[asterisk-users] Asterisk Playback sound dropping on linphone

[Matteo Fortini]

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
>
>