you could set the variable bypass_media to true before you call bridge<br><br>&lt;action application=&quot;set&quot; data=&quot;bypass_media=true/&gt;<br>&lt;action application=&quot;bridge&quot; data=&quot;sofia/internal/someuser@somehost&quot;/&gt;<br>
<br>that will negotiate a point to point media connection between the caller and callee<br><br><br><div class="gmail_quote">On Tue, Feb 17, 2009 at 5:51 PM,  <span dir="ltr">&lt;<a href="mailto:fax@virgintechnologies.com">fax@virgintechnologies.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;"><div bgcolor="#ffffff"><div><font face="Verdana" size="2">I have Freeswitch running successfully with a fairly basic config.&nbsp; Nat traversal is working well on both the client and server side.&nbsp; I want to start running all RTP streams through a media gateway, and use Freeswitch for SIP registrations and signalling only.</font></div>

<div><font size="2">I believe that I need to have Freeswitch invite the SIP phone to send the RTP stream directly to the media gateway when a call starts.&nbsp; Where can I start with this?&nbsp; Does anyone have any example configs?</font></div>

<div><font size="2">Justin</font></div></div>
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