Mode Code for Centralized Voice Mail? n Flexible Billing? n

DS; Reviewed:

SPOC 03/25/2012



7 of 46

AC3K_MX52_T1

Navigate to Page 5 and verify the field Private Networking is set to y.

change system-parameters customer-options Page 5 of 11 OPTIONAL FEATURES Multinational Locations? n Station and Trunk MSP? y Multiple Level Precedence & Preemption? n Station as Virtual Extension? y Multiple Locations? n System Management Data Transfer? n Personal Station Access (PSA)? y Tenant Partitioning? y PNC Duplication? n Terminal Trans. Init. (TTI)? y Port Network Support? y Time of Day Routing? y Posted Messages? y TN2501 VAL Maximum Capacity? y Uniform Dialing Plan? y Private Networking? y Usage Allocation Enhancements? y Processor and System MSP? y Processor Ethernet? y Wideband Switching? y Wireless? n Remote Office? y Restrict Call Forward Off Net? y Secondary Data Module? y

Navigate to Page 8 and verify the following fields:

• Check Basic Call Setup is set to y

• Check Basic Supplementary Service is set to y

change system-parameters customer-options Page 8 of 11 QSIG OPTIONAL FEATURES Basic Call Setup? y Basic Supplementary Services? y Centralized Attendant? y Interworking with DCS? y Supplementary Services with Rerouting? y Transfer into QSIG Voice Mail? y Value-Added (VALU)? y

DS; Reviewed:

SPOC 03/25/2012



8 of 46

AC3K_MX52_T1

2.2. Define DS1 Circuit Pack

Using the command add ds1 <board> i.e. add ds1 01A08, configure the circuit with the

following commands:

• Location Board Identifier i.e. 01A08

• Name Name to identify the circuit i.e. T1ISDN to M3K

• Bit Rate Set to 1.544 for T1

• Line Coding Set to b8zs for T1

• Signaling Mode Set to isdn-pri

• Connect Set to pbx

• Interface Set to network (One side of the T1 link is defined with the

interface network and the opposite end will be defined as

User)

• Country Protocol Set to 1 (country protocol used by the local telephone company

central office at which this link terminates)

• Interface Companding Set to mulaw

• Idle Code Set to 11111111

add ds1 01a08 Page 1 of 2 DS1 CIRCUIT PACK Location: 01A08 Name: T1ISDN to M3K Bit Rate: 1.544 Line Coding: b8zs Line Compensation: 1 Framing Mode: esf Signaling Mode: isdn-pri Connect: pbx Interface: network TN-C7 Long Timers? n Country Protocol: 1 Interworking Message: PROGress Protocol Version: a Interface Companding: mulaw CRC? n Idle Code: 11111111 DCP/Analog Bearer Capability: 3.1kHz T303 Timer(sec): 4 Slip Detection? n Near-end CSU Type: other Echo Cancellation? n

DS; Reviewed:

SPOC 03/25/2012



9 of 46

AC3K_MX52_T1

2.3. Define Dial Plan attributes

Using the command change dialplan analysis add in the dial patterns to be used for

communicating with the Meeting Exchange via Mediant 3000. In the example below, the digit

pattern 67 has been added for Meeting Exchange conferencing services. Configure the fields as

follows:

• Dialed String Enter the number pattern to be dialed i.e. 67

• Total Length The maximum number of characters in the number being dialed i.e.5

• Call Type Set to aar for AAR Digit Analysis to compare the dialed number to

determine the route pattern (see Section 2.7)

change dialplan analysis Page 1 of 12 DIAL PLAN ANALYSIS TABLE Location: all Percent Full: 1 Dialed Total Call Dialed Total Call Dialed Total Call String Length Type String Length Type String Length Type 1 4 dac 35 5 aar 40 5 ext 45 5 ext 56 6 aar 67 5 aar 799 3 fac * 3 fac # 3 fac

DS; Reviewed:

SPOC 03/25/2012



10 of 46

AC3K_MX52_T1

2.4. Create a Trunk group for PSTN/T1

Using the command add trunk-group n where n is the number to be assigned to the trunk

group. There are three pages to be configured. Configure the following fields on Page 1:

• Group Number Group number created from the initial add command

• Group Type Set to isdn

• Group Name Suitable name to identify the trunk group

• COR Set Class of Restriction to 1

• TN Set Tenant Number to 1 (Default)

• TAC Set Trunk Access Code to 1150

• Direction Set to two-way

• Carrier Medium Set to PRI/BRI

• Service Type Set to tie

add trunk-group 150 Page 1 of 21 TRUNK GROUP Group Number: 150 Group Type: isdn CDR Reports: y Group Name: T1ISDN to M3K COR: 1 TN: 1 TAC: 1150 Direction: two-way Outgoing Display? n Carrier Medium: PRI/BRI Dial Access? n Busy Threshold: 255 Night Service: Queue Length: 0 Service Type: tie Auth Code? n TestCall ITC: rest Far End Test Line No: TestCall BCC: 4

Navigate to Page 2 and set the field Supplementary Service Protocol to a for National

Services.

add trunk-group 150 Page 2 of 21 Group Type: isdn TRUNK PARAMETERS Codeset to Send Display: 6 Codeset to Send National IEs: 6 Max Message Size to Send: 260 Charge Advice: none Supplementary Service Protocol: a Digit Handling (in/out): enbloc/enbloc Trunk Hunt: cyclical Digital Loss Group: 13 Incoming Calling Number - Delete: Insert: Format: Bit Rate: 1200 Synchronization: async Duplex: full Disconnect Supervision - In? y Out? n Answer Supervision Timeout: 0 Administer Timers? n CONNECT Reliable When Call Leaves ISDN? n XOIP Treatment: auto Delay Call Setup When Accessed Via IGAR? n

