Try calling the public conference at <a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br>that is a properly setup box and can accept any registrations.<br><br>The sip and the conference all work fine so it's for sure a problem on your end.<br>
<br><br><br><br><div class="gmail_quote">On Fri, Apr 2, 2010 at 2:04 PM, Clint Popetz <span dir="ltr"><<a href="mailto:clint@42lines.net">clint@42lines.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<br><br><div class="gmail_quote"><div class="im">On Fri, Apr 2, 2010 at 1:25 PM, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<br><br><div class="gmail_quote"><div>On Fri, Apr 2, 2010 at 9:38 AM, Clint Popetz <span dir="ltr"><<a href="mailto:clint@42lines.net" target="_blank">clint@42lines.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hi,<div><br></div><div>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?</div>
<div></div></blockquote></div><div><br>Do you mean that when you speak, it takes 10-12 seconds before the audio is heard by someone else in the conference? </div></div></blockquote><div><br></div></div><div>Correct.</div>
<div class="im"><div>
</div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"><div class="gmail_quote"><div>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.<br>
</div></div></blockquote><div><br></div></div><div>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.</div>
<div><br></div><div>(It's entirely possible/probably I just have sip misconfigured...I'm just using the default configuration in that regard.)</div><div><br></div><div>-Clint</div><div><br></div><div><br></div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div class="gmail_quote"><div>
-MC<br></div></div><br>
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