[asterisk-users] incoming call problem

[David fire]

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
>
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