@AtwellPBX wrote:

I’m new to the phone system business. I have a PBX setup working just fine. I wanted to know if there is a way to restrict cellular calls on specific extensions, any help would be appreciated.

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@sinh2000 wrote:

Yesterday morning, we received the following notification from our FreePBX system:

"SECURITY NOTICE:

Updated Certificates:
Some SSL/TLS Certificates have been automatically updated. You may need to
ensure all services have the correctly update certificate by restarting PBX
services"

Seeing this, I tried to run “core restart gracefully”. It hung (would not complete) and I had to restart the box (CentOS).

Since then, I am constantly restarting Asterisk. It continues to be unresponsive. Once I restart it will work for anywhere between 5 minutes and 2 hours.

I am seeing some errors in /var/log/asterisk/full and /var/log/asterisk/freepbx.log but there are no errors when asterisk hangs and stops working. The errors I am seeing in “full” happen before the system hangs so not sure if they’re related or not. I see them when doing a fwconsole restart:

[2019-05-09 08:54:38] ERROR[14837] config_options.c: Unable to load config file ‘acl.conf’
[2019-05-09 08:54:38] NOTICE[14837] cdr.c: CDR simple logging enabled.
[2019-05-09 08:54:38] NOTICE[14837] loader.c: 318 modules will be loaded.
[2019-05-09 08:54:39] ERROR[14837] config_options.c: Unable to load config file ‘statsd.conf’
[2019-05-09 08:54:39] ERROR[14837] res_sorcery_config.c: Unable to load config file ‘codecs.conf’
[2019-05-09 08:54:39] ERROR[14837] res_sorcery_config.c: Unable to load config file ‘pjproject.conf’
[2019-05-09 08:54:39] ERROR[14837] res_pjsip_config_wizard.c: Unable to load config file ‘pjsip_wizard.conf’
[2019-05-09 08:54:39] ERROR[14837] res_pjsip_config_wizard.c: Unable to load config file ‘pjsip_wizard.conf’
[2019-05-09 08:54:39] ERROR[14837] res_pjsip_config_wizard.c: Unable to load config file ‘pjsip_wizard.conf’
[2019-05-09 08:54:39] NOTICE[14837] res_smdi.c: Unable to load config smdi.conf: SMDI disabled
[2019-05-09 08:54:39] NOTICE[14837] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener.
[2019-05-09 08:54:39] WARNING[14837] res_stun_monitor.c: Unable to load config res_stun_monitor.conf
[2019-05-09 08:54:39] ERROR[14837] config_options.c: Unable to load config file ‘hep.conf’
[2019-05-09 08:54:39] ERROR[14837] res_calendar.c: Unable to load config calendar.conf
[2019-05-09 08:54:39] ERROR[14837] pbx_lua.c: Error loading extensions.lua: cannot open ‘/etc/asterisk/extensions.lua’ for reading: No such file or directory

The only error I see in freepbx.log is this:

[CRITICAL] (admin/bootstrap.php:260) - Connection attmempt to AMI failed

Any help would be appreciated. I’m not sure where to go with this. We installed FreePBX about 6 months ago and it’s been working great until now.

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@charlesc83 wrote:

Is there a way to exclude certain modules? I have one module that errors every time I try to run the conversion even if I uninstall it from the donor and new box. Or is there a way to see what the error is?

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@slip_cougan wrote:

So while trying to figure out why my voicemail email notifications were sometimes taking a while to be delivered I had a go at disabling ipv6 - big mistake.

Google can be dangerous sometimes!
I added the following lines to /etc/sysctl.conf:

net.ipv6.conf.all.disable_ipv6 = 1
net.ipv6.conf.default.disable_ipv6 = 1

On restart I’m now getting “unknown output from mailq” in the Dashboard.

I undid the changes and still get the same error.

So now I checked /etc/exim4/exim4.conf.template
Under MAIN CONFIGURATION SETTINGS
I see that ‘queue_list_requires_admin = false’ is there.
I ran update-exim4.conf anyway as I’d added this line and got an error stating it was duplicated.

Not sure why the error won’t clear but it’s driving the OCD in me nuts.

On restarting I get emails from Fail2Bin, and if I leave voicemails I do get the email notifications.
Can anyone help me with my OCD and help get rid of that annoying red warning?

Thanks
-s

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@AtwellPBX wrote:

So I am using I am using a GrandStream gateway( GXW410X) for 2 traditional lines I use with FreePBX. These two lines are set to roll over if one line is busy (I believe this is called hunt group) but it doesn’t work that way when using FreePBX if someone else were to call while the line is busy it wouldn’t roll over it would say that the line is out of services so how do I go about having the “hunt group” functionality to be in sync with FreePBX.

