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.