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

[Brian West]

Alan,
	Welcome to FreeSWITCH, Lets start out with some questions... Is FreeSWITCH behind nat?

Can you type "sofia status profile internal" and reply with that?

/b

On Jun 4, 2010, at 11:12 AM, Alan Frisch 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.