type the following at the cli <div><br></div><div>console loglevel debug</div><div>sofia profile internal siptrace on</div><div><br></div><div>repeat and post the trace to <a href="http://pastebin.freeswitch.org">http://pastebin.freeswitch.org</a> (pass is in the dialog box)</div>
<div><br><div><br><div class="gmail_quote">On Fri, Jun 4, 2010 at 11:12 AM, Alan Frisch <span dir="ltr"><<a href="mailto:frisch.alan@gmail.com">frisch.alan@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Hi,<br>
<br>
Just tinkering around with FreeSWITCH pretty much using the default<br>
configuration, with some customization to connect my Polycom 430 and<br>
my ITSP. The problem I am having is that the first few seconds of<br>
calls (or early media if available) have no audio.<br>
<br>
When I NGREP'd the SIP dialog between my Phone <> FS and FS <> ITSP<br>
with timestamps, I can see that FreeSwitch is holding onto the 183 and<br>
subsequent 200 OK for about 2 seconds before relaying it onto the<br>
Polycom. FWIW, server is an Intel Atom 330.<br>
<br>
Any idea why this is... I am sure it must be a configuration issue<br>
somewhere? I did not have this issue with Asterisk, so I am hoping to<br>
troubleshoot it and move on with FreeSWITCH<br>
<br>
Thanks.<br>
<br>
AF.<br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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