@bksales wrote:

This is a new one for me. They want to do their annual meeting virtually and want to use the PBX to have everyone be able to call in and listen while muted. Not sure what the practical and theoretical limits on the PBX are or what the best solution would be. I know we used the broadcast module last week to do 90 concurrent calls and that seemed to work fine, but I’m nervous about trying 1000 concurrent calls let alone 1000 members of a conference bridge.

Any ideas would be appreciated.

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@mlaihk wrote:

TLS1.0 and TLS1.1 is going to be rated insecure coming Jan 2020. Is it possible to disable TLS1.0/TLS1.1 for the Freepbx 15 WebUI ?

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@bksales wrote:

This is a new one for me. They want to do their annual meeting virtually and want to use the PBX to have everyone be able to call in and listen while muted. Not sure what the practical and theoretical limits on the PBX are or what the best solution would be. I know we used the broadcast module last week to do 90 concurrent calls and that seemed to work fine, but I’m nervous about trying 1000 concurrent calls let alone 1000 members of a conference bridge.

Any ideas would be appreciated.

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@swalker2001 wrote:

I am getting an “SSL Connect Error” when attempting to install Allison’s English language pack (same error on any language pack). Sound Languages module is installed. I also removed it via cli and reinstalled it but no change.

This is FreePBX 13.0.197.14 and it is fully up to date. Full text of the error is:

Exception
SSL connect error
File:/var/www/html/admin/modules/soundlang/soundlang.class.php:1277

Not sure what to do. This is a fairly new install and I just need a single announcement set up but can’t do that until I have a language installed.

Any help would be appreciated.

Steve

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@avayax wrote:

With Kari’s law now going into effect, does the CallerID that is sent on a 911 call have to be a number that the emergency operator has to be able to call back and reach the phone the call was made from?

If that is the case I would have to get a DID and E911 for every phone in our organization, which would be very expensive and impractical.
We do have our system notify our security dept on every 911 call with exact location and number of the phone and they rush to the caller immediately.
The emergency callerID that we send on 911 calls is our main company number that goes to an IVR if you call it.

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@becpartners wrote:

Hi,
I have a RasPBX – Asterisk for Raspberry Pi server since 2 years with a very basic configuration, I’m really a noob with PBX related configurations.
I want to put a message for every call between two dates that say that our business is closed, giving our holidays date and wishing a merry christmas, the basic things.
What is the easiest way to do it ?

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@daveric wrote:

I am a user of the latest version of freepbx, 15, with asterisk 16.
For some time now, I have noticed that the calls stop hearing as when the mobile loses coverage per second. I have two teams and in both cases the same. I thought it could be a router problem and it would change without satisfactory result. I thought it could be a problem for the internet operator and what it is with a different one without satisfactory results. Does anyone know what could be happening and how to fix it? It has always worked perfectly. a hug.
Issues of any asterisk update?

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@ITconsultant wrote:

in freepbx GUI when I click on reports and then I click on asterisk info and then on the right I click on peers I can see my sip trunk IP address on 5060 but how can I see what IP address the media is going to

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@shakawkaw wrote:

Hi

Sometimes a module app_queue.so crashes with an error. And I can not find a reason for this behavior. Possible, someone will be able to indicate what I’m doing wrong.

# dmesg -T | grep -P ‘segfault’
[Wed Nov 13 10:33:52 2019] asterisk[18466]: segfault at 10 ip 00007f21859aad9c sp 00007f21707aeb40 error 4 in app_queue.so[7f2185992000+35000]
[Wed Nov 20 15:13:52 2019] asterisk[310]: segfault at 10 ip 00007feff3540d9c sp 00007fefd19b7b40 error 4 in app_queue.so[7feff3528000+35000]
[Thu Nov 21 10:06:08 2019] asterisk[10940]: segfault at 10 ip 00007f9888f67d9c sp 00007f982f9a5b40 error 4 in app_queue.so[7f9888f4f000+35000]

