[asterisk-dev] Help T.38

[michel freiha]

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
under General context...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-dev/attachments/20090301/f6f9893a/attachment.htm