paste your sip.conf. David 2009/2/26 michel freiha <mich...@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.
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