Abstract. Avaya Solution & Interoperability Test Lab

Transcription

1 Avaya Solution & Interoperability Test Lab Application Notes for ObjectTel CLASSONE ICAS with Avaya Aura Application Enablement Services, Avaya Aura Communication Manager and Avaya Meeting Exchange Issue 1.0 Abstract These Application Notes contain instructions for ObjectTel CLASSONE ICAS and Avaya Aura Application Enablement Services, Avaya Aura Communication Manager and Avaya Meeting Exchange to successfully interoperate. Information in these Application Notes has been obtained through DevConnect compliance testing and additional technical discussions. Testing was conducted via the DevConnect Program at the Avaya Solution and Interoperability Test Lab. 1 of 45

2 1. Introduction These Application Notes contain instructions for ObjectTel CLASSONE ICAS and Avaya Aura Application Enablement Services, Avaya Aura Communication Manager and Avaya Meeting Exchange to successfully interoperate. The tested CLASSONE icas solution consisted of the following components: CLASSONE icas Application Server CLASSONE icas Operator Console H.323 CLASSONE icas Radio Gateway - SIP The CLASSONE ICAS is a system-of-systems enabling operators to take control of their communication network and manage multiple transactions from many types of devices CLASSONE ICAS enables operators to handle inbound calls, connect with radio dispatch, bridge various radio talk groups and frequencies with each other and with back office voice systems, collaborate and manage field operations regardless of the type of voice-enabled device, while maintaining the highest level of business continuity and interoperability. 2 of 45

3 2. General Test Approach and Test Results DevConnect Compliance Testing is conducted jointly by Avaya and DevConnect members. The jointly-defined test plan focuses on exercising APIs and/or standards-based interfaces pertinent to the interoperability of the tested products and their functionalities. DevConnect Compliance Testing is not intended to substitute full product performance or feature testing performed by DevConnect members, nor is it to be construed as an endorsement by Avaya of the suitability or completeness of a DevConnect member s solution Interoperability Compliance Testing During Interoperability Compliance testing, functional call routing scenarios were tested: Placing calls to/from icas Operator Console and Radio Gateway Hold and Transfer scenarios Conference calls onto Avaya Meeting Exchange Endpoint Management using icas Application Server Please note that performance testing or load testing were not part of this test effort Test Results All planned test cases were passed 2.3. Support CLASSONE icas support can be obtained via following means: Phone: Web: rtisupport@objecttel.com 3 of 45

4 3. Reference Configuration Figure 1 illustrates a sample configuration that consists of Avaya Products and ObjectTel CLASSONE icas. Figure 1: Test Configuration of CLASSONE icas and Avaya Products 4 of 45

5 4. Equipment and Software Validated The following equipment and software were used for the sample configuration provided: Equipment/Software Release/Version Avaya S8300D Server Avaya Aura Communication Manager 6.3 SP1 Avaya G450 Media Gateway Avaya Aura Application Enablement Services 6.3 Avaya Meeting Exchange 5.2 Avaya Aura Session Manager Avaya Aura System Manager ObjectTel CLASSONE icas of 45

6 5. Configure Avaya Aura Communication Manager This section contains steps necessary to configure ObjectTel CLASSONE ICAS successfully with Avaya Aura Communication Manager. All configurations in Communication Manager were performed via SAT terminal Administer IP Network Region Use the change ip-network-region n command to configure a network region, where n is an existing network region. Configure this network region as follows: Set Location to 1 Set Codec Set to 1 Set Intra-region IP-IP Direct Audio to yes Set Inter-region IP-IP Direct Audio to yes Enter and Authoritative Domain, e.g. avaya.com change ip-network-region 1 Page 1 of 20 IP NETWORK REGION Region: 1 Location: 1 Authoritative Domain: avaya.com Name: 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 Call Control PHB Value: 46 Audio PHB Value: 46 Video PHB Value: P/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 6 of 45

