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 _______________________________________________ --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
