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

[Alan Frisch]

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 at 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 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.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>