7 5.2. Administer IP Codec Set Use the change ip-codec-set n command to configure IP codec set, where n is an existing codec set number. Configure this codec set as follows, on Page 1: Set Audio Codec 1 to G.711MU change ip-codec-set 1 Page 1 of 2 Codec Set: 1 IP Codec Set Audio Silence Frames Packet Codec Suppression Per Pkt Size(ms) 1: G.711MU n : 3: 4: 5: 6: 7: 1: 2: 3: Media Encryption 5.3. Administer IP Node Names Use the change node-names ip command to add an entry for Session Manager. For compliance testing, sm and entry was added. change node-names ip Page 1 of 2 IP NODE NAMES Name IP Address default msgsrvr procr procr6 :: sm of 45

8 5.4. Administer SIP Signaling Group Use the add signaling-group n command to add a new signaling group, where n is an available signaling group number. Configure this signaling group as follows: Set Group Type to sip Set Near-end Node Name to procr Set Far-end Node Name to the configured Session Manager in Section 5.4, i.e. sm Set Far-end Network region to the configured region in Section 5.2, i.e. 1 Enter a Far-end Domain, e.g. avaya.com add signaling-group 1 Page 1 of 2 SIGNALING GROUP Group Number: 1 Group Type: sip IMS Enabled? n Transport Method: tls Q-SIP? n IP Video? n Peer Detection Enabled? y Peer Server: Others Enforce SIPS URI for SRTP? y Near-end Node Name: procr Far-end Node Name: sm Near-end Listen Port: 5061 Far-end Listen Port: 5061 Far-end Network Region: 1 Far-end Domain: avaya.com Bypass If IP Threshold Exceeded? n Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y Session Establishment Timer(min): 3 IP Audio Hairpinning? n Enable Layer 3 Test? y Initial IP-IP Direct Media? n H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6 Note: Signaling Group, Trunk Group and Route Pattern for simulated PSTN calls for inter-site calls over ISDN/PRI and SIP were pre-configured and are not shown in this document. 8 of 45

9 5.5. Administer SIP Trunk Group Use the add trunk-group n command to add a trunk group, where n is an available trunk group number. Configure this trunk group as follows, on Page 1: Set Group Type to sip Enter a Group Name, e.g. SM Enter a valid TAC, e.g. *001 Set Service Type to tie Enter Signaling Group value to the signaling group configured in Section 5.5, i.e. 1 Enter a desired number in Number of Member field add trunk-group 1 Page 1 of 21 TRUNK GROUP Group Number: 1 Group Type: sip CDR Reports: y Group Name: SM COR: 1 TN: 1 TAC: *001 Direction: two-way Outgoing Display? n Dial Access? n Night Service: Queue Length: 0 Service Type: tie Auth Code? n Member Assignment Method: auto Signaling Group: 1 Number of Members: 25 On Page 3: Set Number Format to private add trunk-group 1 Page 3 of 21 TRUNK FEATURES ACA Assignment? n Measured: none Maintenance Tests? y Numbering Format: private UUI Treatment: service-provider Replace Restricted Numbers? n Replace Unavailable Numbers? n 9 of 45

10 5.6. Administer Route Pattern Use the change route-pattern n command to configure a route pattern, where n is an available route pattern. Configure this route pattern as follows: Type a name in Pattern Name field For line 1, set Grp No to the trunk group configured in Section 5.6, i.e. 1 For line 1, set FRL to 0 change route-pattern 1 Page 1 of 3 Pattern Number: 1 Pattern Name: Voice and Fax 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: 1 0 n user 2: n user 5.7. Administer Private Numbering Use the change private-numbering 1 command to define the calling party number to send to Session Manager. Configure private numbering as follows: Add entries for trunk group configured in Section 5.6 Note: For compliance testing, 5-digit extensions beginning with 29 routed over trunk group 1 resulted in a 5-digit calling party number. change private-numbering 1 Page 1 of 2 NUMBERING - PRIVATE FORMAT Ext Ext Trk Private Total Len Code Grp(s) Prefix Len Total Administered: 1 Maximum Entries: of 45

