@amardeeprana wrote:

Hi
Asterisk Service Taking High CPU USAGE 100% ( 4 Cores ).

FreePBX 14.0.13.34

Please suggest.

Thanks
Amardeep

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@713West wrote:

Newb here with a brand new installation of FreePBX 15.0.16.73. I have most things working but I am having a problem with call recording.

I set up call recording for one of my extensions. After a call is recorded it can be found as a wav file with a numeric name in /var/spool/asterisk/monitor/. However, another 1kb file with the expected type-destination-source-datestamp-timestamp-uniqueid name is also present in /var/spool/asterisk/monitor/2020/09/01. I thought the recording would have been moved to the date sub-directory but all I am getting is this empty 1kb file. What could be causing this?

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@gtibbetts wrote:

I just finished installing free pbx. I just tried to go to the ip address of the server through the web browser and it says it is taking too long to load so try again. I think I installed Free PBX 16. I did it on the page

Has anyone seen this issue? Can anyone help?

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@rzworks wrote:

Hi guys, I was wondering if any of you are is using a sofware to integrate with Freepbx that give some statistic about the CDR Calls with reports and graphs. I tried to look at http://www.cdr-stats.org/ but seems 5 years old, and https://www.asternic.net/index.php but seems is not so full of options (for example see statistic by country etc… Do you guys use anything (paid or free) that could reach this goal?
Thank you for you help!

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@AllianceDoug wrote:

The FreePBX 14.0.13.34 installation is on a virtual machine with a 20 GB disk.

I’m getting the following email notifications hourly:

Storage space is getting high on the following drives of your system:
/dev/mapper/SangomaVG-root is 76% full

Seems like the email text is probably meant to be storage space is getting “low” rather than “high”…

Based on searching through other posts, I have checked the following directories using the “ls -alh | less” command to give the total size of all files in the directory:

/var/log/asterisk total 746M
/var/spool/asterisk/backup total 4K
/var/spool/asterisk/monitor total 0
/var/lib/asterisk/backups no such directory
/var/spool/asterisk/backup/Default_backup total 7.7GB

Here is the Default_backup folder:

FreePBX Default_backup Folder.png819×418 9.51 KB

Looks like the backups stopped in February for some reason.

Under Admin -> Backup and Restore, here are my backup settings:

Here is the SangomaVG directory:

Under Applications -> Call Recording, there are no recording plans and it has always been turned off.

Under Settings -> Voicemail Admin, voicemail disk usage is very low:

Total drive usage:

FreePBX Storage State.png770×560 35.9 KB

Not sure where to go from here. We appreciate any input.

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@osvalcosta wrote:

How many simultaneous calls FreePBX can handle with good hardware ??

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@scristopher71 wrote:

Hello,
I’m a little stumped on this one. I just migrated a freepbx 12 server to 15 and everything seemed to of gone very smoothly, however the only thing that is not working is the intercom feature. I noticed there is no feature code for the intercom in the feature code manager, and even though intercom is enabled for the extensions, I am getting an error when trying to intercom them:

> 0x7f64e80ae840 -- Strict RTP learning after remote address set to: 192.168.1.132:3000
[2020-09-03 10:15:50] NOTICE[2780][C-00000087]: chan_sip.c:26817 handle_request_invite: 

Call from '200' (xx.xxx.xxx.xx:5060) to extension '*801000' rejected because extension not found in context 'from-internal'.

I can see that intercom is enabled for this extension 1000:

[root@freepbx asterisk]# rasterisk -rx 'database show' | grep intercom | grep  1000
/AMPUSER/1000/intercom                            : enabled                  
/AMPUSER/1000/intercom/override                   : reject     

I compared with another system and it appears that in extensions_additional.conf the intercom code is actually disabled:

[root@freepbx asterisk]# grep 'INTERCOMCODE =' extensions_additional.conf 
INTERCOMCODE = nointercom

On a system where it is working properly it is set to *80:

[root@pbx asterisk]# grep 'INTERCOMCODE =' extensions_additional.conf 
INTERCOMCODE = *80

The paging module is installed (I uninstalled pagepro though to see if maybe there was a conflict)