DS; Reviewed:

SPOC 03/25/2012



11 of 46

AC3K_MX52_T1

Navigate to Page 3 and configure the following fields:

• Send Name Set to y

• Send Calling Number Set to y

• Format Set to public

add trunk-group 150 Page 3 of 21 TRUNK FEATURES ACA Assignment? n Measured: none Wideband Support? n Internal Alert? n Maintenance Tests? y Data Restriction? n NCA-TSC Trunk Member: Send Name: y Send Calling Number: y Used for DCS? n Send EMU Visitor CPN? n Suppress # Outpulsing? n Format: public Outgoing Channel ID Encoding: preferred UUI IE Treatment: service-provider Replace Restricted Numbers? n Replace Unavailable Numbers? n Send Connected Number: n Network Call Redirection: none Hold/Unhold Notifications? n Send UUI IE? y Modify Tandem Calling Number: no Send UCID? n Send Codeset 6/7 LAI IE? y Ds1 Echo Cancellation? n Apply Local Ringback? n US NI Delayed Calling Name Update? n Show ANSWERED BY on Display? y Network (Japan) Needs Connect Before Disconnect? n

Trunks will be assigned to the trunk group at a later stage after further configuration has taken

place – see Section 2.6.

2.5. Define Signaling Group for PSTN/T1 trunk

Use the command add signaling-group n where n is the number to be used for the signaling

group – common practice is to define this number to be the same as trunk group number.

• Group Type Set to isdn-pri

• Primary D-Channel Set the D channel for the board i.e. 01A0824, where 01A08 is

the board and channel 24 is the indicated to be the D channel

• TSC Supplementary Service Protocol

Set to a. This should be set to the same value as

Supplementary Service Protocol value found in Page 2 of the

Trunk Group screen. (See Section 2.4)

add signaling-group 150 Page 1 of 5 SIGNALING GROUP Group Number: 150 Group Type: isdn-pri Associated Signaling? y Max number of NCA TSC: 0 Primary D-Channel: 01A0824 Max number of CA TSC: 0 Trunk Group for NCA TSC: 150 Trunk Group for Channel Selection: 150 X-Mobility/Wireless Type: NONE TSC Supplementary Service Protocol: a Network Call Transfer? n

DS; Reviewed:

SPOC 03/25/2012



12 of 46

AC3K_MX52_T1

2.6. Assigning the trunks to the Trunk Group

Using the command change trunk-group n, where n is the trunk group created in Section 2.4,

navigate to Page 5. Add in the required trunks and assign the signaling group.

• Port Enter each board and trunk number

• Sig Grp Enter the Signaling Group created in Section 2.5

change trunk-group 150 Page 5 of 21 TRUNK GROUP Administered Members (min/max): 1/23 GROUP MEMBER ASSIGNMENTS Total Administered Members: 23 Port Code Sfx Name Night Sig Grp 1: 01A0801 TN2464 C 150 2: 01A0802 TN2464 C 150 3: 01A0803 TN2464 C 150 4: 01A0804 TN2464 C 150 5: 01A0805 TN2464 C 150 6: 01A0806 TN2464 C 150 7: 01A0807 TN2464 C 150 8: 01A0808 TN2464 C 150 9: 01A0809 TN2464 C 150 10: 01A0810 TN2464 C 150 11: 01A0811 TN2464 C 150 12: 01A0812 TN2464 C 150 13: 01A0813 TN2464 C 150 14: 01A0814 TN2464 C 150 15: 01A0815 TN2464 C 150

Navigate to Page 6 to add the remaining trunks.

change trunk-group 150 Page 6 of 21 TRUNK GROUP Administered Members (min/max): 1/23 GROUP MEMBER ASSIGNMENTS Total Administered Members: 23 Port Code Sfx Name Night Sig Grp 16: 01A0816 TN2464 C 150 17: 01A0817 TN2464 C 150 18: 01A0818 TN2464 C 150 19: 01A0819 TN2464 C 150 20: 01A0820 TN2464 C 150 21: 01A0821 TN2464 C 150 22: 01A0822 TN2464 C 150 23: 01A0823 TN2464 C 150

DS; Reviewed:

SPOC 03/25/2012



13 of 46

AC3K_MX52_T1

2.7. Define Route Pattern

Each route pattern contains a list of trunk groups that can be used to route the call. The screen is

used to insert or delete digits so AAR or ARS calls route over different trunk groups, convert a

number to an international format or provide dial strings for alternative carriers in the event the

primary carrier is unavailable. Use the command change route-pattern n where n is the Trunk

Group number created in Section 2.4

• Grp No Set the desired trunk group number i.e. 150

• FRL Set to 0 for Facility Restriction Level (0-Lowest Restriction Level)

• Inserted Digits Prepend dialed number with international digits

change route-pattern 150 Page 1 of 3 Pattern Number: 150 Pattern Name: PSTN to M3K SCCAN? n Secure SIP? n Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC No Mrk Lmt List Del Digits QSIG Dgts Intw 1: 150 0 0 00353917 n user 2: n user 3: n user 4: n user 5: n user 6: n user BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR 0 1 2 M 4 W Request Dgts Format Subaddress 1: y y y y y n n rest none 2: y y y y y n n rest none 3: y y y y y n n rest none 4: y y y y y n n rest none 5: y y y y y n n rest none 6: y y y y y n n rest none

2.8. Define Public Numbering

Specify the desired digits for the Calling Number IE and the Connected Number IE for any

extension in the Public and/or Unknown Number Plans. Use the command change public-

unknown-numbering 0.

