https://internet-in-a-box.org[Internet-in-a-Box (IIAB)] can install https://asterisk.org/[Asterisk] and https://freepbx.org/[FreePBX] for Voice over IP (VoIP) calls using regular Android and iPhone softphone (SIP) apps — e.g. for low-cost and rural telephony.
As of March 2023, IIAB installs https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Documentation[Asterisk 20] and https://www.freepbx.org/freepbx-16-is-now-released-for-general-availability/[FreePBX 16].
*PHP 7.4 is REQUIRED (https://github.com/iiab/iiab/pull/2899[PR #2899]) and PHP 8.x does not yet work (https://github.com/iiab/iiab/pull/3019#issuecomment-962469346[PR #3109]) — this remains true in 2023, and will likely remain true until https://github.com/FreePBX/framework/tree/release/17.0[FreePBX 17] is eventually released — so please consider installing on https://github.com/iiab/iiab/wiki/IIAB-Platforms#operating-systems[Debian 11 "Bullseye", or 64-bit Raspberry Pi OS versions based on "Bullseye"] (https://github.com/iiab/iiab/pull/3523[PR #3523]).*
As of August 2021, IIAB installs https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Documentation[Asterisk 18] and https://www.freepbx.org/freepbx-16-beta-is-here/[FreePBX 16 Beta], as required by the latest PHP 7.4 Linux OS's (https://github.com/iiab/iiab/pull/2899[PR #2899]). Please consider installing this on https://github.com/iiab/iiab/wiki/IIAB-Platforms#operating-systems[Ubuntu 20.04+, Debian 11 — or the imminent Raspberry Pi OS 11 "Bullseye"].
_PLEASE UNDERSTAND THIS MEANS THAT: IIAB no longer supports FreePBX 15 (Linux with PHP < 7.4, e.g. Raspberry Pi OS 10 "Buster"). Thank you for your understanding, as we look to the future together!_
_Upcoming:_ IIAB will consider supporting Asterisk 19, on or around its 2021-09-28 expected release date (https://github.com/iiab/iiab/issues/2934[#2934]).
_Historical:_ Back in February 2019, IIAB had installed Asterisk 16 and FreePBX 15, e.g. for Ubuntu 18.04, Debian 9 "Stretch" and experimentally, Raspberry Pi (https://github.com/iiab/iiab/issues/1467[#1467]).
https://en.wikipedia.org/wiki/Asterisk_(PBX)[Asterisk] is a software implementation of a private branch exchange (PBX). In conjunction with suitable telephony hardware interfaces and network applications, Asterisk is used to establish and control telephone calls between telecommunication endpoints, such as customary telephone sets, destinations on the public switched telephone network (PSTN), and devices or services on Voice over Internet Protocol (VoIP) networks. Its name comes from the asterisk (*) symbol for a signal used in dual-tone multi-frequency (DTMF) dialing.
https://en.wikipedia.org/wiki/FreePBX[FreePBX] is a web-based open source GUI (graphical user interface) that controls and manages Asterisk (PBX), the open source communications server.
Accept the challenge! Make sure your IIAB configuration file (https://wiki.iiab.io/go/FAQ#What_is_local_vars.yml_and_how_do_I_customize_it%3F[/etc/iiab/local_vars.yml]) contains:
FreePBX can be used with either or both web servers — NGINX on port 80 (http://box/freepbx) and/or using the old approach with Apache on port 83 (http://box:83/freepbx).
If you still want the older Apache approach, set this line in your https://wiki.iiab.io/go/FAQ#What_is_local_vars.yml_and_how_do_I_customize_it%3F[/etc/iiab/local_vars.yml] prior to installing IIAB:
If using PBX intensively, please adjust `/etc/php/X.Y/apache2/php.ini`, `/etc/php/X.Y/cli/php.ini` and/or `/etc/php/X.Y/nginx/php.ini` (where `X.Y` is typically 7.4) as outlined within link:../www_options/tasks/main.yml#L86-L129[/opt/iiab/iiab/roles/www_options/tasks/main.yml] — some of which happens automatically if you also set:
As of April 2023 (https://github.com/iiab/iiab/pull/3523[PR #3523]) IIAB will patch Asterisk automatically (https://github.com/asterisk/asterisk/pull/32[PR asterisk/asterisk#32]) so it can be run experimentally on Raspberry Pi, so long as you keep this default settings:
Optionally, you may want to enable https://github.com/wdoekes/asterisk-chan-dongle[chan_dongle], which is a channel driver for Huawei UMTS cards (e.g. 3G USB dongles) allowing regular voice calls over GSM mobile networks. You will need to configure a dongle post-install, for it to be recognized properly:
// After installing PBX as part of IIAB, please visit http://box.lan/freepbx (NGINX) or http://box.lan:83/freepbx (Apache) and proceed with initial configuration (no login/password is required initially — you will be asked to set this up!)
