I looked at that, but I think that will cause issues with the NAT traversal. Our phones will all be in external networks. I forgot to mention that. -----Original Message----- From: Anthony Minessale [mailto:anthony.minessale@gmail.com] Sent: Tuesday, February 17, 2009 05:34 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Sending media streams to a media gateway 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 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm mailto:MSN%3Aanthony_minessale@hotmail.com GTALK/JABBER/mailto:PAYPAL%3Aanthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference mailto:sip%3A888@conference.freeswitch.org http://iax:guest@conference.freeswitch.org/888 mailto:googletalk%3Aconf%2B888@conference.freeswitch.org pstn:213-799-1400