Discussion thread for official Zabbix Template Asterisk

[datacare]

I like this template and have it working on Zabbix 6.0 with Freepbx just fine.

My only question is can there be a way to show "calls processed" in a time like today instead of since asterisk restart. I would like to show the calls processed so far today. If anyone can help with modifying this item I would greatly appreciate it!!

Rob

[zavvin]

Hello, Zabbix 6.2.6, Freepbx15, Asterisk 15, the data element "Asterisk: Get Stats" returns an error "Cannot perform request: Received HTTP/0.9 when not allowed". Can I somehow fix it yourself? Thank you.

[defkev]

Cannot perform request: Received HTTP/0.9 when not allowed

Since i just ran into this myself, you are trying to connect to AMI using HTTP, which of course doesn't work.

In addition to enabling the web manager in manager.conf (webenabled=yes) you also need to enable the embedded (mini) http server in http.conf, the bare minimum looks like:

[general]
enabled = yes
bindaddr = <your_asterisk_ipv4>

Note you will have to update the {$AMI.URL} macro to http://<your_asterisk_ipv4>:8088/rawman because the default prefix is blank.

Cheers

[Gofaizen]

Hello.....
I have error message in zabbix_server.log after I have upgraded Zabbix server from 6.2.7 to 6.2.8 :

1735820:20230309:173014.360 error reason for "sip:asterisk.get_stats" changed: Preprocessing failed for: HTTP/1.1 200 OK..Server: sip.samtelecom.ru..Date: Thu,
09 Mar 2023 13:30:12 GMT..Cache-Control: n...
1. Failed: Error: log exceeds the maximum size of 8388608 bytes.
at [anon] (zabbix.c:84) internal
at [anon] () native strict preventsyield
at [anon] (function:10)
at [anon] (function:256) preventsyield

Everything worked fine before upgrade.

Can anyone help in solving this problem ?

[basildane]

I've been using this template for years and it worked great! I just upgrade Zabbix to 6.4, and now it is broken.
Every item returns this error:

Preprocessing failed for: Preprocessing failed for: HTTP/1.1 200 OK..Server: Asterisk/19.8.0..Date: Fri, 10 Mar 2023 17:14:...
1. Failed: cannot extract value from json by path "$.version": cannot convert value to string

I also tested AMI manually http://server:8088/rawman?action=log...XXX&secret=XXX
And it works normally.

Any thoughts?

[cmartin510]

I am having the same issue after upgrading to 6.4 as well. The issue stems from the JavaScript in the GetStats item. I tried to find what log size was being exceeded. After looking through the 6.4 manual regarding preprocessing and the JavaScript objects section (https://www.zabbix.com/documentation...script_objects) I noticed it says 'The total size of all logged messages is limited to 8 MB per script execution.'. So to test, I commented out the 3 places at the top of the script with 'Zabbix.log(...'. After that the script has run without error. I don't think of this as a solution, but a workaround until the actual cause of the log size being exceeded can be determined.

[micah.quinn]

I can confirm the same behaviour after upgrading to 6.4 Removing the Zabbix.log lines in the Asterisk template prevent the preprocessing from erroring out and the subsequent problems. Before commenting out the log lines I tried increasing the maximum log size to 32M, but that did not help.

[elefuntik]

Hi everyone, I need help. I'm trying to set up asterisk template and I have a problem "Cannot perform request: Couldn't connect to server"



Zabbix-server 192.168.5.5
FreePBX 192.168.5.3


Zabbix-Server Version is 6.2.9


I'm using FreePBX-Distro with Asterisk 16



Status firewalld on zabbix-server inactive
Status iptables on freepbx-distro inactive



timedatectl

on Zabbix-server:

[root@zabbix62 ~]# timedatectl
               Local time: Wed 2023-04-05 11:17:27 +03
           Universal time: Wed 2023-04-05 08:17:27 UTC
                 RTC time: Wed 2023-04-05 08:17:24
                Time zone: Europe/Minsk (+03, +0300)
System clock synchronized: no
              NTP service: n/a
          RTC in local TZ: no​

on FreePBX-Distro:

[root@freepbx ~]# timedatectl
      Local time: Wed 2023-04-05 11:03:58 +03
  Universal time: Wed 2023-04-05 08:03:58 UTC
        RTC time: Wed 2023-04-05 11:03:53
       Time zone: Europe/Minsk (+03, +0300)
     NTP enabled: no
NTP synchronized: no
 RTC in local TZ: yes
      DST active: n/a

Warning: The system is configured to read the RTC time in the local time zone.
         This mode can not be fully supported. It will create various problems
         with time zone changes and daylight saving time adjustments. The RTC
         time is never updated, it relies on external facilities to maintain it.
         If at all possible, use RTC in UTC by calling
         'timedatectl set-local-rtc 0'.​

/etc/asterisk/manager.conf



From Zabbix-server to FreePBX-Distro

[root@zabbix62 ~]# telnet 192.168.5.3 5038
Trying 192.168.5.3...
Connected to 192.168.5.3.
Escape character is '^]'.
Asterisk Call Manager/5.0.3​

Macros in Asterisk template:

[binlog]

Hi, elefuntik. Did you find a solution?

[kb9mfd]

Was there a solution for this? I am having the exact same issue and cannot find a solution. I have everything correct and did the same tests and am getting the same results, I can connect from the zabbix server on the command line but zabbix says it cannot connect in recent data. Everything is a fresh install of the latest.

