Hi, please assist with this problem adding Trixbox SIP phones to a CME setup.
We have had a Cisco UC560 device with 20 phones or so, running fine for a couple of years, connecting to the PSTN via a telco provided SIP trunk. A third party supplier has now provided a Trixbox and some SIP phones, with which we need to integrate with the current setup.
We got as far as adding dial-peers for the extensions that are on the Trixbox phones, and using a test set of dial-peers, managed to get internal Cisco to TB phone calls functioning perfectly. It is also possible dial in from the PSTN to the TB phone, and the call proceeds perfectly. However, outgoing calls from the TB phone to the PSTN do not function correctly. The call sets up fine, and remains active until one party ends it, but there is no audio in either direction.
Using "debug voip rtp" showed that indeed, the UC560 was receiving packets from both the Trixbox (192.168.92.9) and the SIP trunk provider (220.127.116.11x), but was not transmitting packets to either end point.
We are not sure why the incoming calls function correctly, but the same "debug voip rtp" trace reveals that as well as packets being sent to and from the end points, they are sent to and from the local IP source address for the CME (10.1.1.1) and the gateway address of the integrated service engine on 10.1.10.2.
I have attached the two rtp debugs from the successful incoming and failing outgoing traces. Will also post some running config excerpts up shortly.
All suggestions gratefully received!
The debug seems incomplete:
UC receives INVITE sip:firstname.lastname@example.org SIP/2.0
User-Agent: Asterisk PBX 18.104.22.168-FONCORE-r78
audio + video
but it doesn't send any sip messages to 192.168.92.9 or to SIP provider.
UC receives a TRYING, 183 session progess, 200 ok from SIP privider and send a finally ACK.
Can you add a complete debug with sent messages from UC?
Do you have already try to disable video capability in the TB?