next i’m get backtrace prom core dump

# gdb /usr/sbin/asterisk /tmp/core.AsterPBX.localdomain-2019-11-21T10:06:12+0300 > ~/core.AsterPBX.localdomain-2019-11-21T10-06-12.new
# gdb /usr/sbin/asterisk /tmp/core.AsterPBX.localdomain-2019-11-20T15:13:56+0300 > ~/core.AsterPBX.localdomain-2019-11-21T15-13-56.new
# gdb /usr/sbin/asterisk /tmp/core.AsterPBX.localdomain-2019-11-13T10:33:56+0300 > ~/core.AsterPBX.localdomain-2019-11-13T10-33-56.new

and get error function from backtrace files

# cat ~/core.AsterPBX.localdomain-2019-11-*new | grep -A 3 Program
Program terminated with signal 11, Segmentation fault.
#0 0x00007f21859aad9c in handle_hangup (userdata=userdata@entry=0x7f21400c5a60, sub=sub@entry=0x7f21401e1460, msg=msg@entry=0x7f224407b910) at app_queue.c:6235
6235 ast_queue_log(queue_data->queue->name, caller_snapshot->uniqueid, queue_data->member->membername,
(gdb) quit

Program terminated with signal 11, Segmentation fault.
#0 0x00007feff3540d9c in handle_hangup (userdata=userdata@entry=0x7ff07005dce0, sub=sub@entry=0x7ff0700d36b0, msg=msg@entry=0x7ff040154140) at app_queue.c:6235
6235 ast_queue_log(queue_data->queue->name, caller_snapshot->uniqueid, queue_data->member->membername,
(gdb) quit

Program terminated with signal 11, Segmentation fault.
#0 0x00007f9888f67d9c in handle_hangup (userdata=userdata@entry=0x7f98e00c00f0, sub=sub@entry=0x7f98e0018ec0, msg=msg@entry=0x7f98f805d340) at app_queue.c:6235
6235 ast_queue_log(queue_data->queue->name, caller_snapshot->uniqueid, queue_data->member->membername,
(gdb) quit

I understand that the problem is in the function “handle_hangup”, but what to do next I can not understand.

asterisk 13.19.1
CentOS Linux release 7.5.1804 (Core) 3.10.0-862.2.3.el7.x86_64
FreePBX framework 14.0.13.12

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@comtech wrote:

Hello,

Any help here is critically appreciated. Came in this morning and our FreePBX could not connect to Asterisk.
FreePBX 14.0.13.17 (12.7.6-1910-1.sng7)
Asterisk 14.7.3

Seeing this error when running asterisk -cvvvvv:

FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x29b10a0 (0) at line 335 in update_file_format_info of media_index.c
[2019-12-20 08:10:36] ERROR[4878]: media_index.c:335 update_file_format_info: Excessive refcount 100000 reached on ao2 object 0x29b10a0
[2019-12-20 08:10:36] ERROR[4878]: media_index.c:335 update_file_format_info: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x29b10a0 (0)

[2019-12-20 08:19:12] NOTICE[6720]: acl.c:715 ast_apply_acl: Manager User ACL: Rejecting '127.0.0.1' due to a failure to pass ACL '(BASELINE)'
[2019-12-20 08:19:12] NOTICE[6720]: manager.c:3392 authenticate: 127.0.0.1 failed to pass IP ACL as 'admin'
[2019-12-20 08:19:12] NOTICE[6720]: manager.c:3426 authenticate: 127.0.0.1 failed to authenticate as 'admin'

Not sure what is causing this nothing has been touched, I believe. We need this server operation in 3 hours or we will miss reminding customers of critical appointments.

Tried so far:

- fwconsole ma update all

- yum update

- fwconsole reload

Reloading FreePBX
Error(s) have occured, the following is the retrieve_conf output:
exit: 1

- fwconsole restart

Hangs at Starting Asterisk

- Checked manager.conf, which we never touch, it has entries for 127.0.0.1/255.255.255.0

    [admin]
    secret = REALLY GOOD ONE
    deny=0.0.0.0/0.0.0.0
    permit=127.0.0.1/255.255.255.0
    read = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message
    write = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message
    writetimeout = 5000

    [admin]
    secret = REALLY GOOD ONE
    deny=0.0.0.0/0.0.0.0
    permit=SERVER STATIC IP/255.255.255.0
    read = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message
    ;write = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message
    ;writetimeout = 5000

Any ideas on what is happening?

