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@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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