[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]
[Linphone-developers] Re: [asterisk-users] Asterisk Playback sound dropp
[Linphone-developers] Re: [asterisk-users] Asterisk Playback sound dropping on linphone
Thu, 11 Nov 2010 18:40:55 +0100
Mozilla/5.0 (X11; U; Linux i686; en-US; rv:126.96.36.199) Gecko/20101030 Lightning/1.0b1 Icedove/3.0.10
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?
Il 11/11/2010 16:35, Matteo Fortini ha scritto:
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!)...