[root@freepbx asterisk]# fwconsole ma list | grep paging
| paging               | 15.0.4.23  | Enabled                           | GPLv3+      |
| pagingpro            |            | Not Installed (Locally available) | Commercial  |

Also it appears I am missing paging in the featurecodes table

On a working system:

MariaDB [asterisk]> select * from featurecodes where modulename='paging';
+------------+-----------------+------------------------+----------+-------------+------------+---------+-------------+
| modulename | featurename     | description            | helptext | defaultcode | customcode | enabled | providedest |
+------------+-----------------+------------------------+----------+-------------+------------+---------+-------------+
| paging     | intercom-off    | User Intercom Disallow |          | *55         |            |       1 |           0 |
| paging     | intercom-on     | User Intercom Allow    |          | *54         |            |       1 |           0 |
| paging     | intercom-prefix | Intercom prefix        |          | *80         |            |       1 |           0 |
+------------+-----------------+------------------------+----------+-------------+------------+---------+-------------+
3 rows in set (0.01 sec)

On the broken system:

MariaDB [asterisk]>  select * from featurecodes where modulename='paging';
Empty set (0.00 sec)

Now I see where the issue here is but just am not sure where to fix it, anyone know? Im contemplating just adding the values into the featurecodes table and seeing what happens but not sure if that will resolve my issue.

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@vbman213 wrote:

Got word from a school I service that their bell paging schedule missed a ring today. Any ideas on what to track down? It works fairly consistently but some rings seem to never reach our multicast endpoints.

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@d2o wrote:

Hello,

I don’t know if I am right here, but I have a problem with my freepbx, using the correct CID, Outbound route and/or trunk.

I have configured 3 trunks, 3 in- and outbound routes and 3 extensions and each extension’s using one inbound route and one trunk (with own number),
All calls to the 3 extensions are working correctly. If I dial phone nr. 1, the call is routed to extension 1 over the trunk 1 etc.

But all extensions are only using trunk 1 for outgoing calls - or using the CID from trunk 1.
I want all extension using its attached inbound, outbound route und trunk.
For example:

Trunk 1 Number: +49 12345678
Extension is attached to trunk 1 for incoming calls with inbound route. Works fine.
Extension is doing outgoing calls, CID of Trunk 1 is used (and shown in the call receiving phone).

Trunk 2 Number: +49 87654321
Extension 2 is attached to trunk 2 for incoming calls with inbound route, Works fine.
Extension 2 is doing outgoing calls, CID of trunk 1 is used. Why? Ext. 2 is attached to its own outbound route with own CID and the outbound route is attached to trunk 2. But on the call receiving phone, the CID of trunk 1 is shown.

Extension 3 is doing the same - using CID from trunk 1 or using trunk1. Should use outbound route 3 with trunk 3 (and its own CID).

I have no clue, why Extension 2 and 3 are using trunk 1 or its CID. Should be trunk 2 and 3 and CID of trunk 2 and 3

Any hints?

Regards.

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@faisalkhan wrote:

hi all,

I am observing something strange which I need to share with you guys.

Freepbx 15 asterisk 16.9.0 console is not showing any logs of the calls when I connect to asterisk console with the command “asterisk -rvvvv”

asterisk -rvvvv

Asterisk 16.9.0, Copyright © 1999 - 2018, Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 16.9.0 currently running on 01 (pid = 37298)
01CLI>
01CLI>
01CLI>
01CLI>

however I have live traffic of calls and even If I tried to make a call, call went through fine but no logs appearing on the CLI.

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@voipgenius wrote:

Where is fax gateway getting it’s settings from? It appears to be setting it’s own datagram size and ports rather than those listed in the udptl.conf, sip.conf, res_fax.conf

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@mvogel4949 wrote:

I have a relatively new install using FreePBX14 and ASterisk 13. I’m seeing a large number of hung channels that stay up even after rebooting the phones. Is there a way to hang up a specific channel or is fwconsole restart the only way?

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@leo85 wrote:

Hi guys,

I never used backup module before, I may need to use it now. I have one vm that I’m trying convert to Hyper-V vm. The process works but it complains about devices missing like/dev/sangoma/swap when vm starts.

