michel freiha wrote: > 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?
Please stop cross-posting your messages, and especially please stop posting messages to the asterisk-dev list that don't belong there. Thanks. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: [email protected] Check us out at www.digium.com & www.asterisk.org _______________________________________________ -- 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
