<div dir="ltr">post a trace of FS after pressing f8 from the cli<br>detailing the entire call and we can have a look.<br><br><br><div class="gmail_quote">On Fri, Sep 19, 2008 at 6:29 PM, Matt Darnell <span dir="ltr"><<a href="mailto:mattdarnell@gmail.com">mattdarnell@gmail.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;">On Wed, Sep 17, 2008 at 10:14 AM, Matt Darnell <<a href="mailto:mattdarnell@gmail.com">mattdarnell@gmail.com</a>> wrote:<br>
> On Fri, Jul 25, 2008 at 8:51 PM, UV <<a href="mailto:uv@talknet.com.au">uv@talknet.com.au</a>> wrote:<br>
>> Yes I did, but you might not even need that.<br>
>> Try adding <param name="pass-rfc2833" value="true"/> in your external SIP<br>
>> profile and see if it solves the problem.<br>
>><br>
><br>
> I am still trying to get the DTMF 100%, I added the value but get this<br>
> message in the debug log:<br>
> [WARNING] mod_sofia.c:787 sofia_receive_message() Cannot pass 2833 on<br>
> a transcoded call<br>
><br>
> It does not appear that any transcoding is happening from the SIP<br>
> setup messages.<br>
<br>
Steve,<br>
<br>
Thanks for the response, here is the architecture:<br>
<br>
SIP Provider <-> SIP UDP <-> Internet <-> Freeswitch <-> LAN <-> SIP<br>
TCP <-> Exch 2007<br>
<br>
It appears to be G711 throughout the entire call, is it transcoded<br>
just because Freeswitch is bridging the call?<br>
<br>
-Matt<br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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