If I backup and apply it to new system what will happen then?
Like, will I have extensions, IVR etc?

Thank you!!!

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@voipgenius wrote:

Where is fax gateway getting it’s settings from? It appears to be setting it’s own datagram size and ports rather than those listed in the udptl.conf, sip.conf, res_fax.conf

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@mvogel4949 wrote:

I had a system calling large number of extensions and all of a sudden it just stopped. I follow the call in the logfiles and this seems to be where the call dies. With more testing I see that over 36 extensions I see the dial argument takes format message and then the call dies.

[2020-09-05 22:26:03] VERBOSE[8551][C-00000316] pbx.c: Executing [s@macro-dial:31] ExecIf("SIP/wiresip1-000002be", "0?Set(ds=PJSIP/2010/sip:2010@10.101.0.73:65060&Local/902010@zulu-call&PJSIP/2011/sip:2011@10.101.0.50:65060&PJSIP/2009/sip:2009@10.101.0.70:65060&Local/902009@zulu-call&PJSIP/2001/sip:2001@10.101.0.116:65060&Local/902001@zulu-call&PJSIP/2003/sip:2003@10.101.0.52:5060&Local/902003@zulu-call&PJSIP/2005/sip:2005@10.101.0.71:65060&Local/902005@zulu-call&PJSIP/2004/sip:2004@10.101.0.63:65060&Local/902004@zulu-call&PJSIP/2007/sip:2007@10.101.0.92:5060&Local/902007@zulu-call&PJSIP/2006/sip:2006@10.101.0.69:65060&Local/902006@zulu-call&PJSIP/2012/sip:2012@10.101.0.67:65060&Local/902012@zulu-call&PJSIP/2013/sip:2013@10.101.0.59:65060&Local/902013@zulu-call&Local/904010@zulu-call&PJSIP/4045/sip:4045@10.20.20.47:65060&Local/904045@zulu-call&PJSIP/4027/sip:4027@10.20.17.31:65060&Local/904027@zulu-call&Local/904167@zulu-call&PJSIP/4116/sip:4116@10.20.16.10:65060&Local/904116@zulu-call&Local/904028@zulu-call&PJSIP/4008/sip:4008@10.20.20.106:65060&Local/904008@zulu-call&PJSIP/4086/sip:4086@10.20.20.188:65060&Local/904086@zulu-call&PJSIP/4108/sip:4108@10.20.20.109:65060&Local/904108@zulu-call&PJSIP/4029/sip:4029@10.20.20.110:65060&Local/904029@zulu-call&PJSIP/4021/sip:4021@10.20.16.151:65060&Local/904021@zulu-call&PJSIP/4173/sip:4173@10.20.20.176:65060&Local/904173@zulu-call&PJSIP/4022/sip:4022@10.20.16.213:65060&Local/904022@zulu-call&PJSIP/4053/sip:4053@10.20.16.202:65060&Local/904053@zulu-call&PJSIP/4478/sip:4478@10.20.16.197:65060&Local/904478@zulu-call&PJSIP/4848/sip:4848@10.20.16.203:65060&Local/904848@zulu-call&PJSIP/4148/sip:4148@10.20.20.90:65060&PJSIP/4160/sip:4160@10.20.20.97:65060&Local/904160@zulu-call&PJSIP/4087/sip:4087@10.20.16.144:65060&PJSIP/4427/sip:4427@10.20.20.185:65060&Local/904427@zulu-call&PJSIP/4473/sip:4473@10.20.16.212:65060&Local/904473@zulu-call&PJSIP/4112/sip:4112@10.20.20.87:65060&Local/904112@zulu-call&PJSIP/4124/sip:4124@10.20.16.207:65060&Local/904124@zulu-call&PJSIP/4157/sip:4157@10.20.16.137:65060&Local/904157@zulu-call&PJSIP/4049/sip:4049@10.20.20.94:65060&Local/904049@zulu-call&PJSg)") in new stack
