mx50 Acm50 Tls

Avaya Solution & Interoperability Test Lab
Configuring Secure SIP Connectivity Utilizing Transport Layer Security (TLS) Between Avaya Communication Manager and the Avaya Meeting Exchange S6200 Conferencing Server - Issue 1.0
Abstract
These Application Notes describe the procedures for configuring secure SIP connectivity utilizing Transport Layer Security (TLS) between Avaya Communication Manager and the Avaya Meeting Exchange S6200 Conferencing Server (Meeting Exchange). Employing this configuration enables call origination/termination between Avaya Communication Manager and Avaya Meeting Exchange, where the signaling is secure SIP and the media is Real-time Transport Protocol (RTP). These Application Notes are an updated version of the Application Notes titled: Configuring secure SIP connectivity utilizing Transport Layer Security (TLS) between Avaya Communication Manager and Avaya Meeting Exchange (S6200) - Issue 1.0 [6].
REB; Reviewed: SPOC 6/2/2008
Solution & Interoperability Test Lab Application Notes 2008 Avaya Inc. All Rights Reserved.
1 of 30 MX50-ACM50-TLS
1. Introduction
These Application Notes describe the procedures for configuring secure SIP connectivity utilizing Transport Layer Security (TLS) between Avaya Communication Manager and the Avaya Meeting Exchange S6200 Conferencing Server (Meeting Exchange). Employing this configuration enables call origination/termination between Avaya Communication Manager and Avaya Meeting Exchange, where the signaling is secure SIP and the media is Real-time Transport Protocol (RTP). Figure 1 illustrates the sample configuration utilized for these Application Notes. Avaya Communication Manager is comprised of a pair of Avaya S8710 Servers and an Avaya G650 Media Gateway. Avaya Communication Manager provides enterprise telephony features and media gateway functionality for SIP, H.323, Digital and Analog telephones present in this sample configuration. Avaya Communication Manager is provisioned for call origination via secure SIP signaling to Avaya Meeting Exchange. Avaya Meeting Exchange is a SIP based voice conferencing solution that provides mid-market enterprise customers with an audio conferencing system that can reside an IP network. For this sample configuration, Avaya Meeting Exchange is provisioned to accept calls from Avaya Communication Manager through call branding that supported both direct and scan call flows. A direct call flow allows access to conferences provisioned on Avaya Meeting Exchange without entering a passcode. Conversely, to enter a conference via a basic call flow requires a passcode. Avaya Meeting Exchange was also administered for call origination via secure SIP signaling to telephones registered to either Avaya Communication Manager or Avaya SIP Enablement Services.
The signaling between Avaya Communication Manager and Avaya Meeting Exchange is depicted by the blue dashed line in Figure 1. To account for the SIP telephones in this sample configuration, Avaya SIP Enablement Services is utilized as a SIP registration server only.
Figure 1: Sample Configuration
2. Equipment and Software Validated

The following equipment and software versions are used for this sample configuration: Equipment Avaya S8710 Servers Avaya G650 Media Gateway Avaya TN2312BP (IPSI) Avaya TN799DP (C-LAN) Avaya TN2302AP (MEDPRO) Avaya Meeting Exchange S6200 Conferencing Server Avaya Bridge Talk Avaya SIP Enablement Services Avaya C364T-PWR Converged Stackable Switch Avaya 4600 Series IP Telephones Avaya 4600 Series IP Telephones Avaya 9600 Series IP Telephones Avaya 6408D+ Digital Telephones Analog Telephones Software Version Avaya Communication Manager 5.0 (R015x.00.0.825.4) HW12 FW042 HW01 FW026 HW20 FW117 MX 5.0 SP1 (mx5.0.1.0.18) 5.0 Build 11 SES 5.0 (5.0-00.0.825.31) 4.5.14 2.8 (H.323) 2.2.2 (SIP) 1.5 (H.323) ---
Table 1: Equipment and Software Versions
3. Avaya Communication Manager Configuration

This section describes the configuration for enabling Avaya Communication Manager to interoperate with Avaya Meeting Exchange. Avaya Communication Manager was administered from the System Access Terminal (SAT). In these Application Notes the SAT screens are shown with a gray shaded background. In some instances, the information from the original screen has been edited or annotated for brevity or clarity in presentation. For example, entries and/or fields in the SAT screens that were either modified or were required for these Application Notes are displayed with boldface type. Refer to [1] and [2] for additional information regarding the administration of Avaya Communication Manager.
3.1. Verify Licensing

