[asterisk-dev] Help T.38

[Kaloyan Kovachev]

On Sun, 1 Mar 2009 16:53:53 -0500, Venefax wrote
> 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?
> 

It is possible to enable udptl in general, but disable it for the OpenSIPS
account ... not sure, but it may work if in the dialplan at fax extension you
forward the call to another "OpenSIPS-T38" account with udptl enabled


> 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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