Motorola Solutions Integration Guide TEAM WSM to Avaya Session Manager with Nortel CS1000 Trunk Side Integration

Motorola Solutions Integration Guide TEAM WSM to Avaya Session Manager with Nortel CS1000 Trunk Side Integration 1

Integration Guide TEAM WSM to Avaya Session Manager and CS1000 Table of Contents 1.0 Summary... 3 2.0 Pre-requisite... 3 3.0 Network Environment... 3 4.0 Supported Features... 4 5.0 Lab Diagram... 5 6.0 PBX Setup Procedure... 6 6.1 Configuring CS1000 - Connection to Session Manager... 6 6.2 Procedure to Setup Session Manager Connections... 23 6.3 Session Manager SIP Trunk Connection to WSM Server... 23 6.4 Session Manager Trunk Connection to CS1000... 29 7.0 Motorola WSM Server Setup Procedure... 34 7.1 WSM Configuration to Session Manager... 34 7.2 WSM - Wireless Subscriber Configuration... 38 7.3 Troubleshooting Logs on WSM Server... 39 2

Integration Guide TEAM WSM to Avaya Session Manager and CS1000 Trunk Side 1.0 Summary This configuration setup covers the of the Motorola TEAM WSM server as a mobile handset solution and expansion to an Avaya Session Manager and CS1000 PBX Gateway. The Avaya Session Manager will front end the connection to the WSM for SIP connectivity via a Trunk Side Integration, as well as interconnect the trunk to the CS1000 PBX. This document is to be used as a reference only as specific configurations will be different from each system installation. Proper planning is essential to coordinate configurations and interworking between the integrated systems and networks referenced. 2.0 Pre-requisite Properly setup TEAM Environment (DNS, DHCP, AAA, etc) o Must have A record and ptr record for the WSM in the DNS server. o DHCP Server with option 43 set for over the air provisioning of Motorola EWP Handsets o AAA is set to internal on the WSM for Mobile Handsets WSM o Release 2.2.15 EWP o Release P1.72_REL01.34.08F-US o Mobile Handset Users = 300x Avaya Session Manager o Release 6.2 CS1000 PBX Gateway o CS1000E with High Availability R7.5 o Voicemail Pilot = 5000 o Deskset Users = 400x 3.0 Network Environment Nortel CS1000 UCM IP @ = 135.9.152.33 SIPGW Node 1001 IP @ = 135.9.152.32 SIPLineSide IP @ = 135.9.152.40 Avaya Session Manager Session Manager Management IP @ = 135.9.169.21 Session Manager SIP Entity IP @ = 135.9.169.22 Motorola WSM WSM Server IP @ = 10.129.132.131/28 3

4.0 Supported Features Abbreviated Dialing Basic Call Call Forwarding Call Fwd Activation Call Fwd Status Subscription Call Hold / Resume Call Transfer (Attended) Call Transfer (Blind) Call Waiting CLID (caller id number) CNID (caller id name) DID/DOD Media Shuffle Multiple (4) Call Appearances MWI Over Dial / DTMF (RFC 2833) Vocoder Support Voicemail Simultaneous Ringing 4

5.0 Lab Diagram 5

6.0 PBX Setup Procedure 6.1 Configuring CS1000 - Connection to Session Manager 1. Log in to the CS 1000 through the Avaya Unified Communications Management terminal, in this example it will be SIP GW Node 1001 IP: 135.9.152.32 2. Once Logged into the CS 1000 GW Node, goto the SIP node IP information of the Call server under the Network/Elements Tab on the right hand side menu. 3. Select the appropriate CS1000 element from the list of servers. 6

4. Next click on System/IP Network link on the left hand side menu on the CS1000 Element Manager to expand the menu. 5. Next Select the IP Network/Nodes, Servers, Media Cards to view the appropriate SIP GW Node. In this case, Node 1001 at IP address 135.9.152.32. 6. Select the SIP GW Node 1001, in this case, to view details. 7

7. Scroll down the highlighted box to find the Gateway link. Click on Gateway link for the SIP trunk configuration. This opens up the Virtual Trunk GW Details to connect to the Avaya Session Manager. 8. Make sure domain name and SIP port 5060 are setup correctly on both Session Manager and CS1000. In this case domain name = avayalabx.com. 8

9. In addition Scroll Down for more details. 10. Scroll down to the Proxy or Redirect Server Parameters 11. Add the Session Manager SIP entity IP Address as the Primary TLAN IP @ and Port 5060. See Above. 12. Also check the option support registration. 9