• Ext Len Set the number of digits

• Ext Code Set the extension code to be used

• Trk Grp(s) Set the trunk groups to be used by this extension range

• CPN Prefix Leave blank unless digits need to be inserted at the beginning of the

extension

• Total CPN Length Total length of the number pattern

change public-unknown-numbering 0 Page 1 of 2 NUMBERING - PUBLIC/UNKNOWN FORMAT Total Ext Ext Trk CPN CPN Len Code Grp(s) Prefix Len Total Administered: 5 5 4 150 5 Maximum Entries: 9999

DS; Reviewed:

SPOC 03/25/2012



14 of 46

AC3K_MX52_T1

2.9. Define AAR Analysis

Communications Manager compares dialed numbers with the dialed strings in this table and

determines the route pattern for the number. Use the command change aar analysis 0 to display

all dialed string entries. Set the following:

• Dialed String Set the dial string created in Section 2.3

• Total Min/Max Set the minimum and maximum characters to be expected in the

dialed string

• Route Pattern Set the Route Pattern created in Section 2.7

• Call Type Set the call type required

• ANI Reqd Set to n

change aar analysis 0 Page 1 of 2

AAR DIGIT ANALYSIS TABLE Location: all Percent Full: 1 Dialed Total Route Call Node ANI String Min Max Pattern Type Num Reqd 35 5 5 1 unku n 45 5 5 1 unku n 56 6 6 50 aar n 67 5 5 150 aar n

2.10. Check Class of Restriction Settings

From the COR that was assigned to the trunk group (see Section 2.4), check that calls are not

barred. Use the command change cor n, where n is the COR number used in the trunk group.

The default is usually 1.

• Calling Party Restriction Set to none

• Called Party Restriction Set to none

change cor 1 Page 1 of 23 CLASS OF RESTRICTION COR Number: 1 COR Description: FRL: 0 APLT? y Can Be Service Observed? n Calling Party Restriction: none Can Be A Service Observer? n Called Party Restriction: none Time of Day Chart: 1 Forced Entry of Account Codes? n Priority Queuing? n Direct Agent Calling? n Restriction Override: none Facility Access Trunk Test? n Restricted Call List? n Can Change Coverage? n Access to MCT? y Fully Restricted Service? n Group II Category For MFC: 7 Hear VDN of Origin Annc.? n Send ANI for MFE? n Add/Remove Agent Skills? n MF ANI Prefix: Automatic Charge Display? n Hear System Music on Hold? y PASTE (Display PBX Data on Phone)? n Can Be Picked Up By Directed Call Pickup? n Can Use Directed Call Pickup? n Group Controlled Restriction: inactive

DS; Reviewed:

SPOC 03/25/2012



15 of 46

AC3K_MX52_T1

2.11. Save Changes to Avaya Aura® Communication Manager

Commit the changes that have been made to the Communications Manager with the command

save translation.

save translation SAVE TRANSLATION Command Completion Status Error Code Success 0

DS; Reviewed:

SPOC 03/25/2012



16 of 46

AC3K_MX52_T1

3. Configure Avaya Meeting Exchange® Enterprise Edition This section describes the steps for configuring the Meeting Exchange to interoperate with

Mediant 3000 via SIP. It is assumed that the Meeting Exchange is installed and licensed as

described in the product documentation (see Section 8 reference [3] and [4]). The following

steps describe the administrative procedures for configuring Meeting Exchange:

• Configure SIP Connectivity

• Configure Dialout

• Map DNIS Entries

• Configure Audio Preferences

• Restarting the Meeting Exchange server

• Configure Bridge Talk

The following instructions require logging in to the Meeting Exchange console using an ssh

connection to access the Command Line Interface (CLI) with the appropriate credentials.

3.1. Configuring SIP Connectivity

Log in to the Meeting Exchange server console using an ssh Client to access the Command Line

Interface (CLI) with the appropriate credentials. Verify the the Meeting Exchange server is

configured to use TCP.

• In /usr/ipcb/config/system.cfg verify the following default values are configured. The IP

address will be the IP address of the Meeting Exchange server. MyListener=sip:[email protected] respContact=<sip:[email protected]:5060;transport=tcp>

# ip address of the server IPAddress=10.10.9.67 # request we will be listening to MyListener=sip:[email protected] # if this setting is populated will Overwrite the contact field in responses respContact=<sip:[email protected]:5060;transport=tcp> MaxChannelCount=3200 # diff serv values that will appear on the TOS field of the IP packet DiffServSignallingTOSValue=34 DiffServMediaTOSValue=46 # vlan values EthernetSignallingVlanValue=0 EthernetMediaVlanValue=0 # SIP settings sessionRefreshTimer=900 minSETimerValue=900

DS; Reviewed:

SPOC 03/25/2012



17 of 46

AC3K_MX52_T1

3.2. Configure Dialout

To enable Dial-Out from the Meeting Exchange to Mediant 3000, edit the telnumToUri.tab file

as follows:

• Edit /usr/ipcb/config/telnumToUri.tab file with a text editor

• Add the following line to the file to route outbound calls from the Meeting Exchange to

the Mediant 3000 ip address.

