freepbx | FreePBX 15 Docker Image CentOS 7.8 build | TCP library

 by   techno-express Python Version: Current License: No License

kandi X-RAY | freepbx Summary

kandi X-RAY | freepbx Summary

freepbx is a Python library typically used in Networking, TCP, Docker applications. freepbx has no bugs and it has low support. However freepbx has 19 vulnerabilities and it build file is not available. You can download it from GitHub.

FreePBX 15 Docker Image CentOS 7.8 build with Asterisk 16 RPM
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              freepbx has a low active ecosystem.
              It has 9 star(s) with 4 fork(s). There are 1 watchers for this library.
              OutlinedDot
              It had no major release in the last 6 months.
              There are 0 open issues and 4 have been closed. On average issues are closed in 527 days. There are no pull requests.
              It has a neutral sentiment in the developer community.
              The latest version of freepbx is current.

            kandi-Quality Quality

              freepbx has no bugs reported.

            kandi-Security Security

              freepbx has 19 vulnerability issues reported (1 critical, 5 high, 13 medium, 0 low).

            kandi-License License

              freepbx does not have a standard license declared.
              Check the repository for any license declaration and review the terms closely.
              OutlinedDot
              Without a license, all rights are reserved, and you cannot use the library in your applications.

            kandi-Reuse Reuse

              freepbx releases are not available. You will need to build from source code and install.
              freepbx has no build file. You will be need to create the build yourself to build the component from source.
              Installation instructions, examples and code snippets are available.

            Top functions reviewed by kandi - BETA

            kandi has reviewed freepbx and discovered the below as its top functions. This is intended to give you an instant insight into freepbx implemented functionality, and help decide if they suit your requirements.
            • 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 .
            Get all kandi verified functions for this library.

            freepbx Key Features

            No Key Features are available at this moment for freepbx.

            freepbx Examples and Code Snippets

            No Code Snippets are available at this moment for freepbx.

            Community Discussions

            QUESTION

            Webphone for Freepbx server
            Asked 2021-Apr-01 at 01:31

            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 01:31

            QUESTION

            How to add data to the queue log?
            Asked 2020-Oct-14 at 17:12

            I have freepbx, Asterisk version 13.32.0 built by mockbuild @ jenkins7.
            We need to include into queue_log file additional field with CallerID number. How can i do this?
            Right now I have a table on mysql - queue_log, but I want to see the Callerid number. Is it possible?

            Thanks!

            ...

            ANSWER

            Answered 2020-Oct-14 at 17:12

            ENTERQUEUE event's data2 is callerid.

            There are no ways add more info, except you can use QueueLog app to add custom info.

            Source https://stackoverflow.com/questions/64353035

            QUESTION

            Not getting Any Events From Asternet.Ari On FreePbx
            Asked 2020-Jan-21 at 13:38

            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 07:17

            Well 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

            Source https://stackoverflow.com/questions/53886515

            QUESTION

            How to fix the missing BLINDTRANSFER and ATTENDEDTRANSFER log entries in the queue_log file?
            Asked 2019-Apr-02 at 04:22

            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 04:22

            If 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

            Source https://stackoverflow.com/questions/55457583

            QUESTION

            Originate a call with text to speech message
            Asked 2019-Feb-27 at 07:58

            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 13:34

            The 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:

            Source https://stackoverflow.com/questions/54809378

            QUESTION

            asterisk queue_log late COMPLETEAGENT
            Asked 2019-Feb-23 at 06:14

            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 06:14

            1) 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.

            Source https://stackoverflow.com/questions/54828265

            QUESTION

            Asterisk start dialpan many times at the same times
            Asked 2019-Feb-21 at 10:00

            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 10:00

            I needed to change the section

            Source https://stackoverflow.com/questions/54652116

            QUESTION

            Asterisk Dialplan AGI Script Not Executing (Possible Asterisk Permissions issue?)
            Asked 2018-Nov-27 at 15:15

            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 15:15

            Just 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.

            Source https://stackoverflow.com/questions/53483000

            QUESTION

            How should I update FreePBX version 2.0.X to 14?
            Asked 2018-Nov-26 at 12:37

            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 12:37

            You 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

            Source https://stackoverflow.com/questions/53402300

            QUESTION

            How to create/remove users on FreePBX+Asterisk using REST ARI & PHP
            Asked 2018-Jun-06 at 16:47

            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 16:47

            There 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.

