[Freeswitch-users] Asynchronous PTIME

[Saeed Ahmed]

on which profile did you put that param?

I've allnet phone if i use g729 with ptime:40 then i can't hear called party
properly, its choppy voice, i put autofix param in both internal and
external profiles, but it didn't help.

if i use ptime: 20 then it works fine.

On Sat, Dec 25, 2010 at 12:02 AM, Jan Riedinger <riedinger at sns.eu> wrote:

> Hi Serge,
>
> I had similar problems, look for my thread "Problematic Behaviour of FS
> regarding ptime negotiation" in October. I could solve the problem by
> setting "rtp-autofix-timing=false", which disabled the (too) smart
> behaviour of FreeSwitch.
>
> BR
>     Jan
>
>
> Am 22.12.2010 23:09, schrieb Serge S. Yuriev:
> > Hello,
> >
> > 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?
>
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