SIP

[85adventures]

CM 6.0.1 Evolution Server, SMGR 6.3.12, SM 6.3.12

I have a SIP trunk between CM and SM, which seems to be configured properly.

I have three SIP phones (9620C, 9611G, 96xx Emulator), registering properly with SM, but aren't connecting to CM. Each has an icon in the upper left of their LCD display of a triangle with an exclamation point within it, indicating they are in failover mode.

When I issue a "status station" in CM, each shows
Service State: out-of-service
TCP Signal Status: not connected

Each SIP station is able to call the other two SIP stations, a digital set is able to call the SIP stations. I cannot call SIP to digital though. This is a sample tac trace when I dial from SIP extension 50203 to digital extension 51234:

LIST TRACE

time data

11:43:37 TRACE STARTED 04/07/2015 CM Release String cold-00.1.510.1-21061
11:43:47 SIP<INVITE sip:51234@domain.com SIP/2.0
11:43:47 Call-ID: 67_5523d1023cc4afff3j246r2l16e3i6v3t1k2x6t1a68
11:43:47 28o6l5h_I5020310.1.115.165
11:43:47 active trunk-group 71 member 1 cid 0x2484
11:43:47 denial event 1817: No rte for unauth domain D1=0x830047 D2=0x141c00
19
11:43:47 SIP>SIP/2.0 500 Server Internal Error (Denial 1817)
11:43:47 Call-ID: 67_5523d1023cc4afff3j246r2l16e3i6v3t1k2x6t1a68
11:43:47 28o6l5h_I5020310.1.115.165
11:43:47 term trunk-group 71 cid 0x2484
11:43:47 idle trunk-group 71 cid 0x2484

If anyone has any suggests as to what the problem might be, I'd appreciate hearing. I've looked through the configs and manuals, but my eyes are beginning to cross and am not sure what to try next. Thanks.

 

[kyle555]

The triangle is indicating "limited service" - not failover mode.

You should review setting up basic SIP telephony with Avaya CM and SM if you've not done it before.

A call from SIP phone A to SIP phone B should basically look like:

-SIP phone A goes off hook
-SM sees this, invokes the application sequence of phone A to CM
-CM sees the phone go off hook
-Phone dials digits, CM processes them
-CM sees the destination station is a SIP telephone with off-pbx station mapping, and that it isn't forwarded to voicemail and sends the call to AAR and a trunk back to SM
-CM sends the call to SM on a signaling group, with matching domains as those used in Session Manager
-SM rings phone B


You need to make sure everything is set up in CM for things like off-pbx station mapping, your private and public unknown numbering tables, AAR, route patterns, etc

You need to make sure SM is set up with its domains and SIP entities properly as well. At a basic level, match up the domain the phones use, the domain SM uses and the domain on the CM signaling group for SM as well as the domain of the network regions associated with CM's processor ethernet and the far end network region of your signaling group to SM.

 

[gwebster]

Two good reference documents:
- Implementing End-to-End SIP Vol 1: Endpoint Deployment (

https://downloads.avaya.com/css/P8/documents/100154012)

- Implementing End-to-End SIP Vol 2: SIP Telephone Signaling and Dial Plan Options (

https://downloads.avaya.com/css/P8/documents/100182873)