I’m using FreePBX 14.0.5.25

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@Wolfie wrote:

Hello all,

I’m new here, I’ve been trying to set up FreePBX with Twilio(with Trial number from USA but I’m from Costa Rica) and Xlite.

I’m able to receive and make calls but sometimes it does not work and or there is a delay on the dial tone and then it works ( from twilio I can see Warning - 32011 Error communicating with your SIP communications infrastructure, it shows when there is a delay and also when does not work).
From Xlite when I get a call, I hit answer but it does nothing until the third time that rings.

Any recommendations or settings to check?

Thanks.

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@herbw wrote:

After trying and failing to configure a Cisco SPA514G manually, I gave up and purchased the Endpoint Manager. However, I am still unable to configure this phone. I’ve made several attempts.

First, I reset the phone to factory defaults.

Then, I connect to the phone using a web browser, and configure the User ID (202), the Password (The Secret for Extension 202), the Display Name, the SIP Port (5160), and the Proxy (the IP Address of my server). When I save the changes, the phone reboots.

After rebooting, the phone displays the correct extension (202), but when I press the Line 1 button, the phone displays either “Line 1 202 Failed (No Response)”, “Line 1 202 Not Registered (No Response)”, or “Line 1 202 Failed Authenticate”

For what it’s worth, I have successfully configured a Snom 821, so I believe my FreePBX is working.

I really didn’t need the Endpoint Manager if it’s not going to make it any easier to get my phones configured.

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@dan_ce wrote:

Is it possible to do the above?

Either to a schedule (like wake up calls) or based on a command line command (like how you can do with ‘call flow control’)

Use case: For an elderly member of the household to be able to use a physical push button to send an MQTT command to the host Pi that causes an extension to ring. When the extension is answered, it begins dialling a preset number.

On Apr 27, 12:34 PM @lgaetz replied:

Pretty much every single click-to-dial utility works this way. You ORIGINATE a call to the local extension which then bridges to another channel on answer.

Sadly with the best will in the world I’ve not been able to find a way to do this.

Basically I want to be able to (for example) via a Node-Red MQTT incoming message from another device on my LAN, cause my kitchen extension to ring. When the extension is answered, I’d like it to immediately being dialling an EXTERNAL (but could be internal) number.

Is that what @lgaetz was referring to? Is there a module to make this simpler?

THANKS

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@moodinsk wrote:

Hi,

I wanted to test the upgrade module on my test box. Installed the updater and it passed all the prereqs.

Started the upgrade and let it run. I came back to it saying i needed to drop to command line and run “fwconsole ma updateall”

When I try running any fwconsole command, i get this:

[root@demo ~]# fwconsole ma list
PHP Warning: require(/var/www/html/admin/libraries/Composer/vendor/composer/…/nikic/fast-route/src/functions.php): failed to open stream: No such file or directory in /var/www/html/admin/libraries/Composer/vendor/composer/autoload_real.php on line 66
PHP Fatal error: require(): Failed opening required ‘/var/www/html/admin/libraries/Composer/vendor/composer/…/nikic/fast-route/src/functions.php’ (include_path=’.:/usr/share/pear:/usr/share/php’) in /var/www/html/admin/libraries/Composer/vendor/composer/autoload_real.php on line 66

I’ve tried updating all packages via yum (everything is up to date now), rebooting, etc. GUI is hosed, asterisk doesnt start, fwconsole is unusable.

Any ideas?

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@slip_cougan wrote:

Good evening community,

I do like these phones but their documentation really is the pits.

I’m trying to invoke the minibrowser to display the video from my fixed CCTV for the door intercom.

I understand the Snom principles and can display video on the phone home screen using the function keys with an ActionURL of:
http://192.168.2.251/live/2/mjpeg.jpg#mjpg
Which does display the camera feed.

I also tried using the same stream request in the Directory entry ActionURL for the door intercom (Ext 3001). The idea being that when 3001 rings the phone, it would display the video and when the phone is back on hook I issue #mjpg_stop to turn off the stream. This does not work because once the phone rings, focus moves away from the home screen so it’s not visible. It also shows a completely blank screen when the extension is ringing. (No caller info)

So apparently the trick is to invoke the minibrowser.
I’ve set up a test function key (P3) with the following ActionURL:
http://192.168.1.254:80/video.xml - this is my webserver, also tried without :80

Here is the video.xml . on the server:

video.xml.jpg700×239 46.2 KB

Apologies it’s an image but trying to paste xml to this site re-formats the text.
This is pretty much a copy/paste from Snom’s Wiki example.

