yes it will.
Sorry I found this message: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 ?ThanksLeonardo
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 ?ThanksLeonardo
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org