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. On May 19, 2015 12:36 PM, "Владимир Федоров" <lexxua@gmail.com> wrote:
In my company we use Yate analyzer module as a call quality monitor http://docs.yate.ro/wiki/Quality_analyzer_module . Results of tests stored in Influxdb. For gateway monitoring we use built in Freeswitch logging of DOWN state of gateway.
On Tue, May 19, 2015 at 12:04 PM Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
I'm working on two different projects for SIP testing and monitoring, and soon there will be more information.
https://github.com/voxserv/rring (documentation is still missing) this will be an automated tester that uses FreeSWITCH to originate and terminate the calls, and it will analyze the SIP messages that are received from remote side.
https://txlab.wordpress.com/2015/05/14/quality-assurance-for-voip-calls/ some scripts and work in progress for voice quality assurance.
I will make a separate posting as soon as I'm ready.
On Tue, May 19, 2015 at 3:27 AM, Jai Rangi <jprangi@didforsale.com> wrote:
Very common concerns from new Asterisk, Freeswitch, opensips and freepbx owners, How can we monitor system, what happens if service stops responding.
Here is a small howto on monitoring any SIP service with nagios. I am sure there are plenty of options and suggestions, but this is one of them and has been working out very well for us for years.
http://www.didforsale.com/monitor-sip-server
Best, -Jai
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