The following steps verify licensing on Avaya Communication Manager that is required to support the configuration described in these Application Notes. If a required feature is not enabled or there is insufficient capacity, contact an authorized Avaya account representative to make the appropriate changes. Step Description 3.1.1 Issue the command display system-parameters customer-options and proceed to page 2. Verify that the licensing for the Maximum Administered SIP Trunks field is sufficient. Note: Each call between two SIP endpoints requires two SIP trunks for the duration of the call. For this sample configuration, Avaya Meeting Exchange is a SIP endpoint. Thus, a call between a SIP station registered to Avaya SIP Enablement Services and Avaya Meeting Exchange will use two SIP trunks. A call between a non-SIP station registered to Avaya Communication Manager and Avaya Meeting Exchange will use only one SIP trunk.
display system-parameters customer-options OPTIONAL FEATURES IP PORT CAPACITIES Maximum Administered H.323 Trunks: Maximum Concurrently Registered IP Stations: Maximum Administered Remote Office Trunks: Maximum Concurrently Registered Remote Office Stations: Maximum Concurrently Registered IP eCons: Max Concur Registered Unauthenticated H.323 Stations: Maximum Video Capable Stations: Maximum Video Capable IP Softphones: Maximum Administered SIP Trunks: Maximum Administered Ad-hoc Video Conferencing Ports: Maximum Number of DS1 Boards with Echo Cancellation: Maximum TN2501 VAL Boards: Maximum Media Gateway VAL Sources: Maximum TN2602 Boards with 80 VoIP Channels: Maximum TN2602 Boards with 320 VoIP Channels: Maximum Number of Expanded Meet-me Conference Ports: 800 12000 0 0 0 100 100 100 800 100 0 10 0 128 128 0 USED 30 12 0 0 0 0 16 7 175 3 0 0 0 0 0 0 Page 2 of 10
(NOTE: You must logoff & login to effect the permission changes.)
3.2. Configure Connectivity