When I run the ActionURL, I get the following errors in the Snom log:

May 10 17:26:50 [NOTICE] PHN: Fetching URL: http://192.168.1.254:80/video.xml?.…
May 10 17:26:50 [INFO ] PHN: Xpath applies: /SnomIPPhoneBatch/
May 10 17:26:50 [INFO ] PHN: Xpath contains unrecognized steps, aborting
May 10 17:26:50 [NOTICE] PHN: Fetching URL: http://192.168.2.251:80/live/2/mjpeg.jpg?.…
May 10 17:26:50 [INFO ] GUI: synth_silent: connected lines: 0 of 0 state: Minibrowser
May 10 17:26:51 [NOTICE] PHN: Fetching URL: http://192.168.1.254:80/video.xml?.…
May 10 17:26:51 [INFO ] PHN: Xpath applies: /SnomIPPhoneBatch/
May 10 17:26:51 [INFO ] PHN: Xpath contains unrecognized steps, aborting
May 10 17:26:51 [NOTICE] PHN: Fetching URL: http://192.168.2.251:80/live/2/mjpeg.jpg?.…
May 10 17:26:51 [NOTICE] PHN: Fetching URL: http://192.168.1.254:80/video.xml?.…
May 10 17:26:51 [INFO ] PHN: Xpath applies: /SnomIPPhoneBatch/
May 10 17:26:51 [INFO ] PHN: Xpath contains unrecognized steps, aborting
May 10 17:26:51 [NOTICE] PHN: Fetching URL: http://192.168.2.251:80/live/2/mjpeg.jpg?.…
May 10 17:26:55 [NOTICE] PHN: Fetching URL: http://192.168.1.254:80/video.xml?.…
May 10 17:26:55 [NOTICE] PHN: TPL: Socket 961 idle/connect timeout
May 10 17:26:56 [INFO ] PHN: Xpath applies: /SnomIPPhoneBatch/
May 10 17:26:56 [INFO ] PHN: Xpath contains unrecognized steps, aborting
May 10 17:26:56 [NOTICE] PHN: Fetching URL: http://192.168.2.251:80/live/2/mjpeg.jpg?.…

No video is shown, the labels for the softkeys disappear until I kill the command with the Esc button.
Can anyone see why this is not working?
Also the phone gets very sluggish if I leave this running.

Once I get this working I can move the ActionURL back into the the Directory Ext 3001 ActionURL field.
Hopefully it will then trigger the minibrowser when 3001 rings in.

Cheers
-s

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@AtwellPBX wrote:

At the moment If someone were to call the line the secretary would pick up the and transfer the call to which ever department the caller needs but when she transfers the call and no one from the department picks up the call it just terminates.

How do you get that transferred call (if not answered) to return to the secretary’s phone so she can let the customer know that there is no one at the desk.

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@herbw wrote:

Using freepbx 14, I have installed two extensions (201 and 202). Each phone displays its own extension. I get a dial tone on both of them. When 202 calls 201, the phone rings, but upon answering, there is no audio in either direction. When 201 calls 202, a call timer begins, but 202 does not ring. When I call *97 (voicemail) from either extension, a call timer starts (showing My Voicemail), but there is no audio, and I am not prompted to enter my password.

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@rlyoxthimer wrote:

I have a situation where calls are failing to the SIP provider due to extra alert_info headers when using Follow Me. When an external party calls an extension’s DID where follow me is active, the call to the user’s mobile works fine. When an internal extension calls another extension with follow me active, FreePBX is placing extra ALERT_INFO headers in the INVITE and the carrier is rejecting the call (400 Bad Request) to the user’s mobile.

Below are captures of the PBX Invites to the carrier to dial the mobile follow me number. The first is acall originated from an external party, the second is what happens when its extension-to-extension.

