@sipman513554 wrote:

Hello All,

I have a FreePBX running
PBX Firmware:10.13.66-17
PBX Service Pack:1.0.0.0

The SIP registrations on the FreePBX are
Switchvox-->(pjsip)Registers To-->>FreePBX<<--Registers To(pjsip)<--Digium ISDN Gateway

FreePBX is only used as a pass-through server

FreePBX has 2 X Incoming routes

DID CID Description Destination
54xx Any Switchvoxto_Gateway Trunks: Gateway (pjsip)
55xx Any GatewayTo_Switchvox Trunks: switchvox (pjsip)

5401 is a SIP endpoint on the ISDN Gateway
5520 is a SIP endpoint on the Switchvox

When 5401 calls 5520 FreePBX sees the INVITE which has the FROM and the TO fields and passes the call through without issues
When 5520 calls 5401 if the Switchvox sends the Caller ID " FROM " as 5520@IP FreePBX does not process the call, i.e there are no invites sent out,However when the CID is hidden as " anonymous" by disabling CID on the Switchvox trunk where FROM=Anonymouns@IP, FreePBX routes the call through to 5401 without issues and 5401 sees the caller as ANONYMOUS.
So basically FreePBX routes through when the CID is anonymous and doesnt route when the CID is a number.

I have tried to set the destination on the Inbound Route to SET CID and then passthrough to the trunk and have had no luck.
I have tried to modify the incoming route to DID= " 54XX" and CID = "55xx" and have had no luck.
I have changed the registration method(Peer) between FreePBX and Switchvox and have had no luck
I am not sure if I can use an Outgoing route in this Scenario,If Yes, How ?

Please help !

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@claloano wrote:

The GUI of my FREEPBX by problems

I do not know why, but crying out the web interface /admin/config.php no longer starts

In apache error log I found this:

[Tue Jul 18 11:24:13.752912 2017] [mpm_prefork:notice] [pid 4411] AH00163: Apache/2.4.10 (Raspbian) configured -- resuming normal operations
[Tue Jul 18 11:24:13.753108 2017] [core:notice] [pid 4411] AH00094: Command line: '/usr/sbin/apache2'
[Tue Jul 18 11:27:08.822030 2017] [mpm_prefork:error] [pid 4411] AH00161: server reached MaxRequestWorkers setting, consider raising the MaxRequestWorkers setting
[Tue Jul 18 11:27:08.860151 2017] [:error] [pid 4415] [client 192.168.1.111:50988] PHP Warning: file_put_contents(/var/log/asterisk/freepbx_debug): failed to open stream: Permission denied in /var/www/html/admin/libraries/utility.functions.php on line 656
[Tue Jul 18 11:27:08.860277 2017] [:error] [pid 4415] [client 192.168.1.111:50988] PHP Notice: Undefined index: PHP_CONSOLE in /var/www/html/admin/libraries/utility.functions.php on line 659
[Tue Jul 18 11:27:08.860439 2017] [:error] [pid 4415] [client 192.168.1.111:50988] PHP Warning: file_put_contents(/var/log/asterisk/freepbx_debug): failed to open stream: Permission denied in /var/www/html/admin/libraries/utility.functions.php on line 656
[Tue Jul 18 11:27:08.860478 2017] [:error] [pid 4415] [client 192.168.1.111:50988] PHP Notice: Undefined index: PHP_CONSOLE in /var/www/html/admin/libraries/utility.functions.php on line 659
[Tue Jul 18 11:27:08.860561 2017] [:error] [pid 4415] [client 192.168.1.111:50988] PHP Fatal error: Class 'Whoops\Run' not found in /var/www/html/admin/bootstrap.php on line 119

If someone has a tip you are welcome!

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@prateeekm wrote:

Hi Everyone,

We have been facing an issue when we are trying to take the backup through backup & restore option available in FreePBX. The backup is happening when taken the backup is taken locally i.e. folder /var/spool/asterisk/10.32.11.30/Backup_10_32_11_30/ but when we are mounting /var/spool/asterisk/10.32.11.30/Backup_10_32_11_30/ folder in NFS, it is giving below mentioned errors.

