you could set the variable bypass_media to true before you call bridge <action application="set" data="bypass_media=true/> <action application="bridge" data="sofia/internal/someuser@somehost"/> that will negotiate a point to point media connection between the caller and callee On Tue, Feb 17, 2009 at 5:51 PM, <fax@virgintechnologies.com> wrote:

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? Justin

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