[Freeswitch-users] Asynchronous PTIME

[Serge S. Yuriev]

Hello,

On Thu, 23 Dec 2010 01:09:55 +0300 "Serge S. Yuriev" <me at nevian.org>
wrote:

Any one?
Captured console log here
http://pastebin.freeswitch.org/14871

I wonna clear on side of this problem - should I beat provider or have
beaten myself..

> 2010-12-20 16:51:13.960502 [WARNING] mod_sofia.c:1036 Asynchronous
> PTIME not supported, changing our end from 20 to 60
> 
> I'm getting this warning and client hears chopped sound :(
> That is "Our end"?
> 
> Call flow is Cisco (sip) FS (sip) YATE (h323) MeraTransitSoftSwitch
> All but MVTS under my control.
> 
> I doesn't see any clue in logs and can't reproduce this with my
> testing via Blink/PortSIP/PhonerLite/Twinkle or Grandstream devices
> 
> Which debug/logs I should take? Any ideas?
> 
> Thanks a lot.
> 
> btw how I can save debug into log not only console?

-- 
Serge S. Yuriev
Lead VoIP engineer