[Freeswitch-users] Error with media ptime

leonardo alves telteclistas at gmail.com
Tue Apr 27 21:19:16 PDT 2010


Sorry I found this message:
http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg10032.html
and the
<param name="inbound-codec-negotiation" value="scrooge"/>
fixed my problem.

Now I have another question.
What exactly this "scrooge" does ? Is this going to affect the other
providers that was working ?
Thanks
Leonardo



On Tue, Apr 27, 2010 at 11:27 PM, leonardo alves <telteclistas at gmail.com>wrote:

> Hello,
>
> I am new to freeswitch and I have just installed the last version of
> freeswitch. I am doing some tests dialing with sip and when the call is
> answer I play an file.
> If I do 10 calls, in 5 or 6 of them I get this error in the console and the
> audio gets cut like with it was a bandwithd problem.
>
> 2010-04-27 22:20:56.236782 [WARNING] mod_sofia.c:999 We were told to use
> ptime 20 but what they meant to say was 30
> This issue has so far been identified to happen on the following broken
> platforms/devices:
> Linksys/Sipura aka Cisco
> ShoreTel
> Sonus/L3
> We will try to fix it but some of the devices on this list are so broken
> who knows what will happen.
>
> Does anyone knows if there is a way to fix this issue ?
>
> Thanks
> Leonardo
>
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