* sip:[email protected]:5060;transport=tcp Mediant3000

Note: Mediant3000 is a comment description for this line.

# telnum to uri conversion table # # This file is for dialing out from the Bridge to an external party. The # digits that are dialed are converted into the Request URI in the SIP INVITE. # For example, if the digits dialed were 936543 and one of the patterns was # "93????" a match would take place. If the conversion for that match was # $1 then the Request URI for the SIP INVITE would be sip:[email protected] # # THE COMMENT COLUMN OR ANY OF THE COLUMNS SHOULD HAVE NO SPACES TelnumPattern TelnumConversion comment * sip:[email protected]:5060;transport=tcp Mediant3000

3.3. Map DNIS Entries

The DNIS entry is the number dialed by Avaya subscribers to access a conference service

(SCAN, FLEX or DIRECT) on Meeting Exchange. The DNIS entry needs to be mapped on

Meeting Exchange to enable access to a conference. To map DNIS entries, run the cbutil utility

on Meeting Exchange. Log in to the Meeting Exchange with a ssh connection with the

appropriate credentials. Enable Dial-In access (via passcode) to conferences provisioned on the

Meeting Exchange as follows:

Add a DNIS entry for a scan call function corresponding to DID 67000 by entering the

following command at the command prompt:

cbutil add <dnis> <rg> <msg> <ps> <ucps> <func> [-o <of> -l <ln> -c <cn> -crs <n> -cre

<n> -cc <code>]

where the variables for add command is defined as follows:

• <dnis> DNIS

• <rg> Reservation Group

• <msg> Annunciator message number

• <ps> Prompt Set number (0-20)

• <ucps> Use Conference Prompt Set (y/n)

• <func> One of: DIRECT/SCAN/ENTER/HANGUP/AUTOVL/FLEX

• –o <of> Optional On-failure function – one of: ENTER/HANGUP or DEFAULT..

• –l <"ln"> Optional line name to associate with caller

• –c <"cn"> Optional company name to associate with caller

DS; Reviewed:

SPOC 03/25/2012



18 of 46

AC3K_MX52_T1

• –crs <n> Optional conference room start number

• –cre <n> Optional conference room end number

• –cc<n> Optional setting only used with DIRECT

In this sample configuration, the DNIS entry for a SCAN call function was added corresponding

to DNIS 67000 by entering the following command at the command prompt:

cbutil add 67000 0 247 0 N SCAN –l TestScan

[sroot@MXAPP2 ~]# cbutil add 67000 0 247 0 N SCAN –l TestScan cbutil Copyright 2004 Avaya, Inc. All rights reserved.

At the command prompt, enter cbutil list to verify the DNIS entries provisioned.

[sroot@MX52-0967 log]# cbutil list cbutil Copyright 2004 Avaya, Inc. All rights reserved. DNIS Grp Msg PS CP Function On Failure Line Name Company Name Room Start Room End Conference Code ---------------- --- ---- --- -- --------- --------- -------------------- ------------------- ----------- ---------- ---------------- 67000 0 247 0 N SCAN DEFAULT TestScan 0 0 67001 0 332 0 N FLEX DEFAULT TestFlex 0 0 67010 0 301 0 N DIRECT DEFAULT TestDirect 0 0

For further details of the parameter fields used, see Section 8 References [3] and [4].

DS; Reviewed:

SPOC 03/25/2012



19 of 46

AC3K_MX52_T1

3.4. Configure the Dial In

It may also be necessary to configure the Meeting Exchange to recognize the format of the URI

inbound from the Mediant 3000. This is configured in the UriToTelnum.tab file.

• Edit /usr/ipcb/config/UriToTelnum.tab file with a text editor

Add in the relevant format of the URI. This can be traced via either a Wireshark trace or by using

the Message Log screen available in Mediant 3000 web interface. (Select Status & Diagnostics

�� System Status �� Message Log). During the call trace, the log will display SIP information

to and from the Meeting Exchange. This information can be used to interpret the best

configuration for the UriToTelnum.tab file. For further details of the formats used within the file

may be found in Section 8 References [3] and [4].

TelnumPattern TelnumConversion comment "*;dnis=*;*" $2 ddi_in_dnis_parameter_followed_by_additional_parameters "*;dnis=*" $2 ddi_in_dnis_parameter_not_followed_by_additional_parameters "*AdhocDirect*@*" $2 diamond_adhoc_conf_support "sip:*@*" $1 ddiRequest_Uri "<sip:*@*" $1 ddi_Request_Uri "<sip:*" "0000" no_Request_Uri "sip:*" "0000" no_Request_Uri "*sip:*@*" $2 AvayaPhoneFormat "sips:*@*" $1 ddiRequest_Uri_secure_sip "<sips:*@*" $1 ddi_Request_Uri_secure_sip "<sips:*" "0000" no_Request_Uri_secure_sip "sips:*" "0000" no_Request_Uri_secure_sip "*sips:*@*" $2 AvayaPhoneFormat_secure_sip * $0 wildcard

DS; Reviewed:

SPOC 03/25/2012



20 of 46

AC3K_MX52_T1

3.5. Configure Audio Preferences

The audioPreferences.cfg file located at /usr/ipcb/config/ specifies the order in which codecs

are offered in the Session Description Protocol. Set the audio codec (i.e. PCMU, PCMA etc.,) as

required and the telephone-event value to payloadType of 101.