11 5.8. Configure AAR Analysis Use change aar analysis n command to add an entry in aar table, where n is an extension number that will be used to route calls to Meeting Exchange. change aar analysis Page 1 of 2 AAR DIGIT ANALYSIS TABLE Location: all Percent Full: 2 Dialed Total Route Call Node ANI String Min Max Pattern Type Num Reqd aar n 5.9. Configure Vectors Use change vector n to configure a Vector, where n is an available Vector number. F change vector 1 Page 1 of 6 CALL VECTOR Number: 1 Name: icas Multimedia? n Attendant Vectoring? n Meet-me Conf? n Lock? n Basic? y EAS? y G3V4 Enhanced? y ANI/II-Digits? y ASAI Routing? y Prompting? y LAI? y G3V4 Adv Route? y CINFO? y BSR? y Holidays? y Variables? y 3.0 Enhanced? y 01 wait-time 2 secs hearing ringback 02 route-to number with cov n if unconditionally 03 stop of 45

12 5.10. Configure VDN Use add vdn n to add a vdn, where n is an available vdn extension. On Page 1: In the Name field, enter a descriptive name In the Destination field, set Vector Number to the vector configured earlier in this document. i.e., Vector Number 1. Set Measured to both add vdn Page 1 of 3 VECTOR DIRECTORY NUMBER Extension: Name*: ObjectTel Classone icas Destination: Vector Number 1 Attendant Vectoring? n Meet-me Conferencing? n Allow VDN Override? n COR: 1 TN*: 1 Measured: both Acceptable Service Level (sec): 20 VDN of Origin Annc. Extension*: 1st Skill*: 2nd Skill*: 3rd Skill*: 12 of 45

13 5.11. Configure AES connection Use change ip-services command to add an entry for AES. On Page 1, In the Service Type field, type AESVCS. In the Enabled field, type y. In the Local Node field, type the Node name procr for the Processor Ethernet Interface. In the Local Port field, use the default of change ip-services Page 1 of 4 IP SERVICES Service Enabled Local Local Remote Remote Type Node Port Node Port AESVCS y procr 8765 On Page 4 of the IP Services form, enter the following values: In the AE Services Server field, type the name obtained from the Application Enablement Services server. In the Password field, type a password to be administered on the Application Enablement Services server. In the Enabled field, type y. change ip-services Page 4 of 4 AE Services Administration Server ID AE Services Password Enabled Status Server 1: aes6_tr1 devconnect123 y in use 2: AES_21_46 Interop y in use 3: 4: 5: 6: 7: 8: 9: 10: 11: 12: 13: 14: 15: 16: 13 of 45

14 5.12. Configure CTI Link Use add cti-link n command, where n is an available CTI link number. In the Extension field, type a valid station extension, in this case In the Type field, type ADJ-IP. In the Name field, type a descriptive name. add cti-link 1 Page 1 of 3 CTI LINK CTI Link: 1 Extension: 6201 Type: ADJ-IP COR: 1 Name: TSAPI 14 of 45

15 6. Configure Avaya Aura Application Enablement Services Configuration of Avaya Aura Application Enablement Services requires a user account be configured for ObjectTel CLASSONE icas Configure User All administration is performed by web browser, A user needs to be created for ObjectTel CLASSONE ICAS to communicate with AES. Navigate to User Management User Admin Add User. Fill in User Id, Common Name, Surname, User Password and Confirm Password. Set the CT User to Yes, and Apply. 15 of 45

16 Navigate to Security Security Database CTI Users List All Users. Select the recently added user and click Edit. Check the box for Unrestricted Access and click Apply Changes. 16 of 45

17 6.2. Configure Communication Manager Switch Connections To add links to the Communication Manager, navigate to the Communication Manager Interface Switch Connections page and enter a name for the new switch connection and click the Add Connection button. This was previously configured as TR18300 for this test environment: Use the Edit Connection button shown above to configure the connection. Enter the Switch Password and check the Processor Ethernet box if using the procr interface, as shown below. This must match the password configured when adding AESVCS connection in Communication Manager. Use the Edit PE/CLAN IPs button (shown in this section s first screen capture above) to configure the procr or CLAN IP Address(es). 17 of 45