This section describes the steps for configuring SIP connectivity utilizing TLS between Avaya Communication Manager and Avaya Meeting Exchange. Step Description 3.2.1 Issue the command change ip-codec-set <n>, where n is the number of a codec set. Add entries for audio codecs that are supported on Avaya Meeting Exchange. For this sample configuration, a single entry for G.711MU was added as displayed.
change ip-codec-set 1 IP Codec Set Codec Set: 1 Audio Codec 1: G.711MU 2: 3: 4: 5: 6: 7: Silence Suppression n Frames Per Pkt 2 Packet Size(ms) 20 Page 1 of 2
Step Description 3.2.2 Issue the command change ip-network-region <n>, where n is the number of an IP network region and administer settings as displayed. Enter the number of the IP codec set provisioned in Step 3.2.1 in the Codec Set field. Verify that the Inter-region IP-IP Direct Audio field is set to yes. This will allow direct IP-to-IP audio connectivity between IP stations registered to either Avaya Communication Manager or Avaya SIP Enablement Services and Avaya Meeting Exchange. By default, the C-LAN and all Avaya IP endpoints are assigned to IP network region 1. For this sample configuration, Avaya Meeting Exchange is associated with this IP network region. Use default settings for remaining fields. Note: It is not required that a second IP network region be created. For this sample configuration, a second IP network region is created to allow for the following: A distinct setting for the Authoritative Domain field. For this sample configuration, the Authoritative Domain field is not configured. An IP codec set to account for the requirements on Avaya Meeting Exchange (see Step 3.2.1). The option of restricting the bandwidth over the Wide Area Network (WAN) interface between IP network regions for calls (see Step 3.2.3).
change ip-network-region 2 Page 1 of 19
IP NETWORK REGION Region: 2 Location: Authoritative Domain: Name: Meeting Exchange MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 1 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? n UDP Port Max: 3329 DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y Call Control PHB Value: 46 RTCP MONITOR SERVER PARAMETERS Audio PHB Value: 46 Use Default Server Parameters? y Video PHB Value: 26 802.1P/Q PARAMETERS Call Control 802.1p Priority: 6 Audio 802.1p Priority: 6 Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS H.323 IP ENDPOINTS RSVP Enabled? n H.323 Link Bounce Recovery? y Idle Traffic Interval (sec): 20 Keep-Alive Interval (sec): 5 Keep-Alive Count: 5
Step Description 3.2.3 Proceed to Page 3 and enable inter-region connectivity between IP network regions 2 and 1 by entering the number of the IP codec set provisioned in Step 3.2.1 in the codec set field.
change ip-network-region 2 Inter Network Region Connection Management src rgn 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 dst codec direct WAN-BW-limits Video rgn set WAN Units Total Norm Prio Shr Intervening-regions 1 1 y NoLimit 2 1 3 4 5 6 7 8 9 10 11 12 13 14 15 Dyn CAC IGAR n Page 3 of 19
3.2.4 Issue the command change node-names ip and add an entry to map the IP address corresponding to Avaya Meeting Exchange to descriptive name. Verify that an entry exists for the Control LAN (CLAN) interface on the Avaya G650 Media Gateway.
change node-names ip IP NODE NAMES IP Address 192.168.11.10 192.168.11.11 192.168.11.20 192.168.13.102 Page 1 of 2
Name CLAN-1A02 MEDPRO-1A03 SES s6200
Step Description 3.2.5 Issue the command add signaling-group <n>, where n is the number of an unallocated signaling group and administer settings as displayed. To enable secure SIP connectivity utilizing TLS, set the Group Type field to sip. The Transport Method field will default to tls. Enter the node name for the CLAN (see Step 3.2.4) in the Near-end Node Name field. Enter the node name of Avaya Meeting Exchange provisioned in Step 3.2.4 in the Farend Node Name field. Enter the number of the IP network region provisioned in Step 3.2.2 in the Far-end Network Region field. Verify that the DTMF over IP field is set to rtp-payload. This value enables Avaya Communication Manager to send DTMF using RFC 2833. Verify that the Direct IP-IP Audio Connections field to y to enable direct IP-to-IP audio connectivity for IP stations utilizing this signaling group. Use default settings for remaining fields. Note: To enable direct IP-to-IP audio connectivity, the following must be administered: [Not Shown] Direct IP-to-IP audio connectivity must be enabled at the system-level on the system-parameters features form. Direct IP-to-IP audio connectivity must be enabled for the IP network region associated with this signaling group (see Step 3.2.2).
add signaling-group 3 SIGNALING GROUP Group Number: 3 IP Video? n Group Type: sip Transport Method: tls Page 1 of 1
Near-end Node Name: CLAN-1A02 Near-end Listen Port: 5061 Far-end Domain:
Far-end Node Name: s6200 Far-end Listen Port: 5061 Far-end Network Region: 2 Bypass If IP Threshold Exceeded? n
DTMF over IP: rtp-payload Enable Layer 3 Test? n Session Establishment Timer(min): 3
Direct IP-IP Audio Connections? y IP Audio Hairpinning? n
Step Description 3.2.6 Issue the command add trunk-group <n>, where n is the number of an unallocated trunk group and administer settings as displayed. Set the Group Type field to sip. Enter a descriptive name for the trunk group in the Group Name field. Enter a number in the TAC (Trunk Access Code) field that is consistent with the configuration for the dial plan. Enter the number of the signaling group provisioned in Step 3.2.5 in the Signaling Group field. Configure additional fields with boldface type as displayed and use default settings for remaining fields.
add trunk-group 3 TRUNK GROUP Group Number: Group Name: Direction: Dial Access? Queue Length: Service Type: 3 s6200 SIP two-way n 0 tie Group Type: sip CDR Reports: y COR: 1 TN: 1 TAC: 103 Outgoing Display? n Night Service: Auth Code? n Signaling Group: 3 Number of Members: 50 Page 1 of 21
Step Description 3.2.7 Proceed to Page 2 and note the default setting of the Preferred Minimum Session Refresh Interval(sec) field. This field corresponds to the lower bound of the session interval as it pertains to the SIP standards. This field is used for negotiating a minimum value for the session interval between Avaya Communication Manager and Avaya Meeting Exchange during call set up. Note that the value assigned to the Preferred Minimum Session Refresh Interval(sec) field is doubled and assigned to the Min-SE Header Field in SIP INVITE messages for calls originating from Avaya Communication Manager. Using the default setting of 600 seconds as an example, the Min-SE Header Field would be configured for 1200 seconds in SIP INVITE messages originating from Avaya Communication Manager. Conversely, the Preferred Minimum Session Refresh Interval(sec) field also applies to calls terminating on Avaya Communication Manager from Avaya Meeting Exchange (see Step 4.1.1).
add trunk-group 3 Group Type: sip TRUNK PARAMETERS Unicode Name? y Redirect On OPTIM Failure: 5000 SCCAN? n Digital Loss Group: 18 Preferred Minimum Session Refresh Interval(sec): 600 Page 2 of 21
3.3. Configure Call Routing