INVITE that is sucessful (carrier completes the call)

Internet Protocol Version 4, Src: 10.31.254.1, Dst: 10.31.254.2
User Datagram Protocol, Src Port: 5060, Dst Port: 5060
Session Initiation Protocol (INVITE)
    Request-Line: INVITE sip:914xxxxxxx@10.31.254.2 SIP/2.0
    Message Header
        Via: SIP/2.0/UDP 10.31.254.1:5060;branch=z9hG4bK4d9e83fa
        Max-Forwards: 70
        From: <sip:username@10.31.254.2>;tag=as67cb89dd
        To: <sip:914xxxxxxx@10.31.254.2>
        Contact: <sip:username@10.31.254.1:5060>
        Call-ID: 0934d225772fcd5506554f7b7536048d@10.31.254.2
        CSeq: 102 INVITE
        User-Agent: FPBX-13.0.194.10(13.15.0)
        Date: Fri, 10 May 2019 22:07:46 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
        Supported: replaces, timer
        Remote-Party-ID: "513xxxxxxx" <sip:513xxxxxxx@10.31.254.2>;party=calling;privacy=off;screen=no
        Content-Type: application/sdp
        Content-Length: 326
    Message Body

INVITE that fails, carrier rejects this invite with 400 Bad Request:

Internet Protocol Version 4, Src: 10.31.254.1, Dst: 10.31.254.2
User Datagram Protocol, Src Port: 5060, Dst Port: 5060
Session Initiation Protocol (INVITE)
    Request-Line: INVITE sip:914xxxxxxx@10.31.254.2 SIP/2.0
    Message Header
        Via: SIP/2.0/UDP 10.31.254.1:5060;branch=z9hG4bK71df91b7
        Max-Forwards: 70
        From: <sip:username@10.31.254.2>;tag=as1e6ab0c0
        To: <sip:914xxxxxxx@10.31.254.2>
        Contact: <sip:username@10.31.254.1:5060>
        Call-ID: 48a7786a165f31ef078f17346561af27@10.31.254.2
        CSeq: 102 INVITE
        User-Agent: FPBX-13.0.194.10(13.15.0)
        Date: Fri, 10 May 2019 22:39:36 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
        Supported: replaces, timer
        Alert-Info: ALERT_INFO:<answer>
        Alert-Info: ALERT_INFO:<answer>
        Alert-Info: ALERT_INFO:<answer>
        Remote-Party-ID: "513xxxxxxx" <sip513xxxxxxx@10.31.254.2>;party=calling;privacy=off;screen=no
        Content-Type: application/sdp
        Content-Length: 328

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@herbw wrote:

Can anyone recommend a FXO (Foreign Exchange Office) Interfaces that can be used to capture the distinctive ring pattern of an incoming phone call on a POTS line and use that to route the call differently in freepbx based upon the incoming ring pattern?

Alternatively, if I capture the incoming distinctive ring pattern using a separate device (An old US Robotics modem does a good job at this task), is there any way to input this information into freepbx as a mechanism to route the call? I would be happy to write software to implement such an interface if there is a clean way to do it.

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@therobust wrote:

Hi,

I recently upgrade my fpbx 13 machine to fpbx14 and after everything was done machine rebooted and webGUI shows “Can not connect to asterisk”

When i try to run fwconsole restart i get following error:

Starting Asterisk…
[--------------->------------] 1 min
In Start.class.php line 187:

_ Unable to connect to Asterisk. Did it start?_

restart [-i|–immediate] [–] []…

but i can run ‘asterisk -r’ command and it takes me to asterisk console where i see following errors :

[2019-05-11 05:09:25] NOTICE[10585]: manager.c:3456 authenticate: 127.0.0.1 failed to authenticate as ‘admin’
[2019-05-11 05:09:27] NOTICE[10596]: manager.c:3456 authenticate: 127.0.0.1 failed to authenticate as ‘admin’
[2019-05-11 05:09:28] NOTICE[10603]: manager.c:3456 authenticate: 127.0.0.1 failed to authenticate as ‘admin’
[2019-05-11 05:09:29] NOTICE[10610]: manager.c:3456 authenticate: 127.0.0.1 failed to authenticate as ‘admin’
[2019-05-11 05:09:30] NOTICE[10616]: manager.c:3456 authenticate: 127.0.0.1 failed to authenticate as ‘admin’
[2019-05-11 05:09:32] NOTICE[10622]: manager.c:3456 authenticate: 127.0.0.1 failed to authenticate as ‘admin’
[2019-05-11 05:09:33] NOTICE[10628]: manager.c:3456 authenticate: 127.0.0.1 failed to authenticate as ‘admin’

Kindly help here.

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@TheN00bBuilder wrote:

Hi folks,

I just put together a FreePBX box with VoIPVoIP services to learn CLI Asterisk. So far it works great for outbound calls, but I just got my DID today and it gives me this error in the Asterisk log and gives a “number unavailable” message. I’m trying to get the full call output but piping it to a file makes it kind of messy, any idea on how I can do that? I already tried turning off allow_sip_guests and a lot of other solutions on here. I think it’s a VoIPVoIP config issue but they provide no support for Asterisk, so I am not entirely sure what to do.

Thanks!