13. Scroll down. 14. Under SIP URI Map add the National and Subscriber numbers. In this case, +1 for North American National and +402 for the Subscriber. 10

15. Leave everything else as default. Make sure the Private Domain Names are filled in as seen above. 16. Next Click on the Save Button to save the configuration. 17. Once back in the Node Details section Click Save Again 11

18. The Message that the node configuration is saved on the call server, will then appear. The new configuration must be then transferred to the associated servers and media cards. Click the Transfer Now button to execute this transfer. 19. Select all Host Names to transfer to and then click Start Sync 12

20. After the servers are Synchronized; Click Restart Application. This will take 5 minutes to come up. Make sure system can be restarted if in production environment as it can affect attached devices. 13

21. Next go to System/Zones and then select Bandwidth Zones from the side menu. 22. Select the Add button to add a bandwidth zone for this IP connection. 14

23. Check with local network IT team for Best Quality options and strategy. This is optional, but should be investigated for proper voice communications over IP network. 15

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24. Next, it is needed to create the signaling D Channel for the SIP trunk by navigating to the Routes and Trunks/D-Channels link on the left hand side configuration menu. 25. Create the D Channel. In this example Channel 200 is being used for this configuration. 26. Enter the following parameters to create the D signaling channel to the Session Manager. D Channel Card Type: Designator: User: Interface Type for D Channel: Meridian 1 Node Type: Release ID of switch at far end: DCIP SIP Integrated Services Signaling Link Dedicated ISLD Meridian1 SL1 Slave to controller USR 25 (can be whatever you select) 17

27. Then Click on Basic Options (BSCOPT) link on the lower left side of the screen. 28. Set the following parameters. Output request Buffers: 128 D channel Transmission Rate: 56kbs when LCMT is AMI Channel Negotiation Option: No alternative acceptable exclusive 29. Next Click on the Edit button on Remote Capabilities parameter. 18

30. Scroll Down 31. Then select Message Waiting Interworking with DMS-100 mwi and Network Name display method 2 ND2 parameters. 32. Scroll down and click Return-Remote Capabilities 19

33. Once back to the main D Channels section. Click Submit. 34. Next it is needed to create the Route and then the Trunks between CS1000 and the Avaya Session Manager. 20

35. Navigate to the Routes and Trunks/Routes and Trunks and Select Add Route. 36. Under Routes and Trunks change the signaling arrangement (SIGO) to Standard STD and Click Submit to save changes. 21

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37. Click Save. 38. After this set up Digit manipulation, Route list Block, and coordinated dialing plan as needed on the CS1000. The CS1000 will need to have a dial plan created to pass calls to and accept calls from the WSM and Session Manager. In this case 6.2 Procedure to Setup Session Manager Connections Next the Session Manager server will be accessed to create a SIP Trunk to the Motorola WSM and CS1000 PBX. The Session Manager is the mandatory connection for integration between CS1000 and WSM server appliance and resides in between the two systems. 6.3 Session Manager SIP Trunk Connection to WSM Server 1. Login to the Avaya System Manager using its IP address via web browser. In this case, https://135.9.169.24/ 23

2. Select Routing from the Elements Menu on the main screen once logged in. 3. Next go to Routing/SIP Entities on the left side menu, to add the WSM as a destination route. 24

4. Click on New and make sure the settings are same as WSM example above. 5. Enter the Name as WSM and the FQDN or IP Address of the WSM 6. Click Commit and then Add Sip Entity links 25

7. Select New and have the settings similar to WSM Box highlighted Name : mefp62-sm_wsm-5060_tcp SIP Entity 1: mefp62-sm (local session manager) Protocol : TCP Port: 5060 SIP Entity 2: WSM Port: 5060 Connection Policy: Trusted 8. Hit Commit to save the configuration 9. Now go to Routing Policies. 26

10. Click New to create a new route policy. 11. In the General section, enter a name for this new policy. In this example, it is Session Manager to WSM and select the SIP Entity as Destination as the WSM previously created in earlier steps. 12. Select the destination as WSM as highlighted and then go to click on Commit to save configuration. 27

13. Select New and make the settings similar to Number 3 pattern highlighted above. In this example, a pattern of 3XXX will be used to communicate to the WSM Mobile handset users. In the General Section of Dial Patterns enter pattern = 3, Min and Max = 4. This will be for a 3XXX pattern to be used to WSM. 14. Hit Commit to save configuration. 28

