I did some more tests, and it's not really a problem with linphone: the rtp capture shows empty packets sent by Asterisk. Since the channel which is doing Playback() is in a MeetMe conference, I tried also to speak on another phone on the same conference: well the rtp capture shows the stream from A* becoming silent, then the new sound from the phone comes up.
Do I have to file a bug? Thank you, Matteo Il 11/11/2010 16:35, Matteo Fortini ha scritto: > Hi, > I dial on A* from a linphonec to a Playback() extension, then suddenly > the sound stops after a while, without any notice. > I enabled debug both in linphone and A*, and the RTP packets are sent > from A* and received from linphone. It doesn't matter whether I choose > alaw, ulaw, gsm as codec (besides changing cpu load of course). > > How can I debug it? I'm using A* 1.6.2 and both linphone 2.x and 3.x. > > I just need a console scriptable softphone, so maybe there's an > alternative to linphone (which seemed good enough anyway!)... > > Thank you, > Matteo > > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