[2020-09-05 22:26:03] VERBOSE[8551][C-00000316] pbx.c: Executing [s@macro-dial:32] Dial("SIP/wiresip1-000002be", "PJSIP/2010/sip:2010@10.101.0.73:65060&Local/902010@zulu-call&PJSIP/2011/sip:2011@10.101.0.50:65060&PJSIP/2009/sip:2009@10.101.0.70:65060&Local/902009@zulu-call&PJSIP/2001/sip:2001@10.101.0.116:65060&Local/902001@zulu-call&PJSIP/2003/sip:2003@10.101.0.52:5060&Local/902003@zulu-call&PJSIP/2005/sip:2005@10.101.0.71:65060&Local/902005@zulu-call&PJSIP/2004/sip:2004@10.101.0.63:65060&Local/902004@zulu-call&PJSIP/2007/sip:2007@10.101.0.92:5060&Local/902007@zulu-call&PJSIP/2006/sip:2006@10.101.0.69:65060&Local/902006@zulu-call&PJSIP/2012/sip:2012@10.101.0.67:65060&Local/902012@zulu-call&PJSIP/2013/sip:2013@10.101.0.59:65060&Local/902013@zulu-call&Local/904010@zulu-call&PJSIP/4045/sip:4045@10.20.20.47:65060&Local/904045@zulu-call&PJSIP/4027/sip:4027@10.20.17.31:65060&Local/904027@zulu-call&Local/904167@zulu-call&PJSIP/4116/sip:4116@10.20.16.10:65060&Local/904116@zulu-call&Local/904028@zulu-call&PJSIP/4008/sip:4008@10.20.20.106:65060&Local/904008@zulu-call&PJSIP/4086/sip:4086@10.20.20.188:65060&Local/904086@zulu-call&PJSIP/4108/sip:4108@10.20.20.109:65060&Local/904108@zulu-call&PJSIP/4029/sip:4029@10.20.20.110:65060&Local/904029@zulu-call&PJSIP/4021/sip:4021@10.20.16.151:65060&Local/904021@zulu-call&PJSIP/4173/sip:4173@10.20.20.176:65060&Local/904173@zulu-call&PJSIP/4022/sip:4022@10.20.16.213:65060&Local/904022@zulu-call&PJSIP/4053/sip:4053@10.20.16.202:65060&Local/904053@zulu-call&PJSIP/4478/sip:4478@10.20.16.197:65060&Local/904478@zulu-call&PJSIP/4848/sip:4848@10.20.16.203:65060&Local/904848@zulu-call&PJSIP/4148/sip:4148@10.20.20.90:65060&PJSIP/4160/sip:4160@10.20.20.97:65060&Local/904160@zulu-call&PJSIP/4087/sip:4087@10.20.16.144:65060&PJSIP/4427/sip:4427@10.20.20.185:65060&Local/904427@zulu-call&PJSIP/4473/sip:4473@10.20.16.212:65060&Local/904473@zulu-call&PJSIP/4112/sip:4112@10.20.20.87:65060&Local/904112@zulu-call&PJSIP/4124/sip:4124@10.20.16.207:65060&Local/904124@zulu-call&PJSIP/4157/sip:4157@10.20.16.137:65060&Local/904157@zulu-call&PJSIP/4049/sip:4049@10.20.20.94:65060&Local/904049@zulu-call&PJSb(func-apply-sipheaders^s^1),") in new stack
[2020-09-05 22:26:03] WARNING[8551][C-00000316] app_dial.c: Dial argument takes format (technology/resource)
[2020-09-05 22:26:03] VERBOSE[8551][C-00000316] app_macro.c: Spawn extension (macro-dial, s, 32) exited non-zero on 'SIP/wiresip1-000002be' in macro 'dial'
[2020-09-05 22:26:03] VERBOSE[8551][C-00000316] pbx.c: Spawn extension (ext-group, 528, 20) exited non-zero on 'SIP/wiresip1-000002be'
[2020-09-05 22:26:03] VERBOSE[8551][C-00000316] pbx.c: Executing [h@ext-group:1] Macro("SIP/wiresip1-000002be", "hangupcall,") in new stack
[2020-09-05 22:26:03] VERBOSE[8551][C-00000316] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("SIP/wiresip1-000002be", "1?theend") in new stack
[2020-09-05 22:26:03] VERBOSE[8551][C-00000316] pbx_builtins.c: Goto (macro-hangupcall,s,3)

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@kfeen wrote:

I just cloned a FreePBX 13 system from an old server to a newer one on a different network. What do I need to change? I’ve updated the firewall, the IP address, and some other key network settings but I know there’s some things I’m missing.

