Hi, On Thu, 2011-07-21 at 15:23 +0800, MingHon wrote: > Hello List, > > > im still trying but no luck. > asterisk canreinvite already set to yes >
What version of asterisk? I think in 1.6.2 canreinvite was replaced with directmedia and directrtp. > > now im testing in lan > i setup kamailio and asterisk in same lan > kamailio&rtpproxy on 192.168.2.3 and asterisk on 192.168.2.23 > > > canreinvite=yes in asterisk. when both ua in the same lan > register directly to asterisk the reinvite work. both ua will have > and direct media flow > > > [ua1]<====>[ua2] > | > | > x > | > v > [asterisk] > > > when ua register to kamailio the audio work and the reinvite message > is same as the first invite message. > > > [ua1]<====>[kamailio]<====>[ua2] > | ^ > | | > | | > v | > [asterisk] > > > how do i stop the media flow between kamailio and asterisk? > make kamailio relay the rtp between both ua. > > > [ua1]<====>[kamailio]<====>[ua2] > | ^ > x x > | | > v | > [asterisk] > > > > > anyone could give some hint? > > > thanks in adv. > > > -- > Regards, > > MingHon S. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