This section describes the steps for configuring call routing from Avaya Communication Manager to Avaya Meeting Exchange. For this sample configuration, Automatic Alternate Routing (AAR) without Feature Access Code (FAC) is utilized to route calls to Avaya Meeting Exchange. Note that other forms of call routing may be utilized. Step Description 3.3.1 Issue the command change dialplan analysis and administer settings to route calls beginning with number 5 and totaling 5 digits in length via AAR as displayed.
change dialplan analysis DIAL PLAN ANALYSIS TABLE Location: all Dialed String Total Call Length Type Page 1 of 12 1
Percent Full: Dialed String Total Call Length Type
0 1 2 3 4 5 6 7 8 9 * #
Dialed String
Total Length 2 3 3 5 5 5 5 5 1 1 2 2
Call Type dac dac aar ext aar aar ext ext fac fac fac fac
Step Description 3.3.2 Issue the command change route-pattern <n>, where n is the number of an unallocated route pattern. Administer settings to utilize the trunk group provisioned in Step 3.2.6 to route calls to Avaya Meeting Exchange. Enter the number of the trunk group that was provisioned in Step 3.2.6 in the Grp No field. To disable restrictions for call routing via this route pattern, set the Facility Restriction Level (FRL) field to the lowest setting. To send three digits to Avaya Meeting Exchange, delete two digits from the five digit string by entering 2 in the No. Del Dgts field. Note that two digits are deleted from the left-most portion of the string. Use default settings for remaining fields.
change route-pattern 3 Page Pattern Number: 3 Pattern Name: s6200 SIP SCCAN? n Secure SIP? n Grp FRL NPA Pfx Hop Toll No. Inserted No Mrk Lmt List Del Digits Dgts 1: 3 0 2 2: 3: 4: 5: 6: BCC VALUE TSC CA-TSC 0 1 2 M 4 W Request 1: 2: 3: 4: 5: 6: y y y y y y y y y y y y y y y y y y y y y y y y y y y y y y n n n n n n n n n n n n ITC BCIE Service/Feature PARM rest rest rest rest rest rest 1 of DCS/ QSIG Intw n n n n n n 3 IXC user user user user user user
No. Numbering LAR Dgts Format Subaddress none none none none none none
Step Description 3.3.3 Issue the command change aar analysis x and add entries in the table to utilize the route pattern provisioned in Step 3.3.2. Enter a number in the Dialed String field to associate with the appropriate route pattern. Enter the number of the route pattern provisioned in Step 3.3.2 in the Route Pattern field. Configure additional fields with boldface type as displayed and use default settings for remaining fields.
change aar analysis 5 AAR DIGIT ANALYSIS TABLE Location: all Total Min Max 5 5 Route Pattern 3 Call Type aar Node Num Page 1 of 2 1
Percent Full: ANI Reqd n
53
Dialed String
4. Avaya Meeting Exchange Configuration

This section describes the configuration for enabling Avaya Meeting Exchange to interoperate with Avaya Communication Manager. Call routing, call branding and SIP connectivity are administered on Avaya Meeting Exchange via a Command Line Interface (CLI) accessed via Secure Shell (SSH). Conference related attributes are administered and maintained via the Avaya Bridge Talk application. Refer to [3], [4] and [5] for additional information regarding the administration of Avaya Meeting Exchange.
4.1. Configure Connectivity

This section describes the steps for configuring SIP connectivity between Avaya Meeting Exchange and Avaya Communication Manager. The provisioning depicted in this section was administered via the CLI. Step Description 4.1.1 Administer settings that enable SIP connectivity between Avaya Meeting Exchange and Avaya Communication Manager by editing the system.cfg file as follows: From the /usr/ipcb/config directory, edit the system.cfg file with a text editor. Enter the IP address of Avaya Meeting Exchange (as defined in the /etc/hosts file) for the IPAddress variable. Enter a SIPS Uniform Resource Identifier (URI) for Avaya Meeting Exchange that conforms to SIP standards for the MyListener variable. This entry is used to populate the From Header Field in SIP INVITE messages from Avaya Meeting Exchange. To enable secure SIP connectivity utilizing TLS, this entry must contain sips, 5061 and transport=tls. The User Field, S6200, must conform to SIP standards and is selected to uniquely identify this server. For example, S6200 will be inserted in the From Header Field of SIP INVITE messages from Avaya Meeting Exchange and will display on a telephone when a call originates from Avaya Meeting Exchange. Enter a SIP-URI that conforms to SIP standards and is bounded by angled brackets for the respContact variable. This entry is used to populate the Contact Header Field in SIP Response messages from Avaya Meeting Exchange and provides Avaya Communication Manager a SIP-URI for acknowledging SIP messages from Avaya Meeting Exchange. To enable secure SIP connectivity utilizing TLS, this entry must contain, 5061 and transport=tls. Enter a value in seconds for the minSETimerValue variable. This entry corresponds to the lower bound of the session interval as it pertains to the SIP standards. It is recommended to provision the minSETimerValue variable to a setting that is greater than or equal to the corresponding setting on Avaya Communication Manager (see Step 3.2.7).
# ip address of the server IPAddress=192.168.13.102 # request we will be listening to MyListener=sips:[email protected]:5061;transport=tls # if this setting is populated will Overwrite the contact field in responses respContact=<sip:[email protected]:5061;transport=tls> # Min SE value in seconds for lower bound of Session Interval for SIP Invite minSETimerValue=1200
4.2. Configure Call Routing

