[Freeswitch-users] Monitoring SIP Service

[Владимир Федоров]

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 at 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 at 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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> > http://www.freeswitchsolutions.com
> >
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> > http://www.cluecon.com
> >
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>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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