[asterisk-users] Problems with Dial In - Dial Out via SIP - no voice

[Christian Peter]

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