6.4 Session Manager Trunk Connection to CS1000 15. Next create a SIP Entity to the CS1000 by navigating to Routing/SIP Entities. 16. Click SIP Entities and then click New for the CS1000 and then have the settings similar to CS1K as seen below. Name: CS1K IP @: 135.9.152.32 SIP Timer: 4 sec 29

17. Click on Commit 18. Then go to Routing/Entity Links 19. Click NEW to create a new entity link and then have settings similar to SM to CS1K example below. Enter the following parameters for the entity to CS1000 PBX GW: Name: SM to CS1k SIP Entity 1: mefp52-sm SIP Entity 2: CS1K Protocol: TCP Port: 5060 30

20. Click Commit to save the configuration. 21. Next goto Routing/Routing Policies to create a new policy to the CS1000. 22. Click New and enter settings as seen in the CS1K to SM example below. Name: CS1K to SM Retries: 0 23. Then select the SIP Entity as Destination previously created for the link to the CS1000. In this example, CS1K. 31

24. Then click on Commit to save changes. 25. Next, go to Routing/Dial Patterns and then click on New and have settings similar to number 4 that is highlighted above. This rule will allow 4xxx numbers to be passed to the CS1000. Pattern: 4 Min: 4 (4 digits) Max: 4 (4 digits) SIP Domain: avayalabx.com 32

26. Enter the Originating Location and Routing Policy to this Dial Pattern, Routing Policy Name: CS1K to SM Routing Policy Destination: CS1K 27. Click on Commit to save. 33

7.0 Motorola WSM Server Setup Procedure 7.1 WSM Configuration to Session Manager 1. Login to the WSM using the nsmconf user login. 2. Once Logged in, goto Configuration Management/WSM Configuration to enter the appropriate parameters to connect to the Avaya Session Manager. SIP address of record domain name: Voice Server Securiy: avayalabx.com none/tcp 34

3. Scroll Down and enter the following additional parameters in the WSM Configuration: Dispatch Service: enable Group Call Service: enable Text Messaging: enable 4. Navigate to the External Services Tab and enter the following parameters: V oice Server host name: 135.9.169.22 (IP @ of Session Manager) Voice Server Port: 5060 Primary AAA: 10.126.132.131 (Internal WSM) Primary AAA Port: 1812 35

5. Navigate to the WLAN Handsets Tab and enter the following parameters: PBX Type: Nortel CS1000 Time Zone: local time VoiceMail Phone #: 5000 (VM pilot) 6. Select Validate to check configuration. 36

7. Push the configuration if validation was successful. 8. Make sure Users are created for handsets to register appropriately to the WSM. Use the nsmserv userid login to manage users as needed in the design of the phone system. 37

7.2 WSM - Wireless Subscriber Configuration 1. Login to the WSM using the nsmserv UserId to provision the wireless handsets. 2. Navigate to the Provisioning/Add Subscriber link on the left hand side of the menu screen. 3. Enter the following parameters to create a subscriber in WSM: UserID: Enterprise Phone: Associate Name: Text Messaging: 300x (SIP User ID) 300x (callable number) name of user enable or disable 4. Next click on the Dispatch Tab to select the appropriate PTT talk groups. 5. Select the Talk Groups the subscriber will be associated with. 6. Set Dispatch Service (one to one PTT) to Active. 7. Set Group Call Service (group PTT) to Active. 8. Set Emergency Call Service (#255 TG - All Call PTT) to Active. 9. Click on submit to save the changes to the wsm. 38

10. Next Select the Provisioning/Change Subscriber Password to set the password for the new user/subscriber being created. 11. Enter the password desired for the subscriber. In this case a single user is changed. This must be done to properly setup subscriber. 7.3 Troubleshooting Logs on WSM Server Enabling and Retrieving SIP Proxy Logs - SIP Proxy 1. Open the WSM Web Page and login with username: dart and password: EnG1ne3R. Click on WSM Log Management 39

2. Then Click On Set and View Subcomponent Debug Level 3. Turn on the Debug level for the SIP proxy (as shown) and click on the Set Debug Level button. 40

4. The log status should show with an *. Click on Alarms, Traces and Statistics Maintenance. 5. Click on View Debug App Logs 41

6. Choose proxy from the WSA Application drop down list and click on the View Debug Log button. 7. The log should be displayed in real time window view. Refresh as needed for log updates during testing of calls. 8. The log can be copied and pasted into a text file or you can save it as a.txt file from the browser. 9. Next Click on. Alarms, Traces and Statistics Maintenance. 10. Click on Set and View Subcomponent Debug Level 11. Turn off the Debug level. *It is important to turn off the debug levels when not capturing traces. 42

End of Procedure 43