Hi, AFAIK, there is no such 'new' behavior of the conference bridge. And I just recalled I read this somewhere that (by default) Asterisk's clock for sending RTP is triggered by receiving RTP. So if it doesn't receive any RTP packets, it won't send any RTP packets too. This might be the case. Regards, nanang On Mon, Oct 6, 2008 at 5:17 AM, Hubert Langevin <hubertlangevin at gmail.com> wrote: > Hi everyone, > > I just realised a "new" behavior from pjsip that i haven't seen > before. I make a call from my user agent to a soft PBX, asterisk in > this case, in order to get music on hold. When i make the call, the > conference port for the new call gets created and i hear music on my > headset. Eveything looks fine. My sound card sits on port 0 and the > new call i made sits on port 1 and they are both transmitting to each > other. However if i do "cd 0 1", that is stop my sound card to > transmit to port 1, i can't hear the music anymore, even though i > still have Port 1, the new conference port playing music on hold, > transmitting to port 0. > > Any ideas? > > Hubert > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >
