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gmaruzz at gmail.com
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 Posted: Wed Apr 21, 2021 5:21 am Post subject: [Freeswitch-users] [ANNOUNCE]: sipnagios, a Nagios Plugin to |
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Hello fellow VoIPers and RTCers,
on GitHub there is an early release of sipnagios, opensource.
check it out: https://github.com/gmaruzz/sipnagios
sipnagios is a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc)
sipnagios implements the Nagios plugin API for monitoring and performance data.sipnagios.c is a modification of the original siprtp.c sample in pjproject distribution. Supposedly, it works on Linux, Windows, and anywhere you can compile pjproject on.It makes a call, checks all the various resulting values (mos, rtt, pdd, tta, jitter, packet loss, bytes and packets transferred, and so on). It verifies these values are included into acceptable, warning, or critical ranges.If the call has gone well, sipnagios print performance data for Nagios graphs, and returns 0.If the call fails, or if its measured values are not inside acceptable ranges, it exits with Nagios conventional WARNING or CRITICAL values.
mos calculation is scraped from Julien Chavanton work (VoIP Patrol, on GitHub too) I can't even understand  (merci Julien!)
Enjoy!
-giovanni
--
Sincerely,
Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18 |
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social at bohboh.info
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| Posted: Wed Apr 21, 2021 2:10 pm Post subject: [Freeswitch-users] [ANNOUNCE]: sipnagios, a Nagios Plugin to |
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I'm testing the plugin on Nagios but no output...
/usr/src/pjproject-2.11/pjsip-apps/bin/samples/x86_64-unknown-linux-gnu/sipnagios -d 60 --local-port=5060 --ip-addr=PBXIP --local-siprealm=asterisk --local-user=100 --local-password=password [url=sip:0749941093@acme.cloudpbx.opentelecom.it:5030]sip:0749941093@acme.cloudpbx.opentelecom.it:5030[/url]
Any hint?
| Quote: | ---
I'm SoCIaL, MayBe |
El 21/04/2021 a las 4:40 a. m., Giovanni Maruzzelli escribió:
| Quote: | Hello fellow VoIPers and RTCers,
on GitHub there is an early release of sipnagios, opensource.
check it out: https://github.com/gmaruzz/sipnagios
sipnagios is a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc)
sipnagios implements the Nagios plugin API for monitoring and performance data.sipnagios.c is a modification of the original siprtp.c sample in pjproject distribution. Supposedly, it works on Linux, Windows, and anywhere you can compile pjproject on.It makes a call, checks all the various resulting values (mos, rtt, pdd, tta, jitter, packet loss, bytes and packets transferred, and so on). It verifies these values are included into acceptable, warning, or critical ranges.If the call has gone well, sipnagios print performance data for Nagios graphs, and returns 0.If the call fails, or if its measured values are not inside acceptable ranges, it exits with Nagios conventional WARNING or CRITICAL values.
mos calculation is scraped from Julien Chavanton work (VoIP Patrol, on GitHub too) I can't even understand (merci Julien!)
Enjoy!
-giovanni
--
Sincerely,
Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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gmaruzz at gmail.com
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| Posted: Thu Apr 22, 2021 2:40 am Post subject: [Freeswitch-users] [ANNOUNCE]: sipnagios, a Nagios Plugin to |
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On Wed, Apr 21, 2021 at 8:38 PM Social Boh <social@bohboh.info (social@bohboh.info)> wrote:
| Quote: |
I'm testing the plugin on Nagios but no output...
/usr/src/pjproject-2.11/pjsip-apps/bin/samples/x86_64-unknown-linux-gnu/sipnagios -d 60 --local-port=5060 --ip-addr=PBXIP --local-siprealm=asterisk --local-user=100 --local-password=password sip:0749941093@acme.cloudpbx.opentelecom.it:5030
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Execute it from the command line, and see what it prints. If needed, set logging options to higher values.
local-port is the port you want to use to go out from sipnagios (the udp socket sipnagios creates)
ip-addr is the local ip address sipnagios uses to create the udp socket, MUST be the ip address of the machine sipnagios is running on (eg: 192.168.1.23)
local-siprealm is the sip domain sipnagios will use to authenticate the invite on the remote sip server
local user is the auth and login user on the remote sip server
password is its password
the last argument is the full sip uri sipnagios will send the invite to, and is not for sure the one you used. MUST be a sip uri that is reachable by the remote sip server
For any other problem, please open a github issue, so we don't squat this mailing list
Have a nice day,
-giovanni
| Quote: |
Any hint?
| Quote: | ---
I'm SoCIaL, MayBe |
El 21/04/2021 a las 4:40 a. m., Giovanni Maruzzelli escribió:
| Quote: | Hello fellow VoIPers and RTCers,
on GitHub there is an early release of sipnagios, opensource.
check it out: https://github.com/gmaruzz/sipnagios
sipnagios is a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc)
sipnagios implements the Nagios plugin API for monitoring and performance data.sipnagios.c is a modification of the original siprtp.c sample in pjproject distribution. Supposedly, it works on Linux, Windows, and anywhere you can compile pjproject on.It makes a call, checks all the various resulting values (mos, rtt, pdd, tta, jitter, packet loss, bytes and packets transferred, and so on). It verifies these values are included into acceptable, warning, or critical ranges.If the call has gone well, sipnagios print performance data for Nagios graphs, and returns 0.If the call fails, or if its measured values are not inside acceptable ranges, it exits with Nagios conventional WARNING or CRITICAL values.
mos calculation is scraped from Julien Chavanton work (VoIP Patrol, on GitHub too) I can't even understand (merci Julien!)
Enjoy!
-giovanni
--
Sincerely,
Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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--
Sincerely,
Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18 |
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