# audioPreferences.cfg # This table is an ordered list of MIME subtypes specifying the codecs supported # by this media server. The list is specified in the order in which an SDP offer # will list the various MIME subtypes on the m=audio line. # For static payload type numbers (i.e. numbers between 0 - 96) please use the # iana registered numbering scheme. # See: http://www.iana.org/assignments/rtp-parameters mimeSubtype payloadType Performance PCMU 0 10 PCMA 8 10 #G722 9 85 G729 18 118 #iLBC30 97 140 #iLBC20 98 140 #wbPCMU 102 10 #wbPCMA 103 10 telephone-event 101 10 #iSAC 104 150 #G726_16 105 54 #G726_24 106 54 G726_32 107 54 #G726_40 108 54

3.6. Restarting the Avaya Meeting Exchange® Enterprise Server

After the configuration changes are made, restart the services issuing the command bridge

restart.

[sroot@MXAPP2 config]# bridge restart /etc/init.d/mx-bridge: Restarting bridge /etc/init.d/mx-bridge: Server type is DCB /etc/init.d/mx-bridge: Stopping DCB conferencing server bridge via uninitdcb.sh notificationCtrlServer not installed Sending CMD_SHUTDOWN level 3 message to the INIT_KEY queue. Waiting for 7 processes to stop Waiting for 2 processes to stop …. System shutdown processes….

…. =========================================== INITDCB ============================== FirstMusic = 3199. FirstLink = 3198. FirstRP = 3197. FirstOper = 3196. numUserLCNs = 3196. [sroot@MXAPP2 config]#

DS; Reviewed:

SPOC 03/25/2012



21 of 46

AC3K_MX52_T1

To confirm the services are back, press enter to get a # prompt on screen. At the prompt type

dcbps and press return. The display should be similar to the one below, with no processes

stopped.

[sroot@MXAPP2 config]# dcbps 12595 pts/0 00:00:00 initdcb 12630 pts/0 00:00:00 log 12633 pts/0 00:00:00 bridgeTranslato 12634 pts/0 00:00:00 netservices 12664 pts/0 00:00:00 timer 12665 pts/0 00:00:00 traffic 12666 pts/0 00:00:00 chdbased 12667 pts/0 00:00:00 startd 12668 pts/0 00:00:00 cdr 12669 pts/0 00:00:00 modapid 12670 pts/0 00:00:00 schapid 12671 pts/0 00:00:00 callhand 12672 pts/0 00:00:00 initipcb 12677 pts/0 00:00:00 sipagent 12678 pts/0 00:00:01 msdispatcher 12679 pts/0 00:00:00 serverComms 12680 pts/0 00:00:00 softms 12681 pts/0 00:00:00 softms 12682 pts/0 00:00:00 softms 12683 pts/0 00:00:00 softms 12685 pts/0 00:00:00 softms 12687 pts/0 00:00:00 softms 11033 ? 00:00:01 postmaster with 26 children

DS; Reviewed:

SPOC 03/25/2012



22 of 46

AC3K_MX52_T1

4. Configuring Avaya Bridge Talk The following steps utilize the Avaya Bridge Talk application to provision a sample conference

on the Meeting Exchange. This sample conference enables both Dial-In and Dial-Out access to

audio conferencing for endpoints on the Public Switched Telephone Network.

Notes: If any of the features displayed in the Avaya Bridge Talk screen captures are not present,

contact an authorized Avaya Sales representative to make the appropriate changes.

4.1.1. Initializing Avaya Bridge Talk

Invoke the Avaya Bridge Talk application as follows:

• Double-click on the desktop icon from a Personal Computer loaded with the Avaya

Bridge Talk application and with network connectivity to the Meeting Exchange (not

shown).

• Enter the appropriate credentials in the Sign-In and Password fields.

• Enter the IP address of the Meeting Exchange server (10.10.9.67 for this sample

configuration) in the Bridge field as shown below.

DS; Reviewed:

SPOC 03/25/2012



23 of 46

AC3K_MX52_T1

4.1.2. Creating a Dial Out list

Provision a dial list that is utilized for Dial-Out (e.g., Blast dial and Fast dial) from the Meeting

Exchange. From the Avaya Bridge Talk Menu Bar, click Fast Dial � New.

DS; Reviewed:

SPOC 03/25/2012



24 of 46

AC3K_MX52_T1

4.1.3. Creating a Dial List

From the Dial List Editor window that is displayed below:

• Enter a descriptive label in the Name field.

• Enable conference participants on the dial list to enter the conference without a passcode

by selecting the Directly to Conf box as displayed.

• Add entries to the dial list by clicking on the Add button and enter Name, Company and

Telephone number for dial out for each participant. [Optional] Moderator privileges may

be granted to a conference participant by checking the Moderator box.

When finished, click on the Save button on the bottom of the screen.

DS; Reviewed:

SPOC 03/25/2012



25 of 46

AC3K_MX52_T1

4.1.4. Conference Scheduler

From the Avaya Bridge Talk menu bar, click View � Conference Scheduler to provision a

conference.

DS; Reviewed:

SPOC 03/25/2012



26 of 46

AC3K_MX52_T1

4.1.5. Provision a Conference

From the Schedule Conference window that is displayed, provision a conference as follows:

• Choose Conference Type as DAILY

• Choose Weekend as YES for the conference to be available at Weekends

• Enter a unique Conferee Code to allow participants access to this conference

• Enter a unique Moderator Code to allow participants access to this conference with

moderator privileges

• Enter a descriptive label in the Conference Name field

• Administer settings to enable an Auto Blast dial by setting Auto/Manual as desired

• Enter a Start Date and End Date. Leave End Date blank for open ended conference

• Set Start Time and End Time to determine when the conference can be activated

• Set the Maximum Lines to be used in the conference

Select a dial list by clicking on the Dial List button. Either select a preconfigured list, by

clicking on the item and clicking select, Edit an existing list, or Create a new list. (Screens not

shown). When finished, click on the Save button on the bottom of the screen.

