freepbx | FreePBX 15 Docker Image CentOS 7.8 build | TCP library
kandi X-RAY | freepbx Summary
kandi X-RAY | freepbx Summary
FreePBX 15 Docker Image CentOS 7.8 build with Asterisk 16 RPM
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Top functions reviewed by kandi - BETA
- start unit
- print result
- Read a sysd file
- Open the system .
- set user and group
- Converts time string to seconds .
- Check if a process exists .
- Compares between two configurations .
- Check if a process is running .
- Check if a process has failed .
freepbx Key Features
freepbx Examples and Code Snippets
Community Discussions
Trending Discussions on freepbx
QUESTION
I am configured freepbx (asteriks) server for audio calling, So I need to create web phone which it can be communicate with server afterthat server should sen request to user who calling another user. What kind of solution should I user, are there any open-source solutions?
...ANSWER
Answered 2021-Apr-01 at 01:31Freepbx UCP module already have phone
QUESTION
ANSWER
Answered 2020-Oct-14 at 17:12ENTERQUEUE event's data2 is callerid.
There are no ways add more info, except you can use QueueLog app to add custom info.
QUESTION
I have set up FreePbx and it is working I can make calls into the pbx and out of the pbx. I have enabled the REST API and added a user and password. I cloned the Asternet.Ari https://github.com/skrusty/AsterNET.ARI.
The program runs and I get the connected event:
...ANSWER
Answered 2019-Feb-10 at 07:17Well your Asterisk Ari is connecting, but to get anything in it, you have to create Extension so your call go to Stasis application.
Please edit your extensions.conf file with following information
QUESTION
I write a wallboard for asterisk queue system. The document says that when a call is transferred away by an agent an ATTENDEDTRANSFER (or BLINDTRANSFER) event log should be added to the queue_log file automatically. Unfortunately there is no line for any transferred calls in the log file (queue_log in my case). Is there any setting to be changed to let the system to log them properly ? When I check the CEL files, I see the transfer logs. So the system is logging correctly for CEL but not for queue_log. I tried to transfer the call to another agent, to an IVR and to another user who is not an agent for any queue. The result is the same, no log for the transfer process.
Any suggestions ?
I use the following: Asterisk Version: 13.22.0 Freepbx 14.0.5.25 All trunks and clients are connected via SIP
...ANSWER
Answered 2019-Apr-02 at 04:22If your phone do transfer via internal features - no log entries.
You have parse AMI events for needed info.
Write your own queue wallboard is VERY hard task. Queue module have really alot of issues. Can recommend read some already written modules like https://www.asternic.net or queuemetric
QUESTION
I use the Asterisk-Manager package for NodeJs
https://www.npmjs.com/package/asterisk-manager
and have a tape announcement as a text which must be translated via text to speech. When I try to call an outgoing phone number how can I setup the text to speech variable and the recipient? An example would be
...ANSWER
Answered 2019-Feb-26 at 13:34The Originate application itself will only send the called number to an application or extension. You should have an audio file created before calling the playback app. So you code will look like this:
QUESTION
I am trying to write a wallboard for my asterisk server. This wallboard will process the queue_log file in /var/log/asterisk.
Here is a scenario in question:
1) A customer calls out call center. Let his number be 44556677889900 and our number 8881234567890.
2) The customer enters the queue 210.
3) Agent 1 takes the call.
4) Agent 1 decides that the call should go to another queue. And transfers it to queue 209
5) Agent 2 takes the call.
6) Agent 2 terminates the call after talking with the customer. (When Agent 2 is talking on the phone Agent 1 is idle and available for a new call.
7) Normally Agent 1 ended his call at 4th step, but the log with COMPLETEAGENT appears just now, even the agent is available since 4th step
Here is the output in the queue_log:
1550582529|1550582516.26480|210|NONE|DID|8881234567890 * 1. step* 1550582529|1550582516.26480|210|NONE|ENTERQUEUE||44556677889900|1 * 2. step* 1550582531|1550582516.26480|210|Test Agent 1|CONNECT|2|1550582529.26493|2 3. step 1550582536|1550582536.26498|209|NONE|DID| ** 4. step** 1550582536|1550582536.26498|209|NONE|ENTERQUEUE||9991|1 4. step 1550582539|1550582536.26498|209|Test Agent 2|CONNECT|3|1550582536.26499|2 5. step 1550582543|1550582536.26498|209|Test Agent 2|COMPLETECALLER|3|4|1 6. step 1550582549|1550582516.26480|210|Test 1|COMPLETEAGENT|2|18|1 7. step
As mentioned in the 7th step, Agent 1 if available for new calls after he transfers the call to queue 209. (In fact if a new call comes, the system send the call to Agent 1). However the log "COMPTELEAGENT" appears only when the customer disconnects.
This makes my wallboard think that Agent 1 is busy even he is not. And worse if he received a new call before Agent 2 finishes, everything gets more complicated.
