Sorry I found this message: http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg10032.h... 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@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