Try calling the public conference at sip:888@conference.freeswitch.org<sip%3A888@conference.freeswitch.org> that is a properly setup box and can accept any registrations. The sip and the conference all work fine so it's for sure a problem on your end. On Fri, Apr 2, 2010 at 2:04 PM, Clint Popetz <clint@42lines.net> wrote:
On Fri, Apr 2, 2010 at 1:25 PM, Michael Collins <msc@freeswitch.org>wrote:
On Fri, Apr 2, 2010 at 9:38 AM, Clint Popetz <clint@42lines.net> wrote:
Hi,
I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with mod_skypopen, and when I call the echo test with skype it is _beautiful_ and has no lag, and the same is true for my coworker, but when we both dial a mod_conference bridge with skype, we get a 10-12 second lag. CPU usage on the machine is nil. Any ideas?
Do you mean that when you speak, it takes 10-12 seconds before the audio is heard by someone else in the conference?
Correct.
Also, can you try the same exercise with a soft phone like x-lite? I'm curious to know if this happens only on Skype calls or on any calls made to a conference.
Sip is a whole other can of worms that's not working either. When connecting with Bria on the mac to the ivr_demo on 5000, I can hear the demo fine, but it won't listen to DTMF numbers to change the menu. When running xopier on linux, I can't hear anything and DTMF doesn't get through. Both those softphones work fine with asterisk, including dtmf. I was punting on getting sip working correctly in freeswitch until I determined whether skypopen solved the conferencing woes I'm facing with MeetMe/asterisk.
(It's entirely possible/probably I just have sip misconfigured...I'm just using the default configuration in that regard.)
-Clint
-MC
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-- Clint Popetz http://42lines.net Scalable Web Application Development
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