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@Dazoman wrote:

Hello all.

I know the annoyance of a newbie asking silly questions, after looking for answers I found it not a straight forward “yes it can be done” or “no you idiot”.

The question I have is… I have a radio station, I am in the office and if i wish to speak to a presenter in the studio am i able to set up a FreePBX onto a computer. Then connect a phone in the studio and one in my office, just to be able to call the extension.

I DO NOT wish to make any outgoing calls or nor to receive any in-coming calls.

I literally just want it set up to ring extension “100” or whatever and they able to call extension “101” to call me in my office.

So, can this be done with just FreePBX and 2 IP phones. Or do i still need a SIP or a provider.

Thank you for reading.

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@pcdub wrote:

FreePBX Version: 14.0.13.34
Today I made some standard GUI Applications changes, then clicked Submit, then clicked Apply Config, and I’m receiving the following error:

Reload failed because retrieve_conf encountered an error: 1
exit: 1
Unable to continue. Cron line added didn’t remain in crontab on final check. Check /tmp/cron.error for reason. in /var/www/html/admin/libraries/BMO/Cron.class.php on line 126
#0 /var/www/html/admin/libraries/BMO/Cron.class.php(203): FreePBX\Cron->addLine(’* * * * * [ -e …’)
#1 /var/www/html/admin/libraries/BMO/Job.class.php(246): FreePBX\Cron->add(’* * * * * [ -e …’)
#2 /var/www/html/admin/libraries/BMO/Job.class.php(91): FreePBX\Job->init()
#3 /var/www/html/admin/libraries/BMO/Job.class.php(73): FreePBX\Job->add(‘timeconditions’, ‘schedtc’, NULL, ‘FreePBX\modules…’, ‘* * * * ', 30, true, 100)
#4 /var/www/html/admin/modules/timeconditions/Timeconditions.class.php(168): FreePBX\Job->addClass(‘timeconditions’, ‘schedtc’, ‘FreePBX\modules…’, ’ * * * *’)
#5 /var/www/html/admin/modules/timeconditions/functions.inc.php(227): FreePBX\modules\Timeconditions->updateCron()
#6 /var/www/html/admin/libraries/BMO/DialplanHooks.class.php(95): timeconditions_get_config(‘asterisk’)
#7 /var/lib/asterisk/bin/retrieve_conf(860): FreePBX\DialplanHooks->processHooks(‘asterisk’, Array)
#8 {main}
1 error(s) occurred, you should view the notification log on the dashboard or main screen to check for more details.

It seemed to start (coincidentally or not) this afternoon about the time when I found out all our scheduled page groups were not playing. Manual tests confirmed the page groups worked, just not the schedules. I made some changes, clicked Submit then received the above error. It was all working last Friday 9/4/20.

I’ve attempted many things based on other posts including:
Checked the /tmp/cron.error file but the file is blank
fwconsole reload
fwconsole restart
Rebooting physical server
Renamed /var/spool/cron/asterisk the fwconsole reload then fwconsole restart to have it recreate the file

chown -R asterisk /var/spool/cron then Apply Config
chmod -R 777 /var/spool/cron then Apply Config

Downloaded the timeconditions module via command line to re-install (it upgraded it).

Many more reloads, restarts, reboots.
Still the same error

Commands outputs:
Contents of /var/spool/cron/asterisk on both this system and from a backup from about two weeks ago when it was all working:
[root@pbx ~]# crontab -e -u asterisk
@daily [ -e /var/www/html/admin/modules/sysadmin/bin/check_portal.php ] && /var/www/html/admin/modules/sysadmin/bin/check_portal.php
@daily [ -x /var/lib/asterisk/agi-bin/ddns_client.php ] && /var/lib/asterisk/agi-bin/ddns_client.php
@daily /usr/sbin/fwconsole pms mk_dirty > /dev/null 2>&1

