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Questions tagged [freepbx]

FreePBX is an open-source web-based GUI for Asterisk (PBX), a voice over IP server. On-topic questions include those about writing FreePBX modules, or extending built-in dialplan. Stack Overflow is not an appropriate place for server management or general support of either FreePBX or Asterisk.

FreePBX allows easy management of the Asterisk PBX and automates the creation of complex applications such as IVRs and conferencing systems, as well as management of user extensions and features. It is comprised of various modules which can be independently installed or uninstalled, and an API is provided for third parties to create their own modules.

FreePBX also provides a Linux distribution based on Red Hat Enterprise Linux 7.0, which provides a pre-installed version of FreePBX that is able to run commercially licensed modules.

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FreePBX/Asterisk Recorded Calls not moving to correct location

FreePBX: 10.13.66-12/ISO install Asterisk: 13.12.2 asterisk-addons: Latest Users reported not being able to see/download on demand recordings from the UCP. The calls are however being recorded, /var/spool/asterisk/monitor is full of files, files…
Patrick L
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Asterisk agi not working

I'm having the simple AGI script, when incoming call is answered run script. But does not working. How can i run my script correctly? By the way sorry for my english. Here's code: extensions_additional.conf [macro-auto-blkvm] include =>…
Sergen Kabakçı
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how to handle multiple instance of asterisk in single freepbx for load sharing?

May i know how to handle multiple instance of Asterisk inside single freepbx for load sharing? As of now my system works well with single instance and freepbx. but when load is more i want to share the calls with other instance of asterisk for…
Senthilvel P
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Twilio Origination Call Issue - Intermittent incoming issue

I'm trying to setup Twilio Elastic SIP Trunking on my Asterisk/ Freepbx instance and have struggle setting up a reliable origination (termination works perfectly fine). Background - I've done all settings as per twilio guides, tons of forum posts on…
tipsytopsy
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Transfer call to custom extension to pause recording and dial externally

We have a call centre and record calls using the MixMonitor. We need to be able to transfer some calls externally, but when we do so the call recording continues. This is fine, apart from calls to a specific number. What I've done to try to resolve…
Luke Cousins
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VoIP server/client on Raspberry Pi

I'm concerned with intercom station project based on RPi and playing now around Asterisk Server/FreePBX. I'm using RasPBX image on SD card. My intention is to use Raspberry Pi, mounted in some box outside the house, as a platform to be able to…
Kordian
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Which options are there for ip phone provisioning servers?

I want to know which options exist to provision (configure) multiple VoIP phones from multiple vendors for use with an Asterisk server. I'd like some kind of interface to manage extensions, configuration templates and so on. Here's what I found so…
bernie
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How do you edit extensions_additonial.conf in asterisk?

I need to change one character in the file, but Asterisk overwrites that file. How can I make the changes persistent?
dscottvtg
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Asterisk / Freepbx does not set CallerID to calling party when queue contains a cell phone

Does Asterisk / FreePBX support the ability to pass the caller ID of an inbound caller to a remote support agent (on a cell phone)? Our work has a queue for incoming calls which contains "remote agents" (people on cell phones). To the cell phone…
user3431540
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How to make a trunk in Asterisk 13 FreePBX to call to other SIP networks, a voip peering?

I have made a trunk to make calls within my sip group (i.e) sip.antisip.com Now I want to make calls to another sip network (i.e) sip.fairytel.at. I know we need to make dedicate trunks for these, but I am not sure of the configurations that I…
Dinesh
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FreePBX showing only Welcome Page. Internal Pages are not working

I have FreePBX setup on CentOS, I was trying to install PHPMyAdmin. But after installation it changed some Apache/httpd configurations which conflicted with the FreePBX configurations. Now FreePBX showing only Welcome Page. I am able to login to…
K Ravi
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chan_sip.c:21050 handle_response_invite: " Failed to authenticate on INVITE to " in asterisk

this problem came up when i tried forwarding calls.. -- Executing [1001@users:1] Macro("SIP/to_freepbx-0000003a", "stduser,1001,tT") in new stack -- Executing [s@macro-stduser:1] GotoIf("SIP/to_freepbx-0000003a", "1?FORWARD") in new stack --…
Efren Al Añora
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AsteriskNOW IP PBX behind NAT, clients cannot connect from outside Network

This is my first time working with asterisk (basically i know nothing, so bear with me) i am running Asterisk 11.6 in a virtualbox with 512/kbps internet connection, which is behind NAT. have two extension 1001 and 1002, these are the situations…
user4631236
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Use of Digium 24-port FXO analog cards for FreePBX system

Does anyone have experience with the installation of Digium 1AEX2400 Series cards in a FreePBX distro to control 24 POTS lines? I am researching a new phone system and want to see if anyone has successfully used this hardware before, and if there…
MicDunDee
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Running Asterisk and it's mysql database in different docker containers

If I have correctly understood the use and purpose of docker, every process should be inside it's own container, thus not interfering with other running processes. So based on that, I would like to run an Asterisk PBX server in one container, mysql…
user4916078
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