<div dir="ltr"><div dir="ltr">On Wed, Apr 21, 2021 at 8:38 PM Social Boh <<a href="mailto:social@bohboh.info">social@bohboh.info</a>> wrote:<br></div><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div>
<p>I'm testing the plugin on Nagios but no output...</p>
<p>/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
<a>sip:0749941093@acme.cloudpbx.opentelecom.it:5030</a></p>
<p></p></div></blockquote><div><br></div><div>Execute it from the command line, and see what it prints. If needed, set logging options to higher values.</div><div><br></div><div>local-port is the port you want to use to go out from sipnagios (the udp socket sipnagios creates)</div><div>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)</div><div>local-siprealm is the sip domain sipnagios will use to authenticate the invite on the remote sip server</div><div>local user is the auth and login user on the remote sip server</div><div>password is its password</div><div>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<br><br>For any other problem, please open a github issue, so we don't squat this mailing list :)</div><div><br></div><div>Have a nice day,</div><div><br></div><div>-giovanni<br><br><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div> </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div><p>Any hint?<br>
</p>
<pre cols="72">---
I'm SoCIaL, MayBe</pre>
<div>El 21/04/2021 a las 4:40 a. m.,
Giovanni Maruzzelli escribió:<br>
</div>
<blockquote type="cite">
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<div>Hello fellow VoIPers and RTCers,<br>
<br>
on GitHub there is an early release of sipnagios, opensource.<br>
<br>
check it out: <a href="https://github.com/gmaruzz/sipnagios" target="_blank">https://github.com/gmaruzz/sipnagios</a><br>
<br>
sipnagios is a Nagios Plugin to check Call Quality in SIP VoIP
(compatible with checkmk, etc)<br>
<br>
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.<br>
<br>
mos calculation is scraped from Julien Chavanton work (VoIP
Patrol, on GitHub too) I can't even understand :) (merci
Julien!)<br>
<br>
Enjoy!<br>
<br>
-giovanni<br>
</div>
-- <br>
<div dir="ltr">Sincerely,<br>
<br>
Giovanni Maruzzelli<br>
OpenTelecom.IT<br>
cell: +39 347 266 56 18<br>
<br>
</div>
</div>
<br>
<fieldset></fieldset>
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</blockquote>
</div>
</blockquote></div><br clear="all"><div><br></div>-- <br><div dir="ltr" class="gmail_signature">Sincerely,<br><br>Giovanni Maruzzelli<br>OpenTelecom.IT<br>cell: +39 347 266 56 18<br><br></div></div>