• • • • • /usr/sbin/fwconsole pms wu_alert > /dev/null 2>&1
          */15 * * * * [ -e /etc/asterisk/firewall.enabled ] && touch /var/spool/asterisk/incron/firewall.firewall
          @daily [ -x /var/lib/asterisk/bin/freepbx_sipstation_check ] && /var/lib/asterisk/bin/freepbx_sipstation_check 2>&1 > /dev/null
          0 0 25 12 * /usr/sbin/fwconsole pagingpro --calendarpage 702 -t 1545714000 2>&1 >/dev/null
          0 0 25 12 * /usr/sbin/fwconsole pagingpro --calendarpage 612 -t 1545714000 2>&1 >/dev/null
          @daily [ -e /var/www/html/admin/modules/sysadmin/bin/check_portal.php ] && /var/www/html/admin/modules/sysadmin/bin/check_portal.php >/dev/null 2>&1
          @monthly ID=freepbx_backup_1 /var/lib/asterisk/bin/backup.php --id=1 >/dev/null 2>&1
          1 1 1,2,3,4,5,6,7,15,16,17,18,19,20,21,29,30,31 * * ID=freepbx_backup_2 /var/lib/asterisk/bin/backup.php --id=2 >/dev/null 2>&1
          1 2 8,9,10,11,12,13,14,22,23,24,25,26,27,28 * * ID=freepbx_backup_3 /var/lib/asterisk/bin/backup.php --id=3 >/dev/null 2>&1
          @hourly [ -x /var/lib/asterisk/bin/storage.php ] && /var/lib/asterisk/bin/storage.php >/dev/null 2>&1
• • • • • /usr/sbin/fwconsole queuestats --syncall >> /tmp/reader.log 2>&1
          28 0 * * * /usr/sbin/fwconsole certificates --updateall -q 2>&1 >/dev/null
          0 1 * * * /var/www/html/admin/modules/iotserver/bin/check_license.php
          0 2 * * * /var/www/html/admin/modules/iotserver/bin/check_certificates.php
          0 4 * * * /var/www/html/admin/modules/iotserver/bin/refresh_tzoffsets.php
          59 23 * * 0 /var/lib/asterisk/bin/queue_reset_stats.php --id=1320
• 4 * * * /usr/sbin/fwconsole util cleanplaybackcache -q
  15 22 * * 5 [ -e /usr/sbin/fwconsole ] && /usr/sbin/fwconsole ma listonline --sendemail -q > /dev/null 2>&1
  15 23 * * 5 [ -e /usr/sbin/fwconsole ] && /usr/sbin/fwconsole sys upgradeall --sendemail -q > /dev/null 2>&1
  15 0 * * 6 [ -e /usr/sbin/fwconsole ] && /usr/sbin/fwconsole ma upgradeall --sendemail -q > /dev/null 2>&1
• • • • • [ -e /usr/sbin/fwconsole ] && sleep $((RANDOM%30)) && /usr/sbin/fwconsole job --run --quiet 2>&1 > /dev/null

