<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Check out <a href="http://wiki.opencsta.org/index.php/SIPp">http://wiki.opencsta.org/index.php/SIPp</a> the registration scenario, or just set up IP based ACL without auth to send you a call every 10 seconds or so in a queue. (or set up another box and some sort of call generation mechanism) - I'm currently using SIPp to register and bailing out of executing after the first registration, then listening for any media and echoing it back for queues. They are very obedient agents :) I'm sure something similar could be done for the conference, just bridge one conference channel to a file - the SIPp buggers will echo it back!<div><div><br><div>You'd be best putting some monitoring software on it too (<a href="http://zabbix.com">zabbix.com</a> is pretty good - might be a bit of a learning curve, but if you wanted to put the agent/proxy on there and route it back to my install i can give you a login - it's phoning home back to the zabbix server is over ssl tunnels)</div><div><br></div><div>Maybe set debug on as well, hopefully that tells you a bit more.</div><div><div><br></div><div><br><div><div>On 17/05/2012, at 1:58 AM, Andrew Cassidy wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">I'm on the 512MB RAM tier. Occasionally FreeSWITCH craps itself and crashes leaving no logs but that's like every few weeks. I currently have my work calls routed through it, and take a fair few of them (one at a time mind, but I have call waiting set up on my phone so can have 2 sip sessions at once. Voicemail seems fine, and I have a conference call set up but it needs further testing.<div>
<br></div><div><font face="arial, helvetica, sans-serif">tel:+44<span style="color:rgb(34,34,34);text-align:center;background-color:rgb(252,253,253)">3300100002,</span> sip:<span style="color:rgb(34,34,34);text-align:center;background-color:rgb(252,253,253)"><a href="http://03300100002@sip.v-pbx.net:5080/">03300100002@sip.v-pbx.net:5080</a> or ISN 1234*1432 will get you into the conference which is probably the best test of the media capabilities.</span></font></div>
<div><font face="arial, helvetica, sans-serif"><span style="color:rgb(34,34,34);text-align:center;background-color:rgb(252,253,253)"><br></span></font></div><div><font face="arial, helvetica, sans-serif"><span style="color:rgb(34,34,34);text-align:center;background-color:rgb(252,253,253)">I've not done a proper load test on it either, I've mainly been using it to build a front end.</span></font></div>
<div><font face="arial, helvetica, sans-serif"><span style="color:rgb(34,34,34);text-align:center;background-color:rgb(252,253,253)"><br></span></font></div><div><font face="arial, helvetica, sans-serif"><span style="color:rgb(34,34,34);text-align:center;background-color:rgb(252,253,253)">I've been told by customers that my voice breaks up a little, but since setting vad=none I've seen better results. Incoming audio seems to have no issues, mind so it could just be my home internet connection which today was upgraded to 30Mbps down 2Mbps up so I may see an improvement yet.</span></font></div>
<div><font face="arial, helvetica, sans-serif"><span style="color:rgb(34,34,34);text-align:center;background-color:rgb(252,253,253)"><br></span></font></div><div><font face="arial, helvetica, sans-serif"><span style="color:rgb(34,34,34);text-align:center;background-color:rgb(252,253,253)">I also installed voipmonitor yesterday, so will take a look at the logs later.</span></font></div>
<div><br><div class="gmail_quote">On 16 May 2012 06:53, Gabriel Gunderson <span dir="ltr"><<a href="mailto:gabe@gundy.org" target="_blank">gabe@gundy.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class="im">On Tue, May 15, 2012 at 10:33 PM, Thaddeus Hogan <<a href="mailto:thaddeus@thogan.com">thaddeus@thogan.com</a>> wrote:<br>
> I have Freeswitch running on a KVM based VM right now, and it doesn't do<br>
> too hot. Voicemail, MoH, and IVR prompts are choppy often but call media<br>
> proxy (no transcode) works okay, though long calls start to deteriorate<br>
> after about 30 minutes.<br>
<br>
</div>I've used KVM with good results... what kind of load are we talking about?<br>
<br>
Gabe<br>
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