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john boxold
Beginner

Cisco CUCM and Nortel CS1000 Session Manager

I am working on a site with a Nortel Cs1000 v7.6 w/Session Manager. I have routing via a PRI but moving over to SIP.  I have gotten the two servers linked, using the Avaya Configuring SIP trunks between Avaya Aura® Session Manager Release 6.2, Avaya Aura® Communication Manager Release 5.2.1 and Cisco Unified Communications Manager Release 8.6.2.

 

I am not able to route calls out to the PSTN from the Cisco to Nortel. I can call between the 2 using internal DN's.

the SIP Trunk is established and in the RTMT I see the call route to the cs1000 and timeout.

 

In the Session manager I have the adaptation, entity, entity links, and routing.

 

Has anyone had any success in routing to the pstn from the CUCM -sip-CS100-sip-PSTN?

1 ACCEPTED SOLUTION

Accepted Solutions
Chris Deren
Hall of Fame Master

You need to ensure that the Avaya and Nortel side can allows dialing PSTN numbers, so whatever digits you sent them are accepted and proper PSTN trunk group can be matched.  This is permissions/dial plan issue on the Avaya/Nortel side and you need to work with whomever programs that side to resolve it.

View solution in original post

2 REPLIES 2
Chris Deren
Hall of Fame Master

You need to ensure that the Avaya and Nortel side can allows dialing PSTN numbers, so whatever digits you sent them are accepted and proper PSTN trunk group can be matched.  This is permissions/dial plan issue on the Avaya/Nortel side and you need to work with whomever programs that side to resolve it.

View solution in original post

Thanks, Chris.

I took your advice and drew everything out on paper.

Even though the UCM side showed the SIP Trunk registered. The Avaya showed the SIP entities were down as 500 service unavailable.In the session manager the Proxy server i was using first was the DR location. I swapped those and reset the UCM SIP trunk. I am able to validate the SIP Entities are registered (again, but correctly). I validated the patterns via RLB and RLI, checked the DMI tables.  Once that was completed i was able to strip the 9 send 1+ 10 for LD, 7 for internal and 10 for Local. I ran into a problem with the calls having intermittent failures with a SIP 408 timeout to the PSTN. I adjusted the SIP Normalization script and have not had a failure since. Most of the guides assume that you are doing the integration from scratch and have a lot more configuration than what is required.

 

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