Hello, On Thu, 23 Dec 2010 01:09:55 +0300 "Serge S. Yuriev" <me@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