Hello everyone,
I am encountering an issue with outbound calls on my FreePBX system. Whenever I attempt to make an outbound call, I receive the message: “All circuits are busy now.”
2 posts - 2 participants
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Outbound Error: all circuits busy now please try call later
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Managing Voicemail Deletion in Blast Groups on FreePBX
Is it possible to configure FreePBX so that when one member of a blast group deletes a voicemail, it is also removed from the voicemail inboxes of the other group members? We have a situation where multiple people receive the same voicemail, and if one person answers and deletes it, the others still have the message, leading to repeated callbacks. Is there a way to avoid this redundancy by ensuring that once one member deletes the voicemail, it is deleted for the entire group?
1 post - 1 participant
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How to fix Vulnerability HTTP TRACE / TRACK
How to fix Vulnerability HTTP TRACE / TRACK Methods Allowed
2 posts - 2 participants
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Extension registered but does not receive a call
Hello, I need some help. I created the internal server to use between the receptionist’s extension and an Intelbras SS 3540 MF Face access controller. I created extension 1001 for the receptionist’s phone and 1002 for the controller and I created 1003 to put on my machine via the MicroSIP software to carry out tests. The problem is that when I call the controller with extension 1002 it doesn’t receive a call, it rings 3 times and the call drops, however if I call from 1003 (my pc) to 1001 (reception) they work. I’ve done several tests, such as putting extension 1002 on my machine, and it works.
2 posts - 2 participants
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VOXBEAM change and require new things about inbound outbound and P-Asserted-Identity
I getting these details from Voxbeam they need like these things and they block my outbound so guide me how I can resolve this issues.
INVITE sip:00111028779781647@sbc.voxbeam.com should be INVITE sip:001110218779781647@sbc.voxbeam.com you are missing the 1 as we only accept E164 numbers
From: “12245857428” sip:12245857428@64.176.184.158 should be From: “12245857428” sip:+12245857428@64.176.184.158 you are missing the ‘+’ the From: needs to be in +E164 format
There is no P-Asserted-Identity: header please add one, for an example it should be P-Asserted-Identity: “12245857428” sip:+12245857428@64.176.184.158
they show my fault and solution but how i solve i am so upset.
1 post - 1 participant
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Debian source repository (deb-src)
Howdy!
I recently set up a fresh Debian Bookworm system with FreePBX 17 and migrated everything over from an old FreePBX 16 install. Yay! Smooth sailing with the installer script and backup restoration.
What I didn’t check into beforehand was whether or not there was a deb-src repo for all the packages that get installed. I’ve been patching the prior versions of Asterisk and building my own RPMs and was hoping to do the same here.
Was hoping that it’d be as simple as adding deb-src [arch=amd64] http://deb.freepbx.org/freepbx17-prod bookworm main to sources.list, but no dice.
Nothing obvious on the repo there either for a separate directory for such.
I’d really like to make a drop-in replacement – it’d be nice to have all the package building bits!
(i.e. debian/{control,rules,patches/})
Any pointers?
Thanks for your time,
morb
6 posts - 2 participants
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FreePBX 17 Dashboard Error: Unable to configure networking service: systemd-networkd conflict
Networking service configuration cannot be performed concurrently with an active systemd-networkd service. To enable FreePBX for managing your networking configuration, you’ll need to disable systemd-networkd.
FreePBX17, Debian 12 - Installed on Azure - Does anyone know how to fix this?
1 post - 1 participant
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Sip trunk provider without kyc?p
is there any sip trunk provider who provides trunking without documents and prices are cheaper. if there is any please let me know . to get company documents is very hard for a startup sometimes
1 post - 1 participant
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Inbound call could it be rerouted outside?
Hello…
As per clients needs…:
inbound calls for a specific DID are requested to be routed to a couple of cell phones outside
I have tried :
- FollowME
- Forward inconditional (destination number#, confirmation call)
- Custom Destination (from-internal)
No luck yet… Any ideas?
any idea ?
