On Mon, Nov 10, 2008 at 8:13 AM, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Hi, > > I want to be able to bridge two sip channels using direct RTP > between my endpoints (Audio IP : not local) but without > using reinvites. So I set up my asterisk sip endpoints as follows: > > [test1] > type=friend > host=dynamic > username=test1 > dtmfmode=info > context=test_rtp > allow=all > canreinvite=no > directrtpsetup=yes > > [test2] > type=friend > host=dynamic > username=test2 > dtmfmode=info > context=test_rtp > allow=all > canreinvite=no > directrtpsetup=yes > > ... but it doesn't work. How can I ensure that the RTP is not going > through my asterisk box and that the re-invite method is not used? > > P.S. Both endpoints are using the same codec, so no codec translation > takes place. >
What version of Asterisk is this? Last I heard (from Olle) this option was very experimental and should not be used on production systems. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
