asterisk users - Oct 2006 - Problems with Dial In - Dial Out via SIP - no voice

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

Sorry to reply to myself,
if I dial out with ISDN it works. I don't have a different SIP account
to test dial in SIP_PROVIDER_1 and dial out with SIP_PROVIDER_2.


Am Donnerstag, den 05.10.2006, 11:14 +0200 schrieb 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
> 
> _______________________________________________
> --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