[asterisk-users] Help T.38
Kevin P. Fleming
kpfleming at digium.com
Sun Mar 1 12:21:20 CST 2009
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: kpfleming at digium.com
Check us out at www.digium.com & www.asterisk.org
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