Saving Backup 13...done!
Intializing Backup 13
Backup Lock acquired!
Running pre-backup hooks...
Adding items...
/bin/tar: Removing leading /' from member names
/bin/tar: Removing leading/' from member names
/bin/tar: Removing leading /' from member names
Bulding manifest...
Creating backup...
Storing backup...
/bin/cp: cannot create regular file/var/spool/asterisk/10.32.11.30/Backup_10_32_11_30/20170718-082818-1500380898-884591137.tgz': No such file or directory
Error copying /var/spool/asterisk/tmp/20170718-082818-1500380898-884591137.tgz to /var/spool/asterisk/10.32.11.30/Backup_10_32_11_30/20170718-082818-1500380898-884591137.tgz:
Running post-backup hooks...
Backup completed with errors!

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@thehammer86 wrote:

The following values are present in sip_general_additional.conf in the latest stable FreePBX 14 Asterisk 13 release.

tos_sip=cs3
tos_audio=ef
tos_video=af41

Adding new values to Settings-->SIP Settings-->Chan-SIP does not seem to override these values. Also, a wireshark analysis of the UDP packets reveals they are being sent with a DSCP tag of 0x05. I cannot match this to a known hex value to then translate into a TOS value.

How would one go about properly changing these values assuming that there isn't a bug preventing this?

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@KenyBDG wrote:

Hello,

I have fresh install of FreePBX (STABLESNG7-FPBX-64bit-1706-1) FreePBX 14.0.1.1 and Asterisk 13.
Unfortunetely I can not configure SIP Trunk based on PJSIP
In Asterisk CLI I saw some error related to PJSIP.
Below, dump from CLI:

[2017-07-19 08:54:17] ERROR[9413]: res_sorcery_config.c:230 sorcery_config_internal_load: Unable to load config file 'pjproject.conf'
[2017-07-19 08:54:17] ERROR[6693]: res_pjsip_config_wizard.c:1086 object_type_loaded_observer: Unable to load config file 'pjsip_wizard.conf'
[2017-07-19 08:54:17] ERROR[6693]: res_pjsip_config_wizard.c:1086 object_type_loaded_observer: Unable to load config file 'pjsip_wizard.conf'
[2017-07-19 08:54:17] ERROR[6693]: res_pjsip_config_wizard.c:1086 object_type_loaded_observer: Unable to load config file 'pjsip_wizard.conf'
[2017-07-19 08:54:17] ERROR[6693]: res_pjsip_config_wizard.c:1086 object_type_loaded_observer: Unable to load config file 'pjsip_wizard.conf'
[2017-07-19 08:54:17] NOTICE[6693]: sorcery.c:1406 sorcery_object_load: Type 'system' is not reloadable, maintaining previous values
[2017-07-19 08:54:17] ERROR[6693]: res_pjsip_outbound_registration.c:1219 sip_outbound_registration_regc_alloc: Invalid outbound proxy URI 'neofon.tp.pl' specified on outbound registration '399500028'
[2017-07-19 08:54:17] ERROR[6693]: res_sorcery_config.c:307 sorcery_config_internal_load: Could not create an object of type 'registration' with id '399500028' from configuration file 'pjsip.conf'
[2017-07-19 08:54:17] NOTICE[6693]: res_sorcery_config.c:318 sorcery_config_internal_load: Retaining existing configuration for object of type 'registration' with id '399500028'
[2017-07-19 08:54:17] ERROR[6693]: res_pjsip_config_wizard.c:1086 object_type_loaded_observer: Unable to load config file 'pjsip_wizard.conf'
[2017-07-19 08:54:17] ERROR[9413]: res_pjsip_config_wizard.c:1086 object_type_loaded_observer: Unable to load config file 'pjsip_wizard.conf'
[2017-07-19 08:54:17] WARNING[9413]: iax2/firmware.c:234 iax_firmware_reload: Error opening firmware directory '/var/lib/asterisk/firmware/iax': No such file or directory
[2017-07-19 08:54:17] NOTICE[9413]: iax2/provision.c:562 iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled.
[2017-07-19 08:54:17] ERROR[9413]: res_pjsip_config_wizard.c:1086 object_type_loaded_observer: Unable to load config file 'pjsip_wizard.conf'

