[asterisk-users] incoming call problem

[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...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
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