DS; Reviewed:

SPOC 03/25/2012



27 of 46

AC3K_MX52_T1

5. Configure the AudioCodes Mediant 3000 Media Gateway This chapter describers the configuration of the Mediant 3000 to provide PSTN connectivity

with both the Meeting Exchange and the Public Switched Telephone Network. Configuration is

performed using the Embedded Web Server which supports gateway configuration, including

loading of configuration files. The Embedded Web Server can be accessed from a standard web

browser. Specifically, users can employ this facility to set up the Gateway Configuration

parameters. Users may also remotely reset the gateway and permanently apply the new

parameters. This document assumes the Mediant 3000 has been initially configured with an IP

address and introduced to the Network.

5.1. Accessing AudioCodes Mediant 3000 via Web Interface

From an Internet Browser, navigate to http://<ip-addr> where <ip-addr> is the IP address

assigned to the Mediant 3000. A login window will appear. Log in with the required credentials.

DS; Reviewed:

SPOC 03/25/2012



28 of 46

AC3K_MX52_T1

Once connected successfully, the Home Screen will be displayed. Certain areas of the screen are

useful to the end user:

• Configuration Access to the menu system for configuration tasks. Choose

between Basic and Full menu displays

• Status & Diagnostics Diagnostic screens for testing / fault tracing

• Submit After making changes use the Submit button to submit the

changes

• Burn Burn the changes made to the flash of the device

• Device Actions The drop down menu provides access to the reset menu

allowing the user to restart the device

• General Information Shows the IP information and firmware version

DS; Reviewed:

SPOC 03/25/2012



29 of 46

AC3K_MX52_T1

5.2. Configuring TDM Bus Settings

On the left click Configuration and navigate to VoIP �� TDM & Timing �� Digital PCM

Settings. Configure the following:

• PCM Law Select Set to MuLaw or ALaw as required

Submit the changes by clicking on the Submit button on the bottom right. This step should be

performed for each configuration change below, prior to navigating to a new screen.

Navigate to VoIP �� TDM & Timing �� System Timing. Configure the following:

• TDM Bus Clock Source Set to Network (for clock source from the

PSTN line or Internal as required)

• TDM Bus Local Reference Set to 1

• TDM Bus PSTN Auto FallBack Clock Set to Disable

• TDM Bus PSTN Auto Clock Reverting Set to Disable

Changes made may require the device to be reset (restarted).

DS; Reviewed:

SPOC 03/25/2012



30 of 46

AC3K_MX52_T1

5.3. Configuring Public Switched Telephone Network Trunk Settings

On the left select Configuration and navigate to VoIP �� PSTN �� Trunk Settings. Click on

the PSTN/T1 trunk to be configured. In this Application Notes, the second port of the Mediant

3000 Gateway is connected to Communication Manager.

Note: Click the Stop Trunk button first in order to modify the selected trunks. (Button is located

in bottom right hand corner.)

In General Settings section of Trunk Settings configure as follows:

• Protocol Type Set to appropriate Protocol Type i.e. T1 5ESS 10 ISDN

DS; Reviewed:

SPOC 03/25/2012



31 of 46

AC3K_MX52_T1

In the Trunk Configuration section of Trunk Settings configure as follows:

• Clock Master Set to Recovered as the clock is recovered from the PSTN

• Line Code Se to B8ZS which is used for T1 trunks

• Framing Method Set to T1 FRAMING ESF CRC6

In the ISDN Configuration section of Trunk Settings configure as follows:

• ISDN Termination Side Set to User side

Confirm the changes by clicking Apply Trunk Settings button.

Submit the changes by clicking on the Submit button on the left.

DS; Reviewed:

SPOC 03/25/2012



32 of 46

AC3K_MX52_T1

Once the submit button is clicked, the trunk indicator should show green.

5.4. Administer PSTN Trunk Group

To configure the trunk group associated with the PSTN/T1 port configured in Section 5.3, click

Configuration and navigate to VoIP �� GW and IP to IP �� Trunk Group �� Trunk Group.

Set the following parameters for Group Index 2, leaving the remaining parameters at default.

• From Trunk Set to 2 (Starting physical trunk in the Trunk Group)

• To Trunk Set to 2 (Ending physical trunk in the Trunk Group)

• Channels Enter 1-23 (the device B-Channels)

• Phone Number Enter a logical phone number that will be used if a call from the

PSTN does not contain a calling number. (Optional)

• Trunk Group ID Set to 2

DS; Reviewed:

SPOC 03/25/2012



33 of 46

AC3K_MX52_T1

5.5. Administer Routing Table

To administer call routing for calls originating from the PSTN to the Meeting Exchange using

TCP, on the left click on Configuration and navigate to VoIP �� GW and IP to IP Routing ��

Tel to IP Routing. Configure the following to indicate calls are to be routed to Meeting

Exchange.

• Src. Trunk Group ID Enter the Trunk Group ID administered in Section 5.3 or

set to * to indicate a wildcard

• Dest. Phone Prefix Enter a rule to match the pattern of the incoming call to

Mediant 3000 from the PSTN. Alternatively use * to route

any incoming call from the PSTN to Meeting Exchange.