If i prepare configuration based on chan_SIP, everything works fine, Asterisk send proper registration to registrar server

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@dgleks wrote:

Greetings folks,
I have successfully installed FreePBX 13 along with Asterisk 13 multiple times over the last year or so, following this guide located at:
https://wiki.freepbx.org/display/FOP/Installing+FreePBX+13+on+Ubuntu+Server+14.04.2+LTS
A few days ago, I found a comment on the bottom of that page which was just what I needed. Modified instructions for installing on Ubuntu 16.04 found here:
https://drive.google.com/file/d/0B8w4ZQV13bCSNDMzNFFmTWVSVVE/edit
Everything went smoothly (with a few glitches along the road that were quickly fixed after researching my issue), until I came to installing FreePBX.
No matter how I run ./install, with or without the -N argument, I get this error: It's a little long, but I figured too much information is better than not enough information.
root@Dglexsoft:/usr/src/freepbx# ./install -n
PHP Warning: Declaration of FreePBX\Install\FreePBXHelpCommand::setCommand(FreePBX\Install\FreePBXInstallCommand $command) should be compatible with Symfony\Component\Console\Command\HelpCommand::setCommand(Symfony\Component\Console\Command\Command $command) in /usr/src/freepbx/installlib/installhelpcommand.class.php on line 15
Assuming you are Database Root
Checking if SELinux is enabled...Its not (good)!
No /etc/asterisk/asterisk.conf file detected. Installing...PHP Warning: Illegal string offset 'directories' in /usr/src/freepbx/amp_conf/htdocs/admin/libraries/BMO/LoadConfig.class.php on line 317
PHP Fatal error: Uncaught Error: Cannot use string offset as an array in /usr/src/freepbx/amp_conf/htdocs/admin/libraries/BMO/LoadConfig.class.php:317
Stack trace:

0 /usr/src/freepbx/amp_conf/htdocs/admin/libraries/BMO/LoadConfig.class.php(81): FreePBX\LoadConfig->explodeConfig(Array)

1 /usr/src/freepbx/amp_conf/htdocs/admin/libraries/BMO/LoadConfig.class.php(41): FreePBX\LoadConfig->loadConfig('asterisk.conf', '/usr/src/freepb...')

2 /usr/src/freepbx/installlib/installer.class.php(47): FreePBX\LoadConfig->__construct('Fake FreePBX Ob...', 'asterisk.conf', '/usr/src/freepb...')

3 /usr/src/freepbx/installlib/installcommand.class.php(217): FreePBX\Install\Installer->asterisk_conf_read('/usr/src/freepb...')

4 /usr/src/freepbx/amp_conf/htdocs/admin/libraries/Composer/vendor/symfony/console/Symfony/Component/Console/Command/Command.php(257): FreePBX\Install\FreePBXInstallCommand->execute(Object(Symfony\Component\Console\Input\ArgvInput), Object(Symfony\Component\Console\Output\ConsoleOutput))

5 /usr/src/freepbx/amp_conf/htdo in /usr/src/freepbx/amp_conf/htdocs/admin/libraries/BMO/LoadConfig.class.php on line 317

I'm not sure what to do. I followed everything in that modified guide exactly, only changing things to point to the proper locations of certain files and whatnot.
Am I doing something wrong? is there something missing in that guide? is that guide totally wrong and should I just toss it in the nearest recycling bin?
I've been using Ubuntu for a few years now, but am still by name a newbie to it, so please bare with me if I don't understand certain things. This goes for programming/coding in PHP/Asterisk/FreePBX as well, which I haven't really done anything at all in.
Also I thought it worth mentioning I'm visually impaired/blind as well, so asking for/posting pictures and/or screenshots wouldn't help. I can, however, copy/paste here certain configuration files or whatnot if needed to trouble shoot the issue.
Thanks so much in advance!

