[Freeswitch-users] FS "Holds On" to ITSP's 183 for 2 Sec.Before Relaying to Phone?

[Anthony Minessale]

type the following at the cli

console loglevel debug
sofia profile internal siptrace on

repeat and post the trace to http://pastebin.freeswitch.org (pass is in the
dialog box)


On Fri, Jun 4, 2010 at 11:12 AM, Alan Frisch <frisch.alan at gmail.com> wrote:

> Hi,
>
> Just tinkering around with FreeSWITCH pretty much using the default
> configuration, with some customization to connect my Polycom 430 and
> my ITSP.  The problem I am having is that the first few seconds of
> calls (or early media if available) have no audio.
>
> When I NGREP'd the SIP dialog between my Phone <> FS and FS <> ITSP
> with timestamps, I can see that FreeSwitch is holding onto the 183 and
> subsequent 200 OK for about 2 seconds before relaying it onto the
> Polycom.  FWIW, server is an Intel Atom 330.
>
> Any idea why this is... I am sure it must be a configuration issue
> somewhere?  I did not have this issue with Asterisk, so I am hoping to
> troubleshoot it and move on with FreeSWITCH
>
> Thanks.
>
> AF.
>
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-- 
Anthony Minessale II

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