            Source https://stackoverflow.com/questions/48781458

            Community Discussions, Code Snippets contain sources that include Stack Exchange Network

            Vulnerabilities

            In Sangoma FreePBX 13 through 15 and sysadmin (aka System Admin) 13.0.92 through 15.0.13.6 modules have a Remote Command Execution vulnerability that results in Privilege Escalation.
            An XSS Injection vulnerability exists in Sangoma FreePBX and PBXact 13, 14, and 15 within the Debug/Test page of the Superfecta module at the admin/config.php?display=superfecta URI. This affects Superfecta through 13.0.4.7, 14.x through 14.0.24, and 15.x through 15.0.2.20.
            Multiple XSS vulnerabilities exist in the Backup & Restore module \ v14.0.10.2 through v14.0.10.7 for FreePBX, as shown at /admin/config.php?display=backup on the FreePBX Administrator web site. An attacker can modify the id parameter of the backup configuration screen and embed malicious XSS code via a link. When another user (such as an admin) clicks the link, the XSS payload will render and execute in the context of the victim user's account.
            An XSS Injection vulnerability exists in Sangoma FreePBX and PBXact 13, 14, and 15 within the Call Event Logging report screen in the cel module at the admin/config.php?display=cel URI via date fields. This affects cel through 13.0.26.9, 14.x through 14.0.2.14, and 15.x through 15.0.15.4.
            In userman 13.0.76.43 through 15.0.20 in Sangoma FreePBX, XSS exists in the User Management screen of the Administrator web site. An attacker with access to the User Control Panel application can submit malicious values in some of the time/date formatting and time-zone fields. These fields are not being properly sanitized. If this is done and a user (such as an admin) visits the User Management screen and views that user's profile, the XSS payload will render and execute in the context of the victim user's account.
            In userman 13.0.76.43 through 15.0.20 in Sangoma FreePBX, XSS exists in the user management screen of the Administrator web site, i.e., the/admin/config.php?display=userman URI. An attacker with sufficient privileges can edit the Display Name of a user and embed malicious XSS code. When another user (such as an admin) visits the main User Management screen, the XSS payload will render and execute in the context of the victim user's account.
            An issue was discovered in FreePBX core before 3.0.122.43, 14.0.18.34, and 5.0.1beta4. By crafting a request for adding Asterisk modules, an attacker is able to store JavaScript commands in a module name.
            An issue was discovered in Contactmanager 13.x before 13.0.45.3, 14.x before 14.0.5.12, and 15.x before 15.0.8.21 for FreePBX 14.0.10.3. In the Contactmanager class (html\admin\modules\contactmanager\Contactmanager.class.php), an unsanitized group variable coming from the URL is reflected in HTML on 2 occasions, leading to XSS. It can be requested via a GET request to /admin/ajax.php?module=contactmanager.
            An issue was discovered in Manager 13.x before 13.0.2.6 and 15.x before 15.0.6 before FreePBX 14.0.10.3. In the Manager module form (html\admin\modules\manager\views\form.php), an unsanitized managerdisplay variable coming from the URL is reflected in HTML, leading to XSS. It can be requested via GET request to /config.php?type=tool&display=manager.
            ** DISPUTED ** FreePBX 10.13.66-32bit and 14.0.1.24 (SNG7-PBX-64bit-1712-2) allow post-authentication SQL injection via the order parameter. NOTE: the vendor disputes this issue because it is intentional that a user can "directly modify SQL tables ... [or] run shell scripts ... once ... logged in to the administration interface; there is no need to try to find input validation errors."
            htdocs_ari/includes/login.php in the ARI Framework module/Asterisk Recording Interface (ARI) in FreePBX before 2.9.0.9, 2.10.x, and 2.11 before 2.11.1.5 allows remote attackers to execute arbitrary code via the ari_auth cookie, related to the PHP unserialize function, as exploited in the wild in September 2014.
            Multiple cross-site scripting (XSS) vulnerabilities in FreePBX 2.9 and earlier allow remote attackers to inject arbitrary web script or HTML via the (1) context parameter to panel/index_amp.php or (2) panel/dhtml/index.php; (3) clid or (4) clidname parameters to panel/flash/mypage.php; (5) PATH_INFO to admin/views/freepbx_reload.php; or (6) login parameter to recordings/index.php.
            Directory traversal vulnerability in page.recordings.php in the System Recordings component in the configuration interface in FreePBX 2.8.0 and earlier allows remote authenticated administrators to create arbitrary files via a .. (dot dot) in the usersnum parameter to admin/config.php, as demonstrated by creating a .php file under the web root.
            Multiple cross-site scripting (XSS) vulnerabilities in FreePBX 2.5.1, and other 2.4.x, 2.5.x, and pre-release 2.6.x versions, allow remote attackers to inject arbitrary web script or HTML via the (1) display parameter to reports.php, the (2) order and (3) extdisplay parameters to config.php, and the (4) sort parameter to recordings/index.php. NOTE: some of these details are obtained from third party information.
            Multiple cross-site request forgery (CSRF) vulnerabilities in FreePBX 2.5.1, and other 2.4.x, 2.5.x, and pre-release 2.6.x versions, allow remote attackers to hijack the authentication of admins for requests that create a new admin account or have unspecified other impact.
            FreePBX 2.5.1, and other 2.4.x, 2.5.x, and pre-release 2.6.x versions, generates different error messages for a failed login attempt depending on whether the user account exists, which allows remote attackers to enumerate valid usernames.

            Install freepbx

            Using the Webmin UI visit: https://ip_or_hostname:9990.
            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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            CLONE
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            https://github.com/techno-express/freepbx.git

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            gh repo clone techno-express/freepbx

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            git@github.com:techno-express/freepbx.git

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