On Fri, Jun 4, 2010 at 11:12 AM, Alan Frisch
<frisch.alan@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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