You have to set it the paramater in the profile, which is used for the call, e. g. set in external.xml: <param name="rtp-autofix-timing" value="false"/> BR Jan Am 29.12.2010 20:04, schrieb 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@sns.eu <mailto: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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