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@Hawkeye wrote:

Not sure as to why it’s constantly displaying message about SSL Certificate expiring in < month when the certificate manger clearly displays 2021-01-21 (398 days). This happens on 13 and 14 FreePBX.

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@jerrm wrote:

As a general FYI…

http://mirror.freepbx.org is currently down

Posting here just in case it’s not maintenance and maybe someone can fix.

I tested from a couple of online accessibility checks as well just to make sure it wasn’t something along my path.

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@Kafluke wrote:

We were swapping out switches at one of our locations. Had to shutdown and reboot both node A and node B servers. Both are running FreePBX and are in HA.

When they came back up they were both in standby mode. Had to force node-A out of standby mode and now it won’t boot back to normal at all. The entire httpd folder is empty and here is the message it displays on boot up:

PHP Warning: require_once(/var/www/html/admin/bootstrap.php): failed to open stream: No such file or directory in /etc/freepbx.conf on line 9
PHP Fatal error: require_once(): Failed opening required ‘/var/www/html/admin/bootstrap.php’ (include_path=’.:/usr/share/pear:/usr/share/php’) in /etc/freepbx.conf on line 9
PHP Warning: require_once(/var/www/html/admin/bootstrap.php): failed to open stream: No such file or directory in /etc/freepbx.conf on line 9
PHP Fatal error: require_once(): Failed opening required ‘/var/www/html/admin/bootstrap.php’ (include_path=’.:/usr/share/pear:/usr/share/php’) in /etc/freepbx.conf on line 9
[root@freepbx-a ~]# wget ‘http://git.freepbx.org/projects/FREEPBX/repos/framework/browse/amp_conf/htdocs/admin/functions.inc.php?at=0aea9f8aea6693439a594b88aedc0df48e30b7a4&raw’ -O /var/www/html/admin/functions.inc.php
/var/www/html/admin/functions.inc.php: No such file or directory

Is there any way to recover this PBX without a complete re-install. I do have a config backup but we use iSymphony and I don’t want to have to set that up from scratch.

Is there such a thing as a repair install?

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@xhemp wrote:

Greetings,

Just upgraded my FreePBX from 14 to 15 and backup is playing up, I tried to remove some modules from backup to see if it went through to no avail (in this case userman is not selected for backup but I believe it’s a dependency). Been trying for the last 3 hours with no luck.
This is what I am getting:

Working with userman module
Exporting Databases from userman

In Database.class.php line 212:

  [PDOException (42000)]
  SQLSTATE[42000]: Syntax error or access violation: 1064 You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near '' at line 1


Exception trace:
 () at /var/www/html/admin/libraries/BMO/Database.class.php:212
 PDO->query() at n/a:n/a
 call_user_func_array() at /var/www/html/admin/libraries/BMO/Database.class.php:212
 FreePBX\Database->query() at /var/www/html/admin/modules/backup/BackupBase.php:87
 FreePBX\modules\Backup\BackupBase->dumpDBTables() at /var/www/html/admin/modules/backup/BackupBase.php:132
 FreePBX\modules\Backup\BackupBase->dumpTables() at /var/www/html/admin/modules/userman/Backup.php:6
 FreePBX\modules\Userman\Backup->runBackup() at /var/www/html/admin/modules/backup/Handlers/Backup/Common.php:99
 FreePBX\modules\Backup\Handlers\Backup\Common->processModule() at /var/www/html/admin/modules/backup/Handlers/Backup/Multiple.php:99
 FreePBX\modules\Backup\Handlers\Backup\Multiple->process() at /var/www/html/admin/modules/backup/Console/Backup.class.php:179
 FreePBX\Console\Command\Backup->execute() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Command/Command.php:255
 Symfony\Component\Console\Command\Command->run() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Application.php:960
 Symfony\Component\Console\Application->doRunCommand() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Application.php:255
 Symfony\Component\Console\Application->doRun() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Application.php:148
 Symfony\Component\Console\Application->run() at /var/lib/asterisk/bin/fwconsole:163

