[Asterisk-Users] Cisco AS5300 --> [SIP] --> Asterisk - NO AUDIO

[Rich Adamson]

> OK, here goes my next problem.
> 
> I have puchased a DID which I can connect to via SIP
> 
> I have been given the following details:
> 
> Username: uka1xxxxxx
> Password: 1000xxxxxx
> 
> Server: brxxxx.net:5160
> 
> My equipment is Asterisk CVS HEAD on Red Hat EL 3.0 (NO NAT)
> 
> The other end is a Cisco AS5300 (NO NAT)
> 
> I can register with the Cisco with no problem.
> 
> When I dial the DID it sends the call to my asterisk server and my
> asterisk server sends back the dial tone, no problem.
> 
> The problem is when I pick up the phone, no audio.

Try inserting canreinvite=no in the sip.conf definition for the phone
and restart asterisk.

The trace suggests that your provider and the phone were told to
establish a sessions between themselves, and that is not happening
correctly.

There is nothing in that trace that would suggest a codec problem,
so I'm not sure how you jumped to that conclusion. In fact, the trace
tells you there are several compatible codecs available between asterisk
and your provider, and it chose g729 successfully.

If that doesn't help, then copy/paste the important sections of sip.conf
and extensions.conf that would reflect the handling of a call, and
post that to this list.