The provisioning depicted in this section was administered via the CLI and describes the steps to enable call routing for Avaya Meeting Exchange, where call routing is defined as follows: For call termination on Avaya Meeting Exchange, URI to telephone number translations are utilized. These translations associate calls to Avaya Meeting Exchange with corresponding call branding, based on incoming SIP-URIs. For call origination from Avaya Meeting Exchange, telephone number to URI translations are utilized. These translations associate a telephone number pattern with a corresponding SIP-URI of a SIP User Agent (UA), thus allowing call origination from Avaya Meeting Exchange to the SIP UA.
Step Description 4.2.1 Administer settings to associate incoming calls to Avaya Meeting Exchange with corresponding call branding by adding URI to telephone number translations to the UriToTelnum.tab file. These translations extract values for both the Direct Inward Dial (DID, also known as DDI in Europe) and the Automatic Number Identification (ANI). From the /usr/ipcb/config directory, edit the UriToTelnum.tab file with a text editor. Add rules, separated by either tabs or single spaces, as a line in the file to match the pattern of the To and From Header Fields in SIP INVITE messages from Avaya Communication Manager. If the match is successful, the DID is extracted from the To Header Field and the ANI is extracted from the From Header Field. Metacharacters such as * or ? may be utilized. o The rules under the TelnumPattern and TelnumConversion columns work in conjunction as follows. Assume Avaya Communication Manager sends a SIP INVITE message with the following To and From Header Fields. The rule ""*"*<sip:*" matches the following: To: "555" <sip:[email protected]>, where $1 utilizes 555, the variable matched by the first asterisk as the DID value for the call. From: "SIP 31001" <sip:[email protected]>, where $1 utilizes SIP 31001, the variable matched by the first asterisk as the ANI for the call. [Not Required] Add rules to support operator dial-in. Refer to [4] for information regarding this feature. For this sample configuration, ""*"*<sip:501*" is utilized. Enable an undefined caller to receive a prompt for operator assistance by adding an entry for a wildcard as the last line in this file. This entry accounts for the condition of an unmatched To Header Field. Note: Entries in this file are read sequentially, therefore, the entry for the wildcard must be the last line in the file. Otherwise, all calls to Avaya Meeting Exchange would match the wildcard and thus receive a prompt for operator assistance.
# request URI to telnum conversion table # # This table converts the Request URI in the SIP INVITE request to the # appropriate value specified when a pattern is matched. For example, if the # request Uri was "<sip:[email protected]>" and one of the patterns was # "<sip:*@*" a match would take place. If the conversion for that match was # $1 then 3333 would be passed as the ddi for the call. If the conversion for # that match were "0000" then 0000 would be passed as ther ddi for the call. # #THE COMMENT COLLUM OR ANY OF THE COLLUMS SHOULD HAVE NO SPACES TelnumPattern ""*"*<sip:501*" ""*"*<sip:*" * TelnumConversion "OP501x1" $1 $0 comment Op1_From_Avaya SES_ACM wildcard
Step Description 4.2.2 Administer settings to enable call origination from Avaya Meeting Exchange to Avaya Communication Manager by adding telephone number to URI translations to the telnumToUri.tab file as follows: From the /usr/ipcb/config directory, edit the telnumToUri.tab file with a text editor. Add rules, separated by either tabs or single spaces, as a line in the file to route calls from Avaya Meeting Exchange to Avaya Communication Manager. Metacharacters such as * (refers to a character string) or ? (refers to a single character) may be utilized. o The rule entered under the TelnumPattern column matches any five digit pattern with a leading 3 and corresponds to station extensions on Avaya Communication Manager. o The SIP-URI entered under the TelnumConversion column routes the call to the CLAN on Avaya Communication Manager. To enable secure SIP connectivity utilizing TLS, the rule must syntactically conform to SIP standards regarding URI and contain 5061 and transport=tls. Avaya Meeting Exchange will replace $0 with the dialstring in outgoing SIP INVITE messages. For example, if 31002 is dialed, Avaya Meeting Exchange will format a SIP INVITE message with the following SIP-URI in the Request-Line and To Header Field: sip:[email protected]:5061;transport=tls Note: Alternatively, call routing to Avaya Communication Manager could have been enabled with either of the following entries: * sip:[email protected]:5061;transport=tls, or 3* sip:[email protected]:5061;transport=tls, where * is a wildcard and matches any digit string.
# 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 COLLUM OR ANY OF THE COLLUMS SHOULD HAVE NO SPACES TelnumPattern 3???? TelnumConversion sip:[email protected]:5061;transport=tls comment AvayaCM
4.2.3 Restart conferencing related processes on Avaya Meeting Exchange for updates to take effect. At the command prompt, enter service mx-bridge restart.
[S6200]> service mx-bridge restart
4.3. Configure Call Branding