backup [--backup BACKUP] [--externbackup EXTERNBACKUP] [--dumpextern DUMPEXTERN] [--transaction TRANSACTION] [--list] [--warmspare] [--implemented] [--filestore FILESTORE] [--restore RESTORE] [--restorelegacycdr] [--modules MODULES] [--restoresingle RESTORESINGLE] [--backupsingle BACKUPSINGLE] [--singlesaveto SINGLESAVETO] [--b64import B64IMPORT] [--fallback]

Also tried update userman and the backup module to the latest edge to no avail. MySQL is running and it’s accessible.
Any idea what could be causing this?
`

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@Razmik wrote:

Hi all,

During several days I am trying to solve this issue, but no luck

I have installed the latest FreePBX Distro on VMware ESXI Hypervisor, and when trying to shut down the VM or reboot, it gets stuck… The only thing I can see is a black screen.

What I have tried:

1. Upgrade VMware ESXI Hypervisor to the latest version, VMware and open-vm tools are also the latest versions
2. Have applied all the possible VM configurations while creating the virtual machine.

The only thing I have figured out is that when I disable the Ethernet it starts working properly.

Any help would be highly appreciated. Thanks!

Best.

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@GSnover wrote:

[2019-11-25 13:54:02] WARNING[96942][C-000002e0]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected ‘=’, expecting $end; Input:
= 1 & 0 = 0
^
[2019-11-25 13:54:02] WARNING[96942][C-000002e0]: ast_expr2.fl:474 ast_yyerror: If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
– Executing [s@macro-outbound-callerid:24] ExecIf(“SIP/239-000005ea”, “?Set(CALLERID(all)=239)”) in new stack
[2019-11-25 13:54:02] WARNING[96942][C-000002e0]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected ‘=’, expecting $end; Input:
= 1 & 0 = 0
^
[2019-11-25 13:54:02] WARNING[96942][C-000002e0]: ast_expr2.fl:474 ast_yyerror: If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables

FreePBX14 box with all modules and updates current - Core 14.28.19. Asterisk 13

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@Autourdupc wrote:

Hi all.

I would like to know if there is a possibility to deactivate a trunk or an outgoing route after a counter of time of outgoing call, and enable it again after a specific reccuring date.

I have a SIM card that permits only 2 hours of free outgoing calls, then after this 2 hours, calls are extra invoiced.
I would like to be able to stop using the SIM card (GSM gateway with chan-dongle) after 2 hours of outgoing calls, then enable it again next month at anniversary date.

Maybe this can be an external script that stop dongle service and restart it after a period.
Any idea how to get the outgoing time counter on a specific trunk ?

Regards,
Laurent.

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@benphone wrote:

i am running a FreePBX 12.0.76.6
asterisk 11
for some reason not all calls are logged to cdr

the calls are logged in queue logs but not in cdr
all of them are abandoned call

looking for a way to debug this i don’t see any errors or anything like this

would appreciate any idea to get to this bug
thanks

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@Inga wrote:

Ive got FreePBX 14 freely installed using ~20 VoIP-phones. It all works perfectly but in week or two (most of the times it’s a week) it stops working. No errors shown in GUI-web interface, the server keeps running, but phones do not work.
It all gets back to normal if I restore a Backup of a newly installed Distro (backed it as soon as it got installed and set up).
So, where do I dig, folks?

It also would help if you could tell me where to find much-much older versions of FreePBX, preferably ver. 2

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@milolethbridge wrote:

Hi all,

Recently, making changes on any of my hosted servers causes all the extensions on that particular instance to show as unavailable in Asterisk Info>Peers. Basically happens as soon as I hit ‘Apply Changes’.

The extensions can still dial out but cannot receive calls.

A reboot of the system brings everything back up, no problem.

Not seen this before and is happening across FreePBX 14 and 15 on both asterisk versions 13 and 16.

I saw a similar thread but it was closed. Is anyone else experiencing this?

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