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@Asjas wrote:

Hi Guys,

I just installed both SNG7-FPBX-64bit-1706-1 and 10.13.66-64bit distros on a new system we are configuring.
Neither of these support adding a ring group as destination for our Inbound Route. Has support for it been disabled?

When I install our old 6.12.65 distro it supports adding a ringgroup as a destination. Can't ring groups be used for Incoming Routes anymore or am I missing something really obvious?

Thanks for your help

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@SamTaylor510 wrote:

New to FreePBX. Have been running a Trixbox server for last 8 years. Needless to say hard drive crashed and hardware was too old to revive, so I have move to freepbx, but am having trouble with the install. Downloaded and installed the freepbx stable 64 bit iso. I have changed IP's to static. I have one NIC and it is registering in the dashboard with traffic. I have two internal eyebeam softphones configured and working. I can call between the two of the softphones and all works as it should. The dashboard shows I have two users, and two trunks online. Problem is calls will not come into the system, and when I try to call out I get the message "all circuits are busy now". I have 7 DID numbers from my SIP Trunk Provider that I have used for years. Not sure where these get entered into FreePBX. I have an outbound route set up, and an inbound route set up. Any help would be appreciated. If there are any logs, or other info you need, please let me know. Will get right out to you. Thanks in advance, Sam

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@oper wrote:

i would like to Call a Function every-time a extension:
- Originate a Call
- Hangup phone
- Receive a Call

any idea where i should locate the function call?

like:
exten => s,n,MyFunction('Close')

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@asankag wrote:

Hi Guys,

I need some help in finding the location where the below settings are stored on the MySQL DB or any other location on the FreePBX GUI.

Settings -> Asterisk SIP Settings ->

• General SIP Settings -> External Address
• Chan_SIP Settings -> Override External IP
• Chan_PSIP Settings -> External IP Address

What I am trying to do is to detect the external IP via a script, update these values to the DB before I start configuration of the PBX. Think of this like a PBX initialization script that I would like to run before starting to build the PBX.

The values above gets stored on the following config files.

/etc/asterisk/pjsip.transports.conf: external_media_address=192.168.1.1
/etc/asterisk/pjsip.transports.conf: external_signaling_address=192.168.1.1
/etc/asterisk/sip_general_additional.conf: externip=192.168.1.1

Regards

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@claloano wrote:

Often use of raspberry as FreePBX and obviously use microSD
In this situation from time to time I have corrupted file system
I'm thinking of using f2fs that should decrease filesystem corruption issues
Do you mind?

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@palkhin wrote:

I have one extension that is making calls automatically. I want to when call is connected, redirect that call to operator(another extension) on another extension. Is this possible ?

Thank you

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@issitsupport wrote:

We have a freePBX system distro 13.0.191.11. We have had the system in place for about a year now, we have added about 15 phones to the system recently, and since that change, we have had issues with dropping to toll free numbers after 17:30 to 17:32. The problem is pretty consistent, we have updated the firmware on the phones Aastra/Mitel 6867i as well as increased the RTP timeout to 1500, which I don't think even makes a difference because we have the session-timers=refuse in the trunk. We have troubleshot with wireshark and found that the call hangs up, I am completely baffled by this issue.

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@trustitalia wrote:

Hi everyone,
we have 2 remote PBXs connected with IAX2, we have configured the same queues on the 2 PBXs.
We would like to be able to do agent logon and logoff from a queue, both from local and from remote queues.
Feature code configured to do logon and logoff is 22 and we have an outbound route that send to the remote site every number dialed with 7 prefix. So for example, if I call the 801 queue it works, if I call 7801 the remote queue works, if I try logging in to the queue with a dynamic agent with 22801 it works but if I try logging in to the remote queue with 227801 it doesn't work and it says "Your call can't be done as you desired" (sorry for the translation as my language phones is in italian).
Please could someone explain me how to achieve this or point me in the right direction?
Thanks in advance for the help!

