[Freeswitch-users] Monitoring SIP Service

[Stanislav Sinyagin]

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 at 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 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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>
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