Dear Keven, I have just post a new email with the same body due to a member advice
Regards On Sun, Mar 1, 2009 at 8:21 PM, Kevin P. Fleming <[email protected]>wrote: > 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 >
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