Dear All, I have created an inbound context in SIP .conf that forward incoming call to opensips server...The problem appears as soon as I enable t38pt_udptl yes...The Asterisk negotiate the SIP session with OpenSIPS without adding voice codec to INVITE packet...It just contains T.38 protocol...When t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with OpenSIPS and cal success..Any suggestion here? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090226/26299f7b/attachment.htm
paste your sip.conf. David 2009/2/26 michel freiha <michofr at gmail.com>> Dear All, > I have created an inbound context in SIP .conf that forward incoming call > to opensips server...The problem appears as soon as I enable t38pt_udptl > yes...The Asterisk negotiate the SIP session with OpenSIPS without adding > voice codec to INVITE packet...It just contains T.38 protocol...When > t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with > OpenSIPS and cal success..Any suggestion here? > > Thanks > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090227/86f07deb/attachment.htm