The following steps provide examples of how to provision direct and scan call branding by utilizing the Call Branding Utility (CBUTIL) on Avaya Meeting Exchange. A command line utility, CBUTIL enables administrators to assign a specific annunciator message, line name, company name, system function, reservation group and prompt sets to a Dialed Number Identification Service (DNIS) entry. Avaya Meeting Exchange parses these entries in numerically ascending order, with the wildcard character ? last in the list. For example, 129? follows 1299. The last entry in the table consists entirely of wildcard characters. Step Description 4.3.1 Administer call branding for a direct call flow as follows: From the /usr/dcb/bin directory, add an entry to the call branding table to map the DID value obtained from procedures in Step 4.2.1 to a conference by entering cbutil add 555 0 301 1 n direct at the command prompt. The syntax for this command is case insensitive and is defined as follows: cbutil add <dnis> <rg> <msg> <ps> <ucps> <func> [-l <ln> -c <cn>], where, o <dnis> DNIS o <rg> Reservation group o <msg> Annunciator message number o <ps> Prompt set number (0-20) o <ucps> Use conference prompt set (y/n) o <func> One of: DIRECT/SCAN/ENTER/HANGUP/AUTOVL/FLEX o -l <"ln"> Optional line name to associate with caller o -c <"cn"> Optional company name to associate with caller
S6200-> cbutil add 555 0 301 1 n direct cbutil Copyright 2004 Avaya, Inc. All rights reserved.
4.3.2 Repeat Step 4.3.1 to add an entry to the call branding table for a scan call flow.
S6200-> cbutil add 500 0 1 1 n scan cbutil Copyright 2004 Avaya, Inc. All rights reserved.
Step Description 4.3.3 At the command prompt, enter cbutil list to verify the entries provisioned in Step 4.3.1 and Step 4.3.2. Note: The last entry in the call branding table, with a DNIS value ???, was added previously and is a wild card entry. This entry captures any wrong number (e.g., unmatched DID values) and places the call into the enter queue for operator assistance.
S6200-> cbutil list cbutil Copyright 2004 Avaya, Inc. All rights reserved. DNIS ---------------500 555 ??? Grp --0 0 0 Msg --1 301 208 PS --1 1 1 CP -N N N Function Line Name Company Name -------- -------------------- ----------------SCAN DIRECT ENTER
4.4. Administer Conferences

The following steps utilize Avaya Bridge Talk to provision conferences on Avaya Meeting Exchange. Avaya Bridge Talk is an application that runs on a standard Windows based PC and is utilized for provisioning and managing conferencing applications on Avaya Meeting Exchange. Refer to [5] for information regarding PC requirements. 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. Figure 2 illustrates the main window of the Avaya Bridge Talk application. The following is a brief description of the task areas that were utilized for these Application Notes. 1. The Menu Bar, which includes menus for both Avaya Meeting Exchange specific and Windows-based commands. 2. The Main Tool Bar, which includes commands for entering command-line text. 3. The Conference Room, which displays information about features and attributes for individual conferences; and lists participants, moderators and their status. 4. The Conference Navigator, which displays a portion of the conferences currently running on the bridge as well as individual conference attributes or features.
Figure 2: Avaya Bridge Talk Main Window
Step Description 4.4.1 Create a dial list of participants on Avaya Meeting Exchange. From the Avaya Bridge Talk Menu Bar, select Fast Dial New. From the New Dial List window that is displayed, add participants to the dial list as follows: Enter a descriptive name for this dial list in the Name field. Add entries to the dial list by clicking Add for each participant. o Enter a descriptive name for each participant in the Name field. o Enter a number in the Telephone field that corresponds to telephones registered to either Avaya Communication Manager or Avaya SIP Enablement Services. Enable conference participants on the dial list to enter the conference without a passcode by checking the Directly to Conf box. Refer to [5] for definitions regarding the remaining fields on this screen. Click Save.
Step Description 4.4.2 Schedule conferences that utilize the call branding for a direct call flow provisioned in Step 4.3.1 as follows. From the Menu Bar, click View Conference Scheduler. From the Conference Scheduler window that is displayed, click File Schedule Conference. From the Schedule Conference window that is displayed, administer settings as follows: Enter a unique passcode in the Conferee Code field to allow access to this conference. Enter a unique passcode in the Moderator Code field to allow access to this conference with moderator/host privileges. Note, to enable access to this conference without entering a passcode, define a Moderator Code that aligns with the provisioning for a direct call flow (see Step 4.3.1). Enter a descriptive name for this conference in the Conference Name field. Administer settings to enable a blast dial by setting the Auto Blast field to Manual and selecting the dial list provisioned in Step 4.4.1 in the Dial List field. o Select a dial list by clicking Dial List. o [Not Shown] Select a dial list from the Create, Select or Edit Dial List window that is displayed. Refer to [5] for definitions regarding the remaining fields on this screen. Click OK.
5. Interoperability Testing
5.1. General Test Approach
The general test approach was to place calls between Avaya Communication Manager and Avaya Meeting Exchange, utilizing the sample configuration displayed in Figure 1. The main objectives were to verify the following: Call origination from Avaya Communication Manager to scheduled and demand conferences provisioned on Avaya Meeting Exchange: o DNIS direct call branding (without participant-access-code) o Scan call branding (with participant-access-code) Call origination from Avaya Meeting Exchange to telephones registered to either Avaya Communication Manager or Avaya SIP Enablement Services: o Auto/manual blast dial o Originator dial-out o Operator fast dial Features available on Avaya Meeting Exchange: o Operator dial-out (Audio Path) o Operator dial-in (Audio Path) o Dial-out to a Flexible Digital Auxiliary Port Interface (FDAPI) channel for audio recording o Line transfer initiated from Avaya Bridge Talk o Conference transfer initiated from Avaya Bridge Talk o Conferencing features for both moderator and participant accessed during a conference call via touchtone commands Features available on Avaya Communication Manager: o Call hold o Attended/unattended call transfer o Call forward o Three-way conference Transport methods for signaling between Avaya Meeting Exchange and Avaya Communication Manager: o TCP utilizing TLS Transport methods for media between Avaya Meeting Exchange and Avaya Communication Manager: o RTP/UDP Codecs: o G711MU Voice quality, verified subjectively using endpoints participating in a conference DTMF transmission as defined by RFC 2833
5.2. Test Results

