<div dir="ltr">Dear Keven,<br>I have just post a new email with the same body due to a member advice<br><br>Regards<br><br><div class="gmail_quote">On Sun, Mar 1, 2009 at 8:21 PM, Kevin P. Fleming <span dir="ltr"><<a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div></div><div class="Wj3C7c">michel freiha wrote:<br>
> Dear All,<br>
> I have created an inbound context in sip.conf that forward incoming call<br>
> to opensips server...The problem appears as soon as I enable t38pt_udptl<br>
> = yes under General context...The Asterisk negotiate the SIP session<br>
> with OpenSIPS without adding voice codec to INVITE packet...It just<br>
> contains T.38 protocol...When t38pt_udptl is disabled everything looks<br>
> OK and Ulaw is negotiated with OpenSIPS and cal success..Any suggestion<br>
> here?<br>
<br>
</div></div>Please stop cross-posting your messages, and especially please stop<br>
posting messages to the asterisk-dev list that don't belong there. Thanks.<br>
<br>
--<br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
skype: kpfleming | jabber: <a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a><br>
Check us out at <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
<br>
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