[Freeswitch-users] Monitoring SIP Service

Discussion:

Jai Rangi

2015-05-19 01:27:36 UTC

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

Stanislav Sinyagin

2015-05-19 09:02:04 UTC

Permalink

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.

Post by Jai Rangi
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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Владимир Федоров

2015-05-19 10:35:44 UTC

Permalink

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.

Post by Stanislav Sinyagin
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.

Post by Jai Rangi
Very common concerns from new Asterisk, Freeswitch, opensips and freepbx
owners, How can we monitor system, what happens if service stops

responding.

Post by Jai Rangi
Here is a small howto on monitoring any SIP service with nagios. I am

sure

Post by Jai Rangi
there are plenty of options and suggestions, but this is one of them and

has

Post by Jai Rangi
been working out very well for us for years.
http://www.didforsale.com/monitor-sip-server
Best,
-Jai
_________________________________________________________________________
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_________________________________________________________________________
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Stanislav Sinyagin

2015-05-19 10:47:59 UTC

Permalink

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.

Post by ÐÐ»Ð°Ð´Ð¸Ð¼Ð¸Ñ Ð¤ÐµÐ´Ð¾ÑÐ¾Ð²
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.

Post by Jai Rangi
Very common concerns from new Asterisk, Freeswitch, opensips and freepbx
owners, How can we monitor system, what happens if service stops

responding.

Post by Jai Rangi
Here is a small howto on monitoring any SIP service with nagios. I am

sure

Post by Jai Rangi
there are plenty of options and suggestions, but this is one of them

and has

Post by Jai Rangi
been working out very well for us for years.
http://www.didforsale.com/monitor-sip-server
Best,
-Jai

_________________________________________________________________________

Post by Jai Rangi
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

http://lists.freeswitch.org/mailman/options/freeswitch-users

Post by Jai Rangi
http://www.freeswitch.org

Volodymyr Fedorov

2015-05-19 19:18:20 UTC

Permalink

Hello, I`m pushed our little python script to Github
https://github.com/lexxua/qcheck it`s not ideal but makes dirty job. Maybe
if I have some more time I will rewrite it for mod_bert usage.

Post by 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.

Post by ÐÐ»Ð°Ð´Ð¸Ð¼Ð¸Ñ Ð¤ÐµÐ´Ð¾ÑÐ¾Ð²
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.

Post by Jai Rangi
Very common concerns from new Asterisk, Freeswitch, opensips and

freepbx

Post by Jai Rangi
owners, How can we monitor system, what happens if service stops

responding.

Post by Jai Rangi
Here is a small howto on monitoring any SIP service with nagios. I am

sure

Post by Jai Rangi
there are plenty of options and suggestions, but this is one of them

and has

Post by Jai Rangi
been working out very well for us for years.
http://www.didforsale.com/monitor-sip-server
Best,
-Jai

_________________________________________________________________________

http://lists.freeswitch.org/mailman/options/freeswitch-users

Post by Jai Rangi
http://www.freeswitch.org

--
Cheers !