All test cases, as defined by the general test approach, passed.
REB; Reviewed: SPOC 6/2/2008 Solution & Interoperability Test Lab Application Notes 2008 Avaya Inc. All Rights Reserved. 24 of 30 MX50-ACM50-TLS
6. Verification Steps
The following steps were used to verify the administrative steps presented in these Application Notes and are applicable for similar configurations in the field. Step Description 6.1 Verify conferencing related processes are running on Avaya Meeting Exchange. From the Avaya Meeting Exchange CLI, enter service mx-bridge status at the command prompt and verify that a Process ID (PID) is present for all processes.
S6200-> service mx-bridge status 2373 pts/1 00:00:00 initdcb 2420 pts/1 00:00:00 log 2423 pts/1 00:00:00 bridgeTranslato 2424 pts/1 00:00:00 netservices 2431 pts/1 00:00:00 timer 2432 pts/1 00:00:00 traffic 2433 pts/1 00:00:00 chdbased 2434 pts/1 00:00:00 startd 2435 pts/1 00:00:00 cdr 2436 pts/1 00:00:00 modapid 2437 pts/1 00:00:00 schapid 2438 pts/1 00:00:00 callhand 2439 pts/1 00:00:00 initipcb 2443 pts/1 00:00:00 sipagent 2451 pts/1 00:00:00 msdispatcher 2454 pts/1 00:00:00 softms 2457 pts/1 00:00:00 serverComms 2311 pts/1 00:00:00 sqlexecd with 5 children
6.2 Verify SIP connectivity between Avaya Communication Manager and Avaya Meeting Exchange by retrieving status regarding the trunk group provisioned in Step 3.2.6. From a SAT session, issue the command status trunk <n>, where n is the number of the trunk group to verify. Verify that all members in the trunk group are in-service/idle.
Step Description 6.3 Validate signaling and media connectivity for call origination from Avaya Communication Manager to Avaya Meeting Exchange. This is accomplished by verifying that the trunk group provisioned in Step 3.2.6 is utilized when a call from a telephone registered to either Avaya Communication Manager or Avaya SIP Enablement Services dials in to a conference provisioned on Avaya Meeting Exchange. From a SAT session, issue the command list trace tac <n>, where n is the TAC defined for the trunk group. From a telephone registered to either Avaya Communication Manager or Avaya SIP Enablement Services, dial 53555 to enter the conference provisioned in Section 4.4 as moderator via the call branding for a direct call flow provisioned in Step 4.3.1. Note: The trace below shows that 53555 was dialed and utilized the call routing and trunk group provisioned in Section 3 to route the call to Avaya Meeting Exchange. Also, note the sequence to enable direct IP-to-IP audio connectivity between the endpoints (Avaya Meeting Exchange and the Avaya 4600 Series IP Telephone) involved in this call. Initially, both Avaya Meeting Exchange (192.168.13.102) and the Avaya 4600 Series IP Telephone (192.168.12.11) are connected to the Media Processor (MEDPRO) 192.168.11.11. Due to the provisioning that enabled direct IP-to-IP audio connectivity as well as congruency for the codec and DTMF requirements on the endpoints, Avaya Communication Manager allowed direct IP-to-IP audio connectivity for this call.
list trace tac 103 LIST TRACE time 10:39:05 10:39:05 10:39:05 10:39:05 10:39:05 10:39:05 10:39:05 10:39:05 10:39:05 10:39:05 10:39:05 10:39:05 10:39:06 10:39:06 data dial 53555 route:AAR term trunk-group 3 cid 0x155 dial 53555 route:AAR route-pattern 3 preference 1 cid 0x155 seize trunk-group 3 member 30 cid 0x155 Calling Number & Name 31001 SIP 31001 Proceed trunk-group 3 member 30 cid 0x155 active trunk-group 3 member 30 cid 0x155 G711MU ss:off ps:20 rn:2/1 192.168.13.102:42018 192.168.11.11:2364 xoip: fax:Relay modem:off tty:US 192.168.11.11:2364 uid:0x500f9 G711MU ss:off ps:20 rn:1/1 192.168.12.11:34008 192.168.11.11:2368 xoip: fax:Relay modem:off tty:US 192.168.11.11:2368 uid:0x50020 G711MU ss:off ps:20 rn:1/2 192.168.12.11:34008 192.168.13.102:42018 G711MU ss:off ps:20 rn:2/1 192.168.13.102:42018 192.168.12.11:34008 Page 1
Step Description 6.4 Validate signaling and media connectivity for call origination from Avaya Meeting Exchange to Avaya Communication Manager. This is accomplished by verifying that the trunk group provisioned in Step 3.2.6 is utilized when a call from a participant in a conference on Avaya Meeting Exchange is placed to a telephone registered to either Avaya Communication Manager or Avaya SIP Enablement Services. From a SAT session, issue the command list trace tac <n>, where n is the TAC defined for the trunk group. From the telephone already in conference (see Step 6.3), enter the appropriate touchtone command to initiate the blast dial feature as provisioned in Section 4.4. Note that the goal of this step is to validate call origination from Avaya Meeting Exchange to Avaya Communication Manager, thus any form of call origination from Avaya Meeting Exchange may be utilized, e.g., originator dial-out. Note: The trace below shows that a call is terminated on Avaya Communication Manager on the trunk group provisioned in Step 3.2.6. Also, note the sequence to enable direct IP-to-IP audio connectivity between the endpoints (Avaya Meeting Exchange and the Avaya 4600 Series IP Telephone) involved in this call.
