Hi On 11/11/2010 03:35 PM, Matteo Fortini wrote: > 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!)...
I use linphonec as well - and haven't found another console sip phone either. I'd be interested if there is another one. Sebastian > > 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