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@Lvalentini wrote:

Hallo,

I need to set up for an incoming DID a special behavior: when a call arrives, it should enter a queue X with 4 static members, let's say extensions A, B, C, D. After 15 seconds, a fifth extension (member/extension E) should start ringing together with the first 4.

The only way I can find to achieve this is to:
1- set a Follow Me on one of the members' extensions with Initial Ring Time set to 15sec, or
2- set a Queue1 , with members A, B, C, D and Max Wait Time 15sec, specifying as a Failover destination a Queue2, with members A, B, C, D,E and Max Wait Time unlimited.

The first solution seems to be more "light" to me, but since the extensions A B C D are members of more queues and only one queue needs the extension E to start ringing after 15 seconds, the first solution does not fit the needs.

Is there a better way to achieve the goal?
I hope that my explanation is clear enough, and that the title of the post is fine.

Thank you to anyone who will answer.
Kind regards
Lucia

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@pcatman wrote:

I'm trying to install FreePBX 14 on a new Mini PC with 32Gb of onboard flash storage as the only storage device. I am getting this error:

The following problem occurred on line 2 of the kickstart file:

Disk "" given in clearpart command does not exist.

Exception AttributeError: "'NoneType' object has no attribute 'undev_unref'" in <bound method Context.del of <pyudev.core.Context object at 0x7f2be71f6cd0>> ignored

From the logs:
anaconda: Running kickstart %%pre script(s)
anaconda.stdout: Running pre-installation scripts
anaconda: Error code 1 running the kickstart script at line 24

Appreciate any assistance on this - thanks!

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@datazero wrote:

hi everyone,
i am totally noob so please forgive me.
I have installed latest stable freepbx distro, Sangoma A200 (2xFXS, 2xFXO) and Sangoma S500 phones.
How can i manually select from which FXO make a call?
Is it possible to program a key on the phones to select each FXO?

Rgrds

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@lucaber wrote:

Hi,
i have a pfsense router directly behind my ISP router. Currently it is not possible to disable NAT on my ISP router.
I have a Telekom Number connected with my FreePBX-System(PJSIP).
When I connect the FreePBX directly to my ISP´s router, audio is working in both directions. (Tested with Echo *43)
I have no port-Forwarding set up.
But when I connect the FreePBX behind my PFSense, audio is only working in one direction(Outgoing).

My settings (PFSense):

firefox_2017-07-24_16-08-33.png1075x595 204 KB

Any ideas?
Thanks in advance.

-Lucaber

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@shivani wrote:

Hello,

We are running FreePBX/Asterisk 11.17.1 on a physical server right now and we would like to move it to a VM on VMWARE cluster. I searched online and could not find anyone confirming if I can use vmware convertor to migrate my FreePBX instance to VM. Is it possible to use VMWARE convertor to do this conversion? If not and if I have to install FreePBX from scratch on a new VM, how do I copy all the extensions, trunks and other configuration from physical installation to virtual machine? Please advise the steps I would need to take to get this done.

I also plan on upgrading the FreePBX to latest version on VMWARE.

Thanks!

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@oscarenzo wrote:

Hi,

I would like to know if is possible assign to one extension a outbound route, i mean that the extension 123 can not use the outbound route B and vice versa, I was checking this module Extension+Routing

But think that now work for my case because in freepbx panel I allowed for one user and block to other, this case:

Extensions:

123 => Outbound route A
124 => Outbound route B

Outbounds
A -> Pattern (dialplan landline spain 9XXXXXXXX)
B -> Pattern (wildcard ".")

When 123 try to call to other destinations that not match with dial pattern of the outbound A, it use the outbound B, and this outbound is not allowed for this extension

Can somebody help me?

Thank you advance.

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