Try setting &quot;absolute_codec_string&quot; in the dialplan prior to the bridge:<br>&lt;action application=&quot;export&quot; data=&quot;nolocal:absolute_codec_string=G722&quot;/&gt;<br><br>Let us know if that does the trick.<br>
-MC<br><br><br><div class="gmail_quote">On Wed, Dec 16, 2009 at 11:13 AM, Kristian Kielhofner <span dir="ltr">&lt;<a href="mailto:kristian.kielhofner@gmail.com">kristian.kielhofner@gmail.com</a>&gt;</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;">
Hello everyone,<br>
<br>
  Pastebin here:<br>
<br>
<a href="http://pastebin.freeswitch.org/11525" target="_blank">http://pastebin.freeswitch.org/11525</a><br>
<br>
  I&#39;ve got my pjsip profile configured for G722 only:<br>
<br>
    &lt;param name=&quot;codec-prefs&quot; value=&quot;G722&quot;/&gt;<br>
<br>
  Yet whenever I send calls using that profile it (mysteriously)<br>
indicates support for PCMU in the INVITE.  The pastebin includes both<br>
the INVITE and &quot;sofia status profile pjsip&quot; to show that only G722 has<br>
been enabled.  Where is PCMU coming from?<br>
<br>
--<br>
Kristian Kielhofner<br>
<a href="http://www.astlinux.org" target="_blank">http://www.astlinux.org</a><br>
<a href="http://blog.krisk.org" target="_blank">http://blog.krisk.org</a><br>
<a href="http://www.star2star.com" target="_blank">http://www.star2star.com</a><br>
<a href="http://www.submityoursip.com" target="_blank">http://www.submityoursip.com</a><br>
<a href="http://www.voalte.com" target="_blank">http://www.voalte.com</a><br></blockquote></div><br>