[root@pbx ~]# fwconsole job --list
±—±-----------------±--------------±-------------±--------------------±---------------------------------------------------------------------------------------------------------------------------------------±--------+
| ID | Module | Job | Cron | Next Run | Action | Enabled |
±—±-----------------±--------------±-------------±--------------------±---------------------------------------------------------------------------------------------------------------------------------------±--------+
| 12 | pagingpro | scheduler | * * * * * | 2020-09-08 23:08:00 | Class: FreePBX\modules\Pagingpro\Job | Yes |
| 1 | calendar | sync | * * * * * | 2020-09-08 23:08:00 | Class: FreePBX\modules\Calendar\Job | Yes |
| 2 | qxact_reports | sync | */5 * * * * | 2020-09-08 23:10:00 | Command: /usr/sbin/fwconsole qxactreports --sync -q 2>&1 > /dev/null | Yes |
| 3 | recording_report | clean | 30 22 * * * | 2020-09-09 22:30:00 | Command: /usr/sbin/fwconsole recordingreports -s -c | Yes |
| 4 | recording_report | backup | 0 0 1 * * | 2020-10-01 00:00:00 | Command: php /var/lib/asterisk/agi-bin/backuprecordings.php | Yes |
| 5 | sysadmin | checkportal | @daily | 2020-09-09 00:00:00 | Command: [ -e /var/www/html/admin/modules/sysadmin/bin/check_portal.php ] && /var/www/html/admin/modules/sysadmin/bin/check_portal.php | Yes |
| 6 | sysadmin | ddnsupdate | @daily | 2020-09-09 00:00:00 | Class: FreePBX\modules\Sysadmin\Job\DdnsUpdate | Yes |
| 7 | sysadmin | checkstorage | @hourly | 2020-09-09 00:00:00 | Command: [ -x /var/lib/asterisk/bin/storage.php ] && /var/lib/asterisk/bin/storage.php | Yes |
| 8 | timeconditions | schedtc | * * * * * | 2020-09-08 23:08:00 | Class: FreePBX\modules\Timeconditions\Job | Yes |
| 9 | userman | syncall | */15 * * * * | 2020-09-08 23:15:00 | Class: FreePBX\modules\Userman\Job | Yes |
| 10 | sysadmin | updatelicense | @daily | 2020-09-09 00:00:00 | Command: [ -x /var/lib/asterisk/agi-bin/update_license.php ] && /var/lib/asterisk/agi-bin/update_license.php --delay | Yes |
| 11 | dashboard | scheduler | * * * * * | 2020-09-08 23:08:00 | Class: FreePBX\modules\Dashboard\Job | Yes |
| 13 | pms | wu_alert | * * * * * | 2020-09-08 23:08:00 | Class: FreePBX\modules\Pms\Job | No |
±—±-----------------±--------------±-------------±--------------------±---------------------------------------------------------------------------------------------------------------------------------------±--------+

I don’t see any errors or corruption in the /var/spool/cron/asterisk file but maybe I’m missing it so am now at a loss what to do next. Please advise. See any errors or corruption? Thoughts? Suggestions?

Thanks!

Patrick

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@kfeen wrote:

I’m looking for a way to see a list of all registered phones and what extension they’re registering to as well as their LAN and WAN IP. Where would I find this information in FreePBX 13?

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@POTShead wrote:

I’m a ‘prosumer’ home user with an updated v15.0.16.72 system for my single-trunk, 6-xtn setup, living a set-and-forget life (except for yesterday when I forced all module and system updates, before confirming my observed problem.)

On May 21 at ~18h00 EDT [-0400] something oddish happened - my MIXMON_POST script stopped being given its usual helping of parameters. (I use this post-call script to upload the newly-created call recording to a storage server.)

In fact it worked properly at 18:01 of that day, and 6 minutes later it was then getting no parameters… and this no parameters problem has persisted through multiple reboots and until yesterday when I got down to finally fixing it.

I normally would set my desired script and its parameters via the GUI - Settings/AdvancedSetttings and then actual setting: PostCallRecordingScript [MIXMON_POST variable], and thusly:

/usr/local/bin/upload_newrec ‘^{ARG2}’ ‘^{ARG3}’ ‘^{FROMEXTEN}’ ‘^{TIMESTR}’ ‘^{UNIQUEID}’ ‘^{CALLERID(name)}’ ‘^{CALLFILENAME}’ ‘^{MIXMON_FORMAT}’


but no matter what fiddling I did with carets, dollarsigns, single-or-doublequotes I could not get parameters passed to the script (the script would fire but always with no parameters).

So my apparently-observed bug is that updating this GUI field to send variables parameters does not seem to send along the variables parameters.

I eventually cottoned on to the fact that I could just attack the cfgfiles directly (but doubtless at my own risk… next time I hit Submit on the GUI likely), and so I was able to restore my original and desired behaviour via ensuring that each of two files has this in its config for the MIXMON_POST variable:

MIXMON_POST = {{see above, right after ‘thusly’}}

and those two files being:

- /etc/amportal.conf
- /etc/asterisk/extensions_additional.conf


Curiously, even when those two files have the desired setting, what is displayed in the GUI as being the MIXMON_POST setting is unrelated (seems to be a one-way write, from GUI to cfgfiles?)

So before I go and log an ‘issue’ over this apparent problem I thought that I’d see if it was unique to me or if anyone had any other insight on what I might feed the GUI to get my desired results?

Cheers & thx!

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