18 Use the Edit H.323 Gatekeeper button (shown in this section s first screen capture above) to configure the procr or CLAN IP Address(es) Configure TSAPI Link Navigate to the AE Services TSAPI TSAPI Links page to add the TSAPI CTI Link. Click Add Link (not shown). Select a Switch Connection using the drop down menu. Select the Switch CTI Link Number using the drop down menu. The Switch CTI Link Number must match the number configured in the cti-link form for Communication Manager. If the application will use Encrypted Links, select Both in the Security selection box. Click Apply Changes. Configuration shown below was previously configured. 18 of 45

19 7. Configure Avaya Aura Session Manager This section provides the procedures for configuring Session Manager, assuming it has been installed and licensed. The procedures include the following items: Specify SIP Domain Add Locations Add SIP Entities Add Entity Links Add Routing Policies Add Dial Patterns Configuration is accomplished by accessing the browser-based GUI of System Manager, using the URL where <ip-address> is the IP address of System Manager. Log in with the appropriate credentials. The menu shown below is displayed. Click on Routing. 19 of 45

20 7.1. Specify SIP Domain Specify the SIP domain for which the communications infrastructure will be authoritative. Select Domains on the left and click the New button on the right. The following screen will be shown. Fill in the following fields and click Commit. Name: The authoritative domain name (e.g. avaya.com) Type Select sip Notes: Descriptive text (optional) 7.2. Add Locations Locations can be used to identify logical and/or physical locations where SIP Entities reside, for the purpose of bandwidth management. To add a location, select Locations on the left and click on the New button on the right. The following screen will be shown. Fill in the following fields: Under General: Name: Notes: Under Location Pattern: IP Address Pattern: Notes: A descriptive name Descriptive text (optional) A pattern used to logically identify the location. In these Application Notes, the pattern represented the networks involved, i.e * and *. Descriptive text (optional) 20 of 45

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22 7.3. Add SIP Entities A SIP Entity must be added for Session Manager and for each SIP-based telephony system connected to it using SIP trunks. In the sample configuration, a SIP Entity is added for Communication Manager, Meeting Exchange, and Session Manager Adding Avaya Aura Communication Manager Select SIP Entities on the left and click on the New button on the right. Fill in the following fields. Under General: Name: A descriptive name FQDN or IP Address: IP address of the procr interface of Communication Manager, i.e Type: Select CM Location: Select the location defined in Section 7.2 Time Zone: Time zone for this entity Defaults can be used for the remaining fields. Click Commit to save the SIP Entity definition. The following screen shows the configuration of the Communication Manager SIP Entity. 22 of 45

23 Adding Avaya Meeting Exchange Select SIP Entities on the left and click on the New button on the right. Under General: Name: A descriptive name FQDN or IP Address: IP address of the signaling interface of Meeting Exchange, i.e Type: Select SIP Trunk Location: Select the location defined in Section 7.2 Time Zone: Time zone for this entity Defaults can be used for the remaining fields. Click Commit to save the SIP Entity definition. The screen below shows the configuration of the Meeting Exchange SIP Entity. 23 of 45

24 Adding Avaya Aura Session Manager Select SIP Entities on the left and click on the New button on the right. Under General: Name: A descriptive name FQDN or IP Address: IP address of the Session Manager signaling interface i.e Type: Select Session Manager Location: Select the location defined in Section 7.2 Time Zone: Time zone for this entity Create three Port definitions for TLS, TCP, and UDP. Under Port, click Add, and edit the fields in the resulting new row as shown below: Port: Port number on which the system listens for SIP requests Protocol: Transport protocol to be used to send SIP requests Default Domain The domain used (e.g., avaya.com) Defaults can be used for the remaining fields. Click Commit to save the SIP Entity definition. 24 of 45