• Source Phone Prefix Enter a rule or * to allow routing for any source number

dialing in Meeting Exchange from the PSTN.

• Dest. IP Address Enter the IP address of Meeting Exchange

• Port Set to 5060

• Transport Type Set to TCP

• IP Profile ID Set to 1 (See Section 5.8 for IP Profile Configuration)

DS; Reviewed:

SPOC 03/25/2012



34 of 46

AC3K_MX52_T1

To administer call routing for calls originating from Meeting Exchange using TCP to PSTN, on

the left click on Configuration and navigate to VoIP �� GW and IP to IP Routing �� IP to

Trunk Group Routing. Configure the following to indicate calls are to be routed to PSTN from

Meeting Exchange.

• Dest. Host Prefix Enter the Mediant 3000 IP address

• Source Host Prefix Enter the Meeting Exchange IP address

• Dest. Phone Prefix Enter a rule or * to allow routing of any destination phone

prefix to PSTN from Meeting Exchange

• Source Phone Prefix Enter a rule or * to allow routing of any source phone prefix to

PSTN from Meeting Exchange

• Source IP Address Enter a rule or * to allow routing of any source IP address to

PSTN from Meeting Exchange

• Trunk Group ID Enter trunk group ID selected in Section 5.3

DS; Reviewed:

SPOC 03/25/2012



35 of 46

AC3K_MX52_T1

5.6. Administer SIP Protocols

Although the calls to the Mediant 3000 are received via PSTN, the onward calls to the Meeting

Exchange are via SIP, so some SIP related parameters are required to be set in Mediant 3000.

5.6.1. Configuring SIP General Parameters for TCP

If using TCP as the transport method for SIP calls, then the system requires configuration. From

the pane on the left select Configuration and navigate to VoIP �� SIP Definitions �� General

Parameters. Set the following and leave the remaining values as default.

• SIP Transport Type Set to TCP

• SIP TCP Port Set to 5060

• Enable SIPS Set to Disable

• SIP Destination Port Set to 5060

DS; Reviewed:

SPOC 03/25/2012



36 of 46

AC3K_MX52_T1

5.6.2. Configure Proxy and Registration

These Application Notes use the call routing table of the Mediant 3000 for outbound calls, rather

than using a default SIP Proxy (i.e. Session Manager). Manually configured routing is useful

when the dialled number rarely changes or when a single number is dialled by a large number of

users. On the left click Configuration and navigate to VoIP �� SIP Definitions �� Proxy &

Registration. Configure the following:

• Use Default Proxy Set to No as the SIP connectivity between the Mediant 3000

and Meeting Exchange is a direct connection

• Prefer Routing Table Set to Yes. The device checks the routing rules in the

“Outbound IP Routing Table” for a match with the Tel-to-

IP call

• Always Use Proxy Set to Disable to use standard SIP routing rules

• Redundant Routing Mode Set to Routing Table. The internal routing table is used to

locate a redundant route

DS; Reviewed:

SPOC 03/25/2012



37 of 46

AC3K_MX52_T1

5.7. Define Audio Codecs Table

The following steps describe the procedure to configure the Mediant 3000 for supporting

preferred codes. From the pane on the left select Configuration and navigate to VoIP �� Coders

and Profiles �� Coders. From the field Coder Name click the drop down and select the

required codec. It will automatically fill in the remaining fields with default values, although

these can still be altered if required.

Note: The first coder is the highest priority coder and is used by the Mediant 3000 whenever

possible. If the far end SIP User Agent cannot use the coder assigned as the first coder, the

gateway attempts to use the next coder and so on.

5.7.1. Coder Group Settings

The Coder Group is used as part if the IP Profile. To configure this click on Configuration and

navigate to VoIP �� Coders and Profiles �� Coders Group Settings. Configure as follows:

• Coder Group ID Select Group ID to 1

• Coder Name Set the codecs as required

DS; Reviewed:

SPOC 03/25/2012



38 of 46

AC3K_MX52_T1

5.8. Configure IP Profile Settings

The IP Profile is used as part of the configuration of the Routing Table (see Section 5.5). To

configure the IP Profile Settings on the left click on Configuration and navigate to VoIP ��

Coders and Profiles �� IP Profile Settings. Set the Profile ID to 1.

Scroll down to the Gateway Parameters section. Set the following parameters:

• Coder Group Set to Coder Group 1

• First Tx DTMF Option Set to RFC 2833

• Declare RFC2833 in SDP Set to Yes

Navigate to VoIP �� Coders and Profiles �� Tel Profile Settings. Set the Coder Group to

Coder Group 1.

DS; Reviewed:

SPOC 03/25/2012



39 of 46

AC3K_MX52_T1

5.9. Configure DTMF Settings

The DTMF & Dialing page is used to configure parameters associated with dual-tone multi-

frequency (DTMF) and dialing. Meeting Exchange can transmit either as in-band or

alternatively as out-of-band or RFC2833.

5.9.1. Configure In-Band Settings

To configure in-band settings click on Configuration on the left and navigate to VoIP �� GW

and IP to IP �� DTMF and Supplementary �� DTMF & Dialing. Configure the following

field:

• Declare RFC 2833 in SDP Set to No

DS; Reviewed:

SPOC 03/25/2012



40 of 46

AC3K_MX52_T1

5.9.2. Configure Out of Band Settings (RFC2833)

To configure out of band settings click on Configuration on the left and navigate to VoIP��GW

and IP to IP��DTMF and Supplementary��DTMF & Dialing. Configure the following

fields:

• Declare RFC 2833 in SDP Set to Yes

• 1st Tx DTMF Option Set to RFC2833

• RFC 2833 Payload Type Set as required, 96 was used below

DS; Reviewed:

SPOC 03/25/2012



41 of 46

AC3K_MX52_T1

5.10. Define SIP Advanced Parameters

Click on Configuration on the left and navigate to VoIP �� SIP Definitions �� Advanced

Parameters. Change the parameter Disconnect on Broken Connection to No. Leave all other

fields as default.

