Brian/Anthony, Looks like I forgot to remove the <-- and --> from around the custom RTP settings. This may have been causing the hang and calls are now getting full early media, without cut-offs. Thanks and have a great weekend! AF. On Fri, Jun 4, 2010 at 12:56 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
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@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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