I seem to recall having to generate silence in such cases. I wanted to implement mute, but ran into this problem. So I ended up just connecting it to a port of my own design that sent a stream of low- level random noise (as total silence sounded to the other end like the connection was terminated.) Norman Franke Answering Service for Directors, Inc. www.myasd.com On Oct 6, 2008, at 3:56 PM, Nanang Izzuddin wrote: > 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 >> > > _______________________________________________ > 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
