<div>Sorry I found this message:</div>
<div><a href="http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg10032.html">http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg10032.html</a></div>
<div>and the</div>
<div><param name="inbound-codec-negotiation" value="scrooge"/></div>
<div>fixed my problem.</div>
<div> </div>
<div>Now I have another question. </div>
<div>What exactly this "scrooge" does ? Is this going to affect the other providers that was working ?</div>
<div>Thanks</div>
<div>Leonardo</div>
<div><br><br> </div>
<div class="gmail_quote">On Tue, Apr 27, 2010 at 11:27 PM, leonardo alves <span dir="ltr"><<a href="mailto:telteclistas@gmail.com">telteclistas@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div>Hello,</div>
<div> </div>
<div>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.</div>
<div>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.</div>
<div> </div>
<div>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<br>This issue has so far been identified to happen on the following broken platforms/devices:<br>Linksys/Sipura aka Cisco<br>
ShoreTel<br>Sonus/L3<br>We will try to fix it but some of the devices on this list are so broken who knows what will happen.</div>
<div> </div>
<div>Does anyone knows if there is a way to fix this issue ?</div>
<div> </div>
<div>Thanks</div>
<div>Leonardo</div></blockquote></div><br>