5 posts - 3 participants
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No Audio when call gets forwarded to a cell phone
Hi, I am wondering if someone can please point me in the right direction. I’ve used this Distro on DigitalOcean many times without any issues, however, I’ve tried installing it on another DO droplet, in fact, 2 separate ones and keep getting the same issue. I set up the inbound route to send the call to a cell phone. The call gets forwarded but, without any sound, and I have not been able to find the reason. I have searched and practically followed every instruction I’ve found here and still can’t get it to properly work. This is the Distro I’m using: FreePBX® | DigitalOcean Marketplace 1-Click App. Thanks in advance.
2 posts - 2 participants
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Paging / Intercom not auto answering Cisco 7975
Hi,
I’ve created my paging group and added the 3 extensions I want to include in my paging/intercom group. When I ring the group the phones ring but don’t auto-answer?
Is there something I’m missing.
Regards.
1 post - 1 participant
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Registering as a SIP station to another PBX
I have a FreePBX instance that is taking care of hosting multiple SIP speakers in a school for paging/intercom. I would like to connect this FreePBX system to an on prem VoIP system the school uses for their normal phones. The simplest way to do this would be to light up a trunk between the systems but that occurs a large license cost on the side of the school district. I have been asked if I can instead register as a SIP ‘station’ to the phone VoIP server.
From some googling, it seems possible to configure a ‘SIP trunk’ to register as a SIP station, but its a little out in the weeds past what I really understand. Anyone able to shed a little light? Thank you!
2 posts - 2 participants
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WebRTC with FreePBX
NOOB Alert!
My apologies in advance. I’m a well seasoned PBX guy. Been working in the SIP world long enough to get by but still have a lot of learning to do.
This is my first exposure to FreePBX, although it is similar to the Switchvox which I can make do just about anything it is capable of.
I am attempting to register a WebRTC client to the FreePBX. For now, I am using the sipML5 client provided by Doubango. That client can be found at sipML5 - The world's first open source HTML5 SIP client
I have read 2 separate articles and followed their FreePBX configuration instructions, step-by-step but am not able to get the sipML5 tool to register to the FreePBX.
Has anyone ever successfully done this? If so, I would sure appreciate your guidance.
Thank you!
Louis
5 posts - 3 participants
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Installing FreePBX on Asterisk 22 using install script?
When running
sng_freepbx_debian_install.sh --noasterisk
on a system with Asterisk 22 installed I get the following:
2024-09-09 15:27:14 - Error at line: 1130 exiting with code 255 (last command was: fwconsole reload >> $log)
2024-09-09 15:27:14 - Exiting script
root@FreePBX-17-Asterisk-22-CM-Patch:/tmp#
the log says:
2024-09-09 15:27:12 - Reloading and restarting FreePBX 17
Reload Started
In Self_Helper.class.php line 214:
Unable to locate the FreePBX BMO Class 'Cdr'A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma
install cdr 2) fwconsole ma enable cdr
root@FreePBX-17-Asterisk-22-CM-Patch:/usr/src/asterisk-22# fwconsole ma install cdr
Detected Missing Dependency of: framework 17.0.1
Found local Dependency of: framework 17.0.19.9
Installing Missing Dependency of: framework 17.0.1
Updating tables admin, ampusers, cronmanager, featurecodes, freepbx_log, freepbx_settings, globals, module_xml, modules, notifications, cron_jobs...Done
<error>Error!</error>
<error>Unsupported Version of 22 </error>
<error>Supported Asterisk versions: 18, 19, 20, 21</error>
Given that Asterisk 22 is supposed to be out in less than a month - and it looks pretty much the same as Asterisk 21 - is there a plan to change the “supported version check” in all of the FreePBX modules to include the number 22 sometime soon?
Just askin! 