25 7.4. Add Entity Links A SIP trunk between Session Manager and a telephony system is described by an Entity link. To add an Entity Link, select Entity Links on the left and click on the New button on the right. Fill in the following fields in the new row that is displayed: Name: SIP Entity 1: Port: SIP Entity 2: Port: Trusted: Protocol: A descriptive name Select the SessionManager SIP Entity Port number to which the far end system sends SIP requests Select the name of the far end system Port number on which the far end system receives SIP requests Check this box, otherwise calls from the SIP Entity specified will be denied Select the transport protocol to align with the far end. In these Application Notes TLS was used for Communication Manager and TCP for Meeting Exchange. Click Commit to save each Entity Link definition. The following screen illustrates adding the Entity Link for Communication Manager. The screen below illustrates the Entity Link for Meeting Exchange. 25 of 45

26 7.5. Add Routing Policies Routing policies describe the condition under which calls will be routed to the SIP Entities specified in Section 7.3. Two routing policies were added: one for Communication Manager and another for Meeting Exchange. To add a routing policy, select Routing Policies on the left and click on the New button on the right. The following screen is displayed. Fill in the following fields: Under General: Enter a descriptive name in Name Under SIP Entity as Destination: Click Select, and then select the appropriate SIP entity to which this routing policy applies Under Time of Day: Click Add, and select the time range configured. In these Application Notes, the predefined 24/7 Time Range is used Defaults can be used for the remaining fields. Click Commit to save each Routing Policy definition. The following picture shows the Routing Policy for Communication Manager. 26 of 45

27 The following screen shows the Routing Policy for Meeting Exchange. 27 of 45

28 7.6. Add Dial Patterns Dial patterns must be defined that will direct calls to the appropriate SIP Entity. To add a dial pattern, select Dial Patterns on the left and click on the New button on the right. Fill in the following fields as specified for the dial pattern that routes calls to Communication Manager: Under General: Pattern: Dialed number or prefix, 25 Min: Minimum length of dialed number, 5 Max: Maximum length of dialed number, 5 SIP Domain: Select -ALL- Under Originating Locations and Routing Policies, click Add, and then select the appropriate location and routing policy from the list. Default values can be used for the remaining fields. Click Commit to save the dial pattern. The following screen shows the configured dial pattern. 28 of 45

29 Repeat the process to add one or more dial patterns for routing calls to Meeting Exchange. Fill in the following fields as specified to route calls to Meeting Exchange: Under General: Pattern: Dialed number or prefix, Min: Minimum length of dialed number, 5 Max: Maximum length of dialed number, 5 SIP Domain: Select -ALL- Under Originating Locations and Routing Policies, click Add, and then select the appropriate location and routing policy from the list. Default values can be used for the remaining fields. Click Commit to save the dial pattern. The following screen shows the configured dial pattern Configure SIP Users During the compliance test, no special users were created for this solution. All users were created prior to the compliance test. However, the steps to configure a user are included. Add new SIP users for each icas Radio Gateway line. To add new SIP users, Navigate to Home Users User Management Manage Users. Click New (not shown) and provide the following information: Identity section o Last Name Enter last name of user. 29 of 45

30 o First Name Enter first name of user. o Login Name Enter extension number@sip domain name. The domain name is defined in Section 5.3. o Authentication Type Verify Basic is selected. o SMGR Login Password Enter password to be used to log into System Manager. o Confirm Password Repeat value entered above. o Enter Localized Display Name o Enter Endpoint Display Name o Select English as Language Preference o Set the appropriate Time Zone. Communication Profile section Provide the following information: o Communication Profile Password Enter a numeric value used to logon to SIP telephone. o Confirm Password Repeat numeric password 30 of 45

31 Verify there is a default entry identified as the Primary profile for the new SIP user. If an entry does not exist, select New and enter values for the following required attributes: o Name Enter Primary. o Default Enter Communication Address sub-section Select New to define a Communication Address for the new SIP user, and provide the following information. o Type Select Avaya SIP using drop-down menu. o Fully Qualified Address Enter same extension number and domain used for Login Name, created previously. Click the Add button to save the Communication Address for the new SIP user. 31 of 45

