[asterisk-dev] Help T.38

[Venefax]

This issue is affecting me as well.
Federico

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Dwayne Hubbard
Sent: Sunday, March 01, 2009 4:35 PM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Help T.38


----- "michel freiha" <michofr at gmail.com> wrote:

> 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?


You are describing a known issue: http://bugs.digium.com/view.php?id=12437

There is a developer branch with the goal of solving this problem, but it
needs people to test it and report feedback.  The branch is located at:
http://svn.digium.com/svn/asterisk/team/file/issue12437


Dwayne Hubbard
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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