Running into an issue with Intergration

[dseevers]

I have been working towards integrating our on-prem 3CX system with our other on-prem PBX, which is an Alcatel system.

Most of this has been successful. Intra-system calling works via 5-digit dialing and Caller ID passes through. For reference, the integration is via regular SIP trunks, not a Bridge.

However, there is an issue with accessing the auto attendants that reside on the Alcatel system.

With the Alcatel system, there is a separate/independent VM that handles voicemail. This same system also handles Auto Attendants. So, when an extension/DID hits an Auto Attendant the call is redirected to the voicemail system. This of course works fine within the Alcatel system. However, there is an issue when trying to reach an Auto Attendant from SIP/3CX.

When dialing an auto attendant from 3CX, the Alcatel system is bouncing back a '302 Moved Temporarily' message, and, as best I can tell, correctly sends the updated contact to start a new call to the voicemail system. But, it seems that 3CX is making no attempt to contact the returned extension. All that happens is the call fails with:

Call(C:262): Call to <sip:[email protected]> has failed; Cause: 302 Moved Temporarily/INVITE from x.x.x.x:5060

Looking at a packet capture (see attached), I am seeing the following

1. 3CX extension 40010 dials Alcatel extension 10000
2. Alcatel responds with a 302 and instructs 3CX to instead dial 19999
3. 3CX fails the call with no apparent attempt to make the call to 19999 as instructed

 

[TheodorosG_3CX]

Please note that bridges with any 3rd party PBX is not supported.

Also note that the 302 in most of the cases indicates that there is a call forwarding.

 

[dstrait]

This integration attempt is via SIP tunnels, not a bridge. 3CX sees the Alcatel system as a Generic SIP provider. The Alcatel system is instructing 3CX to try another extension (call forwarding). Does 3CX not support a 302 response to try a different number?

 

[TheodorosG_3CX]

Even this setup is not something that we test or support.
Currently you are using a generic sip trunk to connect to another PBX and route calls.