32 Session Manager Profile section o Primary Session Manager Select one of the Session Managers. o Secondary Session Manager Select (None) from drop-down menu. o Survivability Server Select (None) from drop-down menu. o Origination Sequence Select Application Sequence defined (not shown) for Communication Manager. o Termination Sequence Select Application Sequence defined (not shown) for Communication Manager. o Home Location Select Location defined in Section 6.2. Endpoint Profile section o System Select Managed Element defined in System Manager (not shown) for Communication Manager. o Use Existing Endpoints - Leave unchecked to automatically create a new endpoint on Communication Manager when the new user is created. Or else, check the box if endpoint is already defined in Communication Manager. o Extension - Enter same extension number used in this section. o Template Select template for type of SIP phone. During the compliance test, 9641SIP_DEFAULT_CM_6_3 was selected. o Security Code Enter numeric value used to logon to SIP telephone. (Note: this field must match the value entered for the Shared Communication Profile Password field.) 32 of 45

33 o Voice Mail Number Enter Pilot Number for Avaya Modular Messaging if installed. Or else, leave field blank. This feature is not used during the compliance test. 33 of 45

34 8. Configure Avaya Meeting Exchange 8.1. Create login users for Avaya Meeting Exchange Log in to Meeting Exchange using appropriate credentials as a super user. Type in dm to launch the System Maintenance Main Menu window Go to Create Maintenance Sign-In window by navigating Administrator Menu Sign-In Management Create Operator Sign-In Create login user by typing in Sign-In Name and Password, followed by ESC, then Y to save. e.g, userid=rtisvr1 and password=rti ( the user type should be administrator or operator type) 34 of 45

35 Verify the newly created users by logging in to them via AVAYA Bridge Talk Continue from above, Administrator Menu Configure Conference Scheduler. This change will force non-connected participants to be HANG-UP, instead of ENTER. Change to HANG-UP from ENTER Configure Conference Scheduler Group Name : schedule Status : ENABLED Invalid Code : ENTER Timeout : ENTER Conference Secured : ENTER Max. Lines Reached : ENTER Invalid Time of Day : ENTER Scan Time (5-20) : 10 Scan Attempts (1-3) : 3 Auto Hang-up : DISABLED More-Next/Prev Page ESC to Exit 35 of 45

36 8.2. DTMF Tone Configuration Continue from above, Administrator Menu Configuration System Config, go to page 2: Set LogUserTransaction to ON Set DTMF Passthrough as ON and DTMF Regeneration as OFF Set softmediaserver.cfg values on Meeting Exchange vi softmediaserver.cfg (not shown), set following values: 1. dtmftminlevel= dtmfreversetwistdifference= dtmfforwardtwistdifference=-11 (MX version 5.x has duplicated tone detection issue. The loss plan for the MX bridge may need to be adjusted on the PBX, i.e. on location-parameters form in the Avaya PBX. Adjust all of the TO parameters for loss group 17 from 3 to -3) To test and validate tone detection in dynamic log display: tail -f /usr/dcb/apilog/usertranslog To have MX pick up the change (this is not a complete reboot): type bridge restart 36 of 45

37 8.3. Authorize CLASSONE to mute/unmute on participants and moderator lecture turned on Log into dm and navigate Administrator Menu Configuration System Configuration Set Self Mute as Anyone (or Participant) Set Moderator Lecture to ON Restart mx-bridge by command: service mx-bridge restart 37 of 45

38 8.4. Firewall Settings Enable unencrypted modapi cd /etc/firewall/conf vi mx_firewall.rulls Search for Unencrypted Modapi and remove the # from the runrule-items below it. Change from #runrule "-A INPUT --protocol tcp --destination-port jump ACCEPT" #runrule "-A INPUT --protocol tcp --destination-port jump ACCEPT" To runrule "-A INPUT --protocol tcp --destination-port jump ACCEPT" runrule "-A INPUT --protocol tcp --destination-port jump ACCEPT" 8.5. Mute Audio Messages Mute the audio message You are now muted (i.e. when dispatcher monitors a radio) Rename the file which corresponds to the audio message 340 ( You are now muted ) (i.e. mv /usr2/prompts/set0/messages/340 /usr2/prompts/set0/messages/340.org ) Rename the file which corresponds to the audio message 341 ( You are no longer muted ) (i.e. mv /usr2/prompts/set0/messages/341 /usr2/prompts/set0/messages/341.org ) 8.6. CBUTIL Config cbutil help for detailed command information. An example: 38 of 45