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@kr0490 wrote:

I have been trying to figure out what happened to my freepbx system and its connection to flowroute that is now causing me to get “all circuits are busy now” from my PBX when i try to dial out. I run the ‘pjsip show registrations command’ in asterisk cli and shows my trunk is registered.

I see this error in the asterisk log
[2019-05-12 18:45:29] ERROR[58265] res_pjsip.c: Endpoint ‘FlowrouteTRUNK’: Could not create dialog to invalid URI ‘FlowrouteTRUNK’. Is endpoint registered and reachable?
[2019-05-12 18:45:29] ERROR[58265] chan_pjsip.c: Failed to create outgoing session to endpoint ‘FlowrouteTRUNK’

Any ideas for things to check?

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@azon2111 wrote:

I am getting these errors updating system updates… thoughts? Thank you

Error: Package: glibc-common-2.17-222.el7.x86_64 (@sng-base)
Requires: glibc = 2.17-222.el7
Removing: glibc-2.17-222.el7.x86_64 (@sng-base)
glibc = 2.17-222.el7
Updated By: glibc-2.17-260.el7_6.4.x86_64 (sng-updates)
glibc = 2.17-260.el7_6.4
Available: glibc-2.17-260.el7.i686 (sng-base)
glibc = 2.17-260.el7
Available: glibc-2.17-260.el7_6.3.i686 (sng-updates)
glibc = 2.17-260.el7_6.3
You could try using --skip-broken to work around the problem

** Found 7 pre-existing rpmdb problem(s), ‘yum check’ output follows:
32:bind-license-9.9.4-73.el7_6.noarch is a duplicate with 32:bind-license-9.9.4-61.el7_5.1.noarch
glibc-common-2.17-260.el7_6.4.x86_64 is a duplicate with glibc-common-2.17-222.el7.x86_64
glibc-common-2.17-260.el7_6.4.x86_64 has missing requires of glibc = (‘0’, ‘2.17’, ‘260.el7_6.4’)
1:grub2-common-2.02-0.76.el7.centos.1.noarch is a duplicate with 1:grub2-common-2.02-0.65.el7.centos.2.noarch

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@YoniNVG wrote:

I’m working with PBXact, Asterisk version 16.3.0.
All extensions are Chan Sip and working without any issues.
Last few days im trying to set new extension to work with two phones so i had to set this extension using pjsip and change the contact from 1 to 2 to get the registration for two phones available.
Now the first phone looks like its working and registered BUT when i try to make a call out or in to that extension call go thru but after one sec the phone is rebooting itself.

I have no idea how to try and find the issue, any advise will be appreciated.
I have looked in the ssh using the asterisk -rvvvvvv but i cant see anything but the call info flowing there. is there any other place/command that can help me find the issue?

Thanks guys.

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@Hawkeye wrote:

PBX Version: 12.7.4-1804-2.sng7
FreePBX Version: 14.0.11
After what appeared to be successful conversion using convert.freepbx.org on a brand new FreePBX 14 in order to convert a production or DONOR FreePBX 13 is having some issues.

When trying to run ‘Apply Config’ button, it produces the following error on the NEW FreePBX 14 Server:

Reload failed because retrieve_conf encountered an error

exit: 1
Unable to continue. Invalid argument supplied for foreach() in /var/www/html/admin/modules/core/functions.inc.php on line 4099
#0 /var/www/html/admin/modules/core/functions.inc.php(4099): Whoops\Run->handleError(2, ‘Invalid argumen…’, ‘/var/www/html/a…’, 4099, Array) #1 /var/www/html/admin/modules/core/functions.inc.php(43): core_devices_get_user_mappings()
#2 /var/www/html/admin/modules/core/functions.inc.php(460): core_conf->map_dev_user(‘101’, ‘mailbox’, ‘101@device’)
#3 /var/www/html/admin/modules/core/functions.inc.php(104): core_conf->generate_sip_additional(‘13.19.1’)
#4 /var/www/html/admin/libraries/BMO/FileHooks.class.php(65): core_conf->generateConf(‘sip_additional…’)
#5 /var/www/html/admin/libraries/BMO/FileHooks.class.php(24): FreePBX\FileHooks->processOldHooks(Array)
#6 /var/lib/asterisk/bin/retrieve_conf(877): FreePBX\FileHooks->processFileHooks(Array)
#7 {main}

All the extensions on the NEW FreePBX are virtual extensions. On the production or Donor box, there are 0 virtual extensions. As a result there are no voice mail settings in the new FreePBX 14 Server for any extension.

Anyone have a solution for this?
thanks.

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