[kb9mfd]

Ok, found the solution. Ran across this post - https://www.zabbix.com/forum/zabbix-...ports-from-gui - and found zabbix by default blocks all ports (except 22, 80, 443, 10050) and you have to do this command - "setsebool -P zabbix_can_network on" (Linux) to get it to allow all ports, after that it worked just fine.

[[email protected]]

Discovery of PJSIP Accounts and Alerting for No Connected Devices

Hello everyone,

I'm reaching out to discuss the official Zabbix Template Asterisk and its monitoring capabilities for PJSIP accounts. Currently, the template works well for monitoring overall Asterisk metrics, but I would like to extend its functionality to discover each PJSIP account and generate an alert if no devices are connected to it.

I've reviewed the existing template and noticed that it focuses on general Asterisk monitoring, but it doesn't provide detailed information about individual PJSIP accounts and their associated devices. I believe it would be beneficial to have a separate template or an enhancement to the existing template that allows listing the PJSIP accounts and their associated devices without overloading the current template.

My goal is to monitor each PJSIP account separately and generate an alert when no devices are connected. This would help me ensure that all PJSIP accounts have at least one active device connected, enabling proactive troubleshooting and ensuring the availability of the accounts.

I would greatly appreciate any insights, suggestions, or guidance on how to achieve this with Zabbix. If there are any existing templates or scripts that can help with this specific requirement, I would be grateful for the pointers.

Thank you in advance for your assistance and expertise.

Best regards,

[uplync]

Semperco-

Did you get any traction on your request to monitor individual PJSIP endpoints? I'm looking for the same thing. Thanks!

[DeAngeloVickers]

Where do you find the Asterisk Manager API URL?

The /rawman one doesn't seem to work and I suspect that most of this thread is outdated...

I'm using FreePBX on AWS. I have the mini http server turned on (although I can't seem to curl it), and I have an AMI manager created.

I see people on here got it to work and I wonder how because there's absolutely no documentation on how to do it. This thread is the only thing that comes close and it's only because people can confirm it working for FreePBX .

[Hamidz]

hello. what is this error ?
asterisk server is member of vicibox

• TypeError: cannot read property 'replace' of undefined at [anon] (duktape.c:60534) internal at escapeChars (function:215) at [anon] (function:181) preventsyield at forEach () native strict preventsyield at getIaxPeerList (function:184) at [anon] (function:295) preventsyield

[Megachip]

Since i just ran into this myself, you are trying to connect to AMI using HTTP, which of course doesn't work.

In addition to enabling the web manager in manager.conf (webenabled=yes) you also need to enable the embedded (mini) http server in http.conf, the bare minimum looks like:

This really should mentioned in template Cause its disabled by default and confusion user.

Also is it required to use web (8088) and ami (5038)?

[Megachip]

Also which read rights are required? Template documentation only mentioned 2 writes.

[Megachip]

https://git.zabbix.com/projects/ZBX/repos/zabbix/browse/templates/tel/asterisk_http

Unfortunately, read permission is not enough. Asterisk requires "system" and "command" write permissions to get the AMI response for "CoreStatus" and "core show channels" requests.

Seems not be Enough

with

read = system,call,log,verbose,command,agent,user
write = system, command

i've got:

  == HTTP Manager 'monitoring' logged on from 1.2.3.4
  == HTTP Manager 'monitoring' timed out from 1.2.3.4
  == Connect attempt from '1.2.3.4' unable to authenticate
  == Connect attempt from '1.2.3.4' unable to authenticate
  == HTTP Manager 'monitoring' logged on from 1.2.3.4

with write=all it works.

So which rights (read an write) are required for monitoring?

[Megachip]

Can we have pjsip show registrations and maybe pjsip show transport-monitors as an seperate item?

[shevchenkosia]

Hello everyone, does anyone use the official template template_tel_asterisk_http.yaml? I set it up with freepbx, I get data but I need to get the IAX trunk status. The manual says that Discovery should work but it doesn't work for me and I don't see any errors. What should I do? Does Discovery work for anyone?
I have 38 item and they work but new ones don't appear.

This XML file does not appear to have any style information associated with it. The document tree is shown below.

]<ajax-response>
<script/>
<response type="object" id="unknown">
<generic response="Success" eventlist="start" message="Peer status list will follow"/>
</response>
<response type="object" id="unknown">
<generic event="PeerEntry" channeltype="IAX2" objectname="iax_dtc_idp/iax_dtc_idp" chanobjecttype="peer" ipaddress="192.168.98.14" ipport="4569" dynamic="no" trunk="yes" encryption="(aes128,keyrotate)" status="OK (20 ms)" description=""/>
</response>
<response type="object" id="unknown">
<generic event="PeerEntry" channeltype="IAX2" objectname="iax_dtc_idp_test/iax_dtc_idp_test" chanobjecttype="peer" ipaddress="192.168.98.114" ipport="4569" dynamic="no" trunk="yes" encryption="(aes128,keyrotate)" status="UNREACHABLE" description=""/>
</response>
<response type="object" id="unknown">
<generic event="PeerEntry" channeltype="IAX2" objectname="iax_idp_avto_pbx02/iax_idp_avto_pbx02" chanobjecttype="peer" ipaddress="192.168.1.14" ipport="4569" dynamic="no" trunk="yes" encryption="(aes128,keyrotate)" status="OK (20 ms)" description=""/>
</response>
<response type="object" id="unknown">
<generic event="PeerlistComplete" eventlist="Complete" listitems="3"/>
</response>
</ajax-response>

I receive data from freepbx correctly