Submit the changes by clicking on the Submit button on the bottom right.

Save the entire configuration to flash memory by clicking the Burn icon at the top.

DS; Reviewed:

SPOC 03/25/2012



42 of 46

AC3K_MX52_T1

The following prompt should be displayed, click OK.

Once the configuration has been successfully saved to flash the following message will be

displayed.

DS; Reviewed:

SPOC 03/25/2012



43 of 46

AC3K_MX52_T1

6. Verification This section provides the verification tests that can be performed on Mediant 3000 Media

Gateway and Meeting Exchange to verify their proper configuration.

6.1. Verify Avaya Meeting Exchange® Enterprise Edition processes are running

Verify all conferencing related processes are running on the Meeting Exchange as follows:

• Log in to the Meeting Exchange server console to access the CLI with the appropriate

credentials.

• cd to /usr/dcb/bin

• At the command prompt, run the script service mx-bridge status and confirm all

processes are running by verifying an associated Process ID (PID) for each process.

[sroot@MXAPP2 ~]# service mx-bridge status 7404 ? 00:00:00 initdcb 7439 ? 00:00:00 log 7442 ? 00:00:00 bridgeTranslato 7443 ? 00:00:00 netservices 7473 ? 00:00:00 timer 7474 ? 00:00:00 traffic 7475 ? 00:00:00 chdbased 7476 ? 00:00:00 startd 7477 ? 00:00:00 cdr 7478 ? 00:00:00 modapid 7479 ? 00:00:00 schapid 7480 ? 00:00:00 callhand 7481 ? 00:00:00 initipcb 7492 ? 00:00:00 sipagent 7499 ? 00:00:01 msdispatcher 7501 ? 00:00:00 serverComms 7502 ? 00:00:00 softms 7503 ? 00:00:00 softms 7504 ? 00:00:00 softms 7506 ? 00:00:00 softms 7507 ? 00:00:00 softms 7508 ? 00:00:00 softms 11434 ? 00:00:00 schapid 3663 ? 00:00:00 postmaster with 27 children

DS; Reviewed:

SPOC 03/25/2012



44 of 46

AC3K_MX52_T1

6.1.1. Verify Call Routing

Verify end to end signaling/media connectivity between the Meeting Exchange and Mediant

3000. This is accomplished by placing calls from Avaya end points to Meeting Exchange. This

step utilizes the Communications Manager to pass calls to the Mediant 3000 via PSTN, which in

turn passes calls to the Meeting Exchange via SIP. The Avaya Bridge Talk application is used to

verify calls to the three different conference types (SCAN, FLEX and DIRECT) are managed

correctly, e.g., callers are added/removed from conferences. This step will also verify the

conferencing applications provisioned.

• Configure a conference with Auto Blast auto-enabled and provision a dial list. From an

Avaya endpoint, dial a number that corresponds to a conference (i.e. 67000 for SCAN

conference). Login as a Moderator (with the relevant passcode) and blast dial is invoked

automatically. When answered these callers enter the conference.

• If not already logged on, log in to the Avaya Bridge Talk application with the appropriate

credentials.

• Double-Click on the highlighted Conf # to open a Conference Room window

• Verify conference participants are added/removed from conferences by observing the

Conference Navigator and/or Conference Room windows.

DS; Reviewed:

SPOC 03/25/2012



45 of 46

AC3K_MX52_T1

7. Conclusion As illustrated in these Application Notes, AudioCodes Mediant 3000 Media Gateway can

interoperate with Avaya Meeting Exchange® Enterprise Edition using SIP and PSTN T1 trunks.

Please refer to Section 1.2 above for details on observations found.

8. Additional References Product documentation for Avaya products may be found at http://support.avaya.com

[1] Administering Avaya Aura® Communication Manager 03-300509 Release 6.0 Issue 6.0

[2] Administering Avaya Aura® Communication Manager Server Options 03-603479

Release 6.0.1, Issue 2.2

[3] Using Meeting Exchange™ Release 5.2.2 04-603698 Issue 1

[4] Administering Meeting Exchange™ Servers Release 5.2.2 04-603708 Issue 1

[5] Avaya BridgeTalk v5.2.2.0.9 integrated Help.

Product documentation for AudioCodes products may be found at the following websites and

within service manuals.

[6] AudioCodes website http://www.audiocodes.com/products-lobby

DS; Reviewed:

SPOC 03/25/2012



46 of 46

AC3K_MX52_T1


Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and

™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks

are the property of their respective owners. The information provided in these Application

Notes is subject to change without notice. The configurations, technical data, and

recommendations provided in these Application Notes are believed to be accurate and

dependable, but are presented without express or implied warranty. Users are responsible for

their application of any products specified in these Application Notes.

Please e-mail any questions or comments pertaining to these Application Notes along with the

full title name and filename, located in the lower right corner, directly to the Avaya Solution &

Interoperability Test Lab at [email protected]

configuring the audiocodes mediant ... - devconnect program

Documents