2 posts - 2 participants
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Keep CallerID after being forwared from queue
Hello all,
When we answer a call form a queue and then forward this call to an extention the number of the agent is being displayed.
Is it possible to keep/retain the orgiginal CallerID after being forwarded from a queue?
1 post - 1 participant
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Audio being received but not sent?
Hi all,
I’m relatively new to VOIP so sorry for in advance for what is hopefully a simple question…
I’ve got a basic FreePBX V17 install going on a Pi. At the moment I’m just trying to get internal calling working - I haven’t configured anything to interface it with the outside world so to speak.
I’ve made 2 extensions and am using MicroSIP as my client. The problem I’m having is that I can’t hear any audio on the other end when I talk. I can hear audio coming from Asterisk - E.G. when the call goes to voicemail I hear the greeting, but no audio appears to being sent - only received. Voicemail hangs up while I record a message, presumably because of some kind of silence detection functionality.
All devices are on the same network. I’ve verified my mic settings on the clients are all correct (I’ve tried multiple computers). I also have an old ATA (Linksys PAP2) which works with my current provider, but experiences the same issue as MicroSIP when I reconfigure it to point to my FreePBX install.
Does anyone have any pointers for things I could try?
Many thanks,
Ben.
2 posts - 1 participant
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Problem with two digit ivr entries.
Hello,
my ivr receives always only one digit. No matter how I set up Force Strict Dial Timeout (Yes or No or No - Legacy) in logs i find:
app_read.c: User entered ‘2’
Even when this first digit leads to second ivr this second ivr doesn’t receive any digit (although I send digits from calling device) and ends with timeout (always).
I tried to call from mobile phone and another VoIP provider.
How to check why ony one digit is recognised by pbx and all subsequent digits are droped/ignored?
1 post - 1 participant
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Can't Install UCP on FPB 17 on Raspi5
Hi,
I’m trying to install UCP on FreePBX 17.0.19.9, running on a Raspberry Pi 5.
Whenever I try to install it via commandline (fwconsole ma downloadinstall ucp) i get:
In Pm2.class.php line 402: Undefined variable $log
When trying to install it via Module Admin I get the same Error Message
Is there any obvious solution to this or any other idea I could follow?
Thanks in advance
2 posts - 2 participants
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Freepbx version 16 upgrade fails
I upgraded from 14 to 15 with no problem and restarted it and then tried to go from 15 to 16. The cli shows this.
~]# fwconsole ma upgradeal
Exception: SQLSTATE [HY000] [2002] No such file or directory: : SQLSTATE [HY000] [2002] No such file or directory in file /var/www/html/admin/libraries/utility. functions. php on line 120
Caused by
PDOException: SQLSTATE [HY000] [2002] No such file or directory in file /var/www/html/admin/libraries/BMO/Database. class.php on line 144
Stack trace:
- Exception->() /var/www/html/admin/libraries/utility. functions.php:120
- die_freepbx () /var/www/html/admin/libraries/BMO/Database.class.php:150
- PDOException->() /var/www/html/admin/libraries/BMO/Database. class.php:144
- PDO->_construct () /var/www/html/admin/libraries/BMO/Database. class.php:144
- FreePBX\Database-> construct () /var/www/html/admin/libraries/BMO/FreePBX.class.php:77
- FreePBX-> construct () /var/www/html/admin/bootstrap.php:144
- require_once () /etc/freepbx.conf:11
B. include_once () /var/lib/asterisk/bin/fwconsole:12
I can’t access the web gui either, I get a This Page isn’t working. pbxfqdn is currently unable to handle this request. HTTP ERROR 500.
php -v shows PHP 7.4.16
What can I do to fix this?
6 posts - 2 participants
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External Extension Hangs Up
Hello All, i have the FreePBX 17 running on local server behind NAT and local extensions work perfectly but external extensions seems to work right for a few minutes and then hang up call after 5/6 minutes randomly. Do you have any idea of what’s happening?
1 post - 1 participant
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