Hi list, I hope somebody already had this kind of problem:
I want to dial in from a SIP provider and then (in the incoming section for the provider) do a SIP Dial() out via the same provider. The dialled out phone number rings and the calls get connected but I can't hear any voice. If I do a monitor() I don't see the wav file growing, so I guess there is no RTP stream. Also a "rtp debug" does not show any data. Can I do something to test further, or, can anybody point me to the SIP messages which are important for debugging this? I had a look at them but with my limited knowledge I can't see where the problem is. I tested Asterisk 1.2.5 and current SVN 1.2. Thanks in advance Regards Christian Peter _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