39 8.7. Schedule a heartbeat conference for each CLASSONE CFBrSrv instance via AVAYA Bridge Talk 1. Start AVAYA Bridge Talk 2. Open Conference Scheduler window by: View menu Conference Scheduler 3. Open Schedule Conference window by: Conference Scheduler s File menu Schedule Conference Fill in data in 'Schedule Conference', refer following screen shot as an example NOTE: 'Conferee Code' field uses the following naming rule - Prefix (The digit immediate before the last digit) 1 for group A and 2 for group B - (The last digit) 2 represent the server# ('2' represents server 2 or server 12) 'Name' and 'Conference Name' fields may be the same. It should indicate the corresponding Meeting Exchange, the failover group and the embedding server. An example is CFBR8044AS12. 'Confirmation No' field must be unique. If error pups up, increase this number 39 of 45

40 8.8. Schedule a conference on AVAYA Bridge Talk for each radio VDN that is registered in CM CLASSONE icas requires every radio VDN defined in Communication Manager and its Conference Code defined on every Bridge to be identical. 1. Start AVAYA Bridge Talk 2. Open Conference Scheduler window by: View menu Conference Scheduler 3. Open Schedule Conference window by: Conference Scheduler s File menu Schedule Conference Fill in data in Schedule Conference, refer following screen shot as an example NOTE: 'Name', 'Conference Name' and 'Conferee Code' fields may all use radio's VDN. 'Maximum Line' field is set up according to business requirement. In Objecttel lab, we set it to 10. (The maximum line should be set to auto-expandable normally on the MX) 'Confirmation No' field must be unique. If error pups up, increase this number. Moderator Code must be unique set DTMF Pass Through to System (if not visible, set as following, for Bridge Talk 5.2 or later) 40 of 45

41 Right click Avaya Bridge Talk shortcut, select Properties then select Find Target to get into the directory where Avaya Bridge Talk.ext installed, edit the template.xml file, change: <Property value="false" type="boolean" name="enabledtmfpassthrough" hidden="false" /> to <Property value="true" type="boolean" name="enabledtmfpassthrough" hidden="false" /> 41 of 45

42 9. Configure ObjectTel CLASSONE icas Configuration of ObjectTel CLASSONE icas is done by designated ObjectTel engineers. Hence, no configuration is provided in this document. 42 of 45

43 10. Verification Steps Via a SAT terminal, verify that Application Enablement Services is enabled and listening using the status aesvcs interface command. status aesvcs interface AE SERVICES INTERFACE STATUS Local Node Enabled? Number of Status Connections procr yes 2 listening Verify communication between Communication Manager and the Application Enablement Services server using the status aesvcs link command. status aesvcs link AE SERVICES LINK STATUS Srvr/ AE Services Remote IP Remote Local Node Msgs Msgs Link Server Port Sent Rcvd 02/01 AES_21_ procr /02 aes6_tr procr of 45

44 11. Conclusion ObjectTel CLASSONE ICAS was able to successfully interoperate with Avaya Aura Communication Manager, Avaya Aura Application Enablement Services and Avaya Meeting Exchange. All executed test cases were passed. 12. Additional References This section references the product documentation relevant for these Application Notes. [1] Administering Avaya Aura Communication Manager, Document , Release 6.3, Issue 10, June [2] Administering Avaya Aura Application Enablement Services Administration and Maintenance Guide, Release 6.3, Document , June Documentation related to ObjectTel CLASSONE ICAS can be directly obtained from ObjectTel. 44 of 45

45 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 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 DevConnect Program at 45 of 45