> 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. _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