list trace tac 103 LIST TRACE time 10:41:57 10:41:57 10:41:57 10:41:57 10:41:57 10:41:57 10:41:57 10:42:00 10:42:00 10:42:00 data Calling party trunk-group 3 member 32 cid 0x15b Calling Number & Name NO-CPNumber NO-CPName active trunk-group 3 member 32 cid 0x15b G711MU ss:off ps:20 rn:2/1 192.168.13.102:42028 192.168.11.11:2420 xoip: fax:Relay modem:off tty:US 192.168.11.11:2420 uid:0x500fb dial 31002 term station 31002 cid 0x15b active station 31002 cid 0x15b G711MU ss:off ps:20 rn:1/2 192.168.12.12:34008 192.168.13.102:4202 G711MU ss:off ps:20 rn:2/1 192.168.13.102:42028 192.168.12.12:34008 Page 1
Step Description 6.5 Verify that calls to and from Avaya Meeting Exchange are managed correctly, e.g., participants are added/removed from conferences. This is accomplished by utilizing the Avaya Bridge Talk application. If not already logged on, log in to the Avaya Bridge Talk application with the appropriate credentials. From the Conference Navigator, double-click the appropriate entry to open the corresponding Conference Room. Verify conference participants are added/removed from conferences by observing the Conference Navigator and/or Conference Room window. Note: The screen capture below displays the conference that was initiated in Step 6.3 and Step 6.4.
7. Conclusion
These Application Notes present a sample configuration comprised of Avaya Communication Manager and the Avaya Meeting Exchange S6200 Conferencing Server (Meeting Exchange). Employing this configuration enables call origination/termination between Avaya Communication Manager and Avaya Meeting Exchange, where the signaling is secure SIP utilizing Transport Layer Security (TLS) and the media is Real-time Transport Protocol (RTP).
8. Additional References
Avaya references are available at http://support.avaya.com. [1] Administrator Guide for Avaya Communication Manager, Issue 4, Doc ID: 03-300509, January 2008. [2] Administration for Network Connectivity for Avaya Communication Manager, Issue 13, Doc ID: 555-233-504, January 2008. [3] Meeting Exchange 5.0 S6200/6800 Administration and Maintenance Guide, Issue 2, Doc ID 04-602167, August 2007. [4] Meeting Exchange 5.0 Service Pack 1 S6200/6800 Configuration Guide, Issue 4, Doc ID 04602171, December 2007. [5] Meeting Exchange 5.0 Bridge Talk User's Guide, Doc ID 04-602163, Issue 1, August 2007. [6] Configuring secure SIP connectivity utilizing Transport Layer Security (TLS) between Avaya Communication Manager and Avaya Meeting Exchange (S6200) - Issue 1.0.
2008 Avaya Inc. All Rights Reserved.
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]

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Avaya Solution & Interoperability Test Lab

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Figure 1: Sample Configuration

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2. Equipment and Software Validated

Table 1: Equipment and Software Versions

3. Avaya Communication Manager Configuration

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3.1. Verify Licensing

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3.2. Configure Connectivity

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Name CLAN-1A02 MEDPRO-1A03 SES s6200

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Direct IP-IP Audio Connections? y IP Audio Hairpinning? n

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3.3. Configure Call Routing

Percent Full: Dialed String Total Call Length Type

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Percent Full: ANI Reqd n

4. Avaya Meeting Exchange Configuration

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4.1. Configure Connectivity

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4.2. Configure Call Routing

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4.3. Configure Call Branding

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4.4. Administer Conferences

Figure 2: Avaya Bridge Talk Main Window

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5.2. Test Results

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2008 Avaya Inc. All Rights Reserved.

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