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You have to set it the paramater in the profile, which is used for
the call, e. g. set in external.xml:<br>
<param name="rtp-autofix-timing" value="false"/><br>
<br>
BR<br>
Jan<br>
<br>
<br>
<br>
Am 29.12.2010 20:04, schrieb Saeed Ahmed:
<blockquote
cite="mid:AANLkTimzmg1kfM4+75e-Y9vG6wErhXjX6f83+4-voNyu@mail.gmail.com"
type="cite">on which profile did you put that param?<br>
<br>
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.
<div>
<br>
</div>
<div>if i use ptime: 20 then it works fine.<br>
<br>
<div class="gmail_quote">On Sat, Dec 25, 2010 at 12:02 AM, Jan
Riedinger <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:riedinger@sns.eu">riedinger@sns.eu</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">Hi Serge,<br>
<br>
I had similar problems, look for my thread "Problematic
Behaviour of FS<br>
regarding ptime negotiation" in October. I could solve the
problem by<br>
setting "rtp-autofix-timing=false", which disabled the (too)
smart<br>
behaviour of FreeSwitch.<br>
<br>
BR<br>
Jan<br>
<br>
<br>
Am 22.12.2010 23:09, schrieb Serge S. Yuriev:<br>
<div class="im">> Hello,<br>
><br>
> 2010-12-20 16:51:13.960502 [WARNING] mod_sofia.c:1036
Asynchronous PTIME<br>
> not supported, changing our end from 20 to 60<br>
><br>
> I'm getting this warning and client hears chopped
sound :(<br>
> That is "Our end"?<br>
><br>
> Call flow is Cisco (sip) FS (sip) YATE (h323)
MeraTransitSoftSwitch<br>
> All but MVTS under my control.<br>
><br>
> I doesn't see any clue in logs and can't reproduce
this with my testing<br>
> via Blink/PortSIP/PhonerLite/Twinkle or Grandstream
devices<br>
><br>
> Which debug/logs I should take? Any ideas?<br>
><br>
> Thanks a lot.<br>
><br>
> btw how I can save debug into log not only console?<br>
<br>
</div>
<div>
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