Questions:
1) How it is possible to make the system send the COMPLETEAGENT at step 4 ?
2) Why is ATTENDEDTRANSFER log missing ? (Not related to this problem directly but can also be connected)
Asterisk Version: 13.22.0
Freepbx 14.0.5.25
Thank you in advance.
...ANSWER
Answered 2019-Feb-23 at 06:141) System should not send COMPLEATEAGENT at 4, becuase thoose event should be sent AFTER END of call. That event is created by QUEUE, not by AGENT. From queue's point of view call not yet finished.
If you want it be finished, do transfer of LEGA, not queue's LEG.
2)Transfer subsystem not related to queue subsystem and SHOULD NOT be related in any realible PBX. You can write your own if you want.
Side notes
- no point parse queue_log, much simpler setup queue_log in mysql or other db and read it.
- you can write your own queue system using Async AGI.
- you can add as many logs as you want by using dialplan CEL or UserEvents.
QUESTION
Please help understand out what's going on.
Asterisk starts dialpan many times for one input call.
I use Asterisk 15.4.0 (FreePBX 14.0.5.25),
My extensions_custom.conf:
ANSWER
Answered 2019-Feb-21 at 10:00I needed to change the section
QUESTION
I have an unusual situation with my Asterisk.
Firstly the machine is a FreePBX 14, running Asterisk 15.4.0
I have a dialplan which takes card details using an IVR, i.e Enter Card Number followed by the Hash key.
This then puts together a whole string to execute a separate Perl Script that will charge the customers card with the requested amount.
If i run the perl script from the CLI, the script executes fine and charges the card.
If I go through the dialplan, providing the relevant card details, when it gets to the end to execute the script, it all appears to work, but nothing happens in terms of charging the card.
In an effort to see the AGI script running and seeing what's going wrong, I run asterisk as 'asterisk -vvvvvc' as root, and do the same again, the payment goes through, and works completely fine.
This leads me to believe that when running asterisk as (asterisk -vvvc) it runs with elevated permissions allowing the script to run properly.
Any ideas as to how i can have this working normally or what permissions I need to fix.
The script is set to 0777, so should be executable by everything, and I've also set the script to be owned by asterisk AND root, and that made no difference.
Here is the command I am using in the dialplan to invoke the script.
...ANSWER
Answered 2018-Nov-27 at 15:15Just in case someone else has an issue similar to this in the future, the problem was most likely due to the Stripe (Payment Server) CLI program I was using, it didn't like being run in a script where it was not in a normal terminal, explains why it worked in both scenarios but not in the dialplan.
QUESTION
I am working on a project and the client is using freePBX version 2.0.X and they want to upgrade their version to 14.0.X. How should I do this keeping in mind that all the data and configurations needs to be migrated to the newer version?
edit: to those who are marking it negative, I am new to this and if I new the answer then, I wouldn't ask.
...ANSWER
Answered 2018-Nov-26 at 12:37You have update 2.0->2.2. After that 2.2 -> 2.4, 2.4->2.5, 2.5->2.6
After that you have updat 2.6 to 13, 13 to 14.
Some of this upgrades not go without problem, huge experience required.
On upgrade from 2.6 to 13 you have also upgrade asterisk to version 13.
If you have no very complex config, in most cases faster will be backup extensions, dids, restore, put all other config items.
ps also you need have archive for thoose version,i think it will be complex to find something below 2.4
QUESTION
I have FreePBX distro installed (containing FreePBX 14 • Linux 7.4 • Asterisk 13) on one machine IP: 192.168.1.129.
I have another machine with XAMPP web server (MyWebApp) IP: 192.168.1.22.
I want to create and remove users in FreePBX from MyWebApp interface. FreePBX wiki says RESTful interaction is possible, however I'm stuck and more confused after going through FreePBX wiki links and Asterisk ARI documents.
Appreciate if anyone can clarify any of the following:
- is it possible in the first place what I'm trying to achieve?
- What needs to be enabled on FreePBX?
- What additional needs to be installed on MyWebApp to interact with FreePBX?
Thanks in advance.
...ANSWER
Answered 2018-Jun-06 at 16:47There is no native REST API for FreePBX, but there is a third-party module.
However, there is no available documentation on that topic; refer to this thread: https://community.freepbx.org/t/rest-api-clarity/35740
You also can always read the source code of the module or ask someone to write a syncing module for your need. Source code of FreePBX project is open-source.
Community Discussions, Code Snippets contain sources that include Stack Exchange Network
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Install freepbx
Certificate/private key file /etc/webmin/letsencrypt-cert.pem
Private key file /etc/webmin/letsencrypt-key.pem
Certificate authorities file /etc/webmin/letsencrypt-ca.pem"
This installation has an menu link pointing to: https://ip_or_hostname/avantfax.
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