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.

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?