<p dir="ltr">Hmm, quite interesting. I think we can develop a testing tool based on this, and make it royalty free. And make it suitable for transcoding and analog paths.</p>
<div class="gmail_quote">On May 19, 2015 12:36 PM, "Владимир Федоров" <<a href="mailto:lexxua@gmail.com">lexxua@gmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">In my company we use Yate <span style="font-size:13.1999998092651px;line-height:19.7999992370605px">analyzer module</span><br>as a call quality monitor <a href="http://docs.yate.ro/wiki/Quality_analyzer_module" target="_blank">http://docs.yate.ro/wiki/Quality_analyzer_module</a> . Results of tests stored in Influxdb.<br><div>For gateway monitoring we use built in Freeswitch logging of DOWN state of gateway.</div><div><br></div><div> </div></div><br><div class="gmail_quote">On Tue, May 19, 2015 at 12:04 PM Stanislav Sinyagin <<a href="mailto:ssinyagin@gmail.com" target="_blank">ssinyagin@gmail.com</a>> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I'm working on two different projects for SIP testing and monitoring,<br>
and soon there will be more information.<br>
<br>
<a href="https://github.com/voxserv/rring" target="_blank">https://github.com/voxserv/rring</a><br>
(documentation is still missing) this will be an automated tester that<br>
uses FreeSWITCH to originate and terminate the calls, and it will<br>
analyze the SIP messages that are received from remote side.<br>
<br>
<br>
<a href="https://txlab.wordpress.com/2015/05/14/quality-assurance-for-voip-calls/" target="_blank">https://txlab.wordpress.com/2015/05/14/quality-assurance-for-voip-calls/</a><br>
some scripts and work in progress for voice quality assurance.<br>
<br>
I will make a separate posting as soon as I'm ready.<br>
<br>
<br>
<br>
<br>
On Tue, May 19, 2015 at 3:27 AM, Jai Rangi <<a href="mailto:jprangi@didforsale.com" target="_blank">jprangi@didforsale.com</a>> wrote:<br>
> Very common concerns from new Asterisk, Freeswitch, opensips and freepbx<br>
> owners, How can we monitor system, what happens if service stops responding.<br>
><br>
> Here is a small howto on monitoring any SIP service with nagios. I am sure<br>
> there are plenty of options and suggestions, but this is one of them and has<br>
> been working out very well for us for years.<br>
><br>
> <a href="http://www.didforsale.com/monitor-sip-server" target="_blank">http://www.didforsale.com/monitor-sip-server</a><br>
><br>
> Best,<br>
> -Jai<br>
><br>
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Official FreeSWITCH Sites<br>
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</blockquote></div>
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