<html><body bgcolor="#ffffff"><DIV><FONT face=Verdana size=2>I have Freeswitch running successfully with a fairly basic config. Nat traversal is working well on both the client and server side. 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. Where can I start with this? Does anyone have any example configs?</FONT></DIV>
<DIV><FONT size=2>Justin</FONT></DIV></body></html>