Browse to http://box/freepbx (NGINX) or http://box:83/freepbx (Apache), and proceed with initial configuration (no login/password is required initially — you will be asked to set this up!)
* Go to *Applications* > *Extensions* > *Add Extension* > *Add New SIP [chan_pjsip] Extension*, and create a phone extension (local phone number) such as the following:
You can now register the extension using a softphone (SIP) app on your smartphone or laptop. In this example we will use the https://en.wikipedia.org/wiki/Linphone[Linphone] app, on an Android phone. After you open the app, follow these steps:
* _If you've created more than one extension, make a call to another extension!_ If you've not yet made more than one extension, try calling an arbitrary extension, or try calling your own extension (your own phone number).
Due to Linphone's inability for phones to ring when the screen is locked ,you can alternatively use a softphone (SIP) app on your smartphone or laptop called Wave Lite. In this example we will use the [https://wiki.zenitel.com/wiki/Grandstream_Wave_Lite_mobile_appapp], on an Android phone. After you open the app, follow these steps:
* Connect your smartphone or laptop to the *Internet in a Box* WiFi hotspot
* Go to Account Settings *
* Add new account *
* Select *SIP ACCOUNT*
** *Account name* is your name,eg John Doe*
** *Sip Server* is your IIAB server's IP address
** *SIP User ID* is your extension number eg 301
** *SIP Authentification ID* is your extension number eg 301
** Please create a https://en.wikipedia.org/wiki/Privacy_policy[Privacy Policy] against abusive surveillance, and explain it to the people in your community. Strongly consider giving them access to their own statistics with the <<#UCP,User Control Panel>> summarized further below.
* Create two extensions *301* and *302*, and configure a dialplan for routing calls (as specified in link:templates/pjsip_custom.conf[pjsip_custom.conf] and link:templates/extensions_custom.conf[extensions_custom.conf], located within `/opt/iiab/iiab/roles/pbx/templates`).
* Make a test call to extension 1000 (that has no physical device associated with it) that plays some sound files.
* After the script completes, it deletes the extensions and reverts file changes to restore asterisk to its original state. If you'd like your changes to persist, read the options below.
Extensions created during the test are retained. Files are not reverted after the test.
+
You may try making calls to extension 302 or 1000 after registering your softphone as <<#Register_Extension,mentioned above>>. The password for the extension is present in pjsip_custom.conf, feel free to change it before executing the script.
Use this option if you executed the script using the *retain* option, but would like to have the changes reverted now. This will only revert the changes to the files and restart asterisk, no other actions are performed.
. Read more about creating https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts[*extensions and dialplans*] if you want to extend this script.
. If you see _Asterisk is not connected_ in red within FreePBX's web interface, check that Asterisk is running using Linux command `sudo pgrep asterisk`. You should see at least 2 different process numbers. If not, please restart Asterisk using Linux command: `sudo fwconsole restart`
A better way however, is to use http://box/freepbx or http://box:83/freepbx > *Settings* > *Advanced Settings* to modify those stored settings — or equivalently, run Linux commands like:
* If you'd like to allow users more control over their own settings, and the ability to review their own statistics, install the https://wiki.freepbx.org/pages/viewpage.action?pageId=28180526[User Control Panel] FreePBX module from http://box/freepbx or http://box:83/freepbx > *Admin* > *Module Admin* > *Check Online*.
. [[Password_Recovery]]If you forget your FreePBX password, you can bypass it using Linux command `sudo fwconsole unlock <Session ID>` and then refresh your FreePBX web page:
* To get the Session ID, press `Ctrl + a` which will highlight all the text on the page. The text highlighted in the middle of the page, is the Session ID:
* Once you've logged in, change your password under *Admin* > *Administrators*. On the right side, you will see the list of available users. You can select the appropriate user and change the password:
_If there's a bug or serious problem with IIAB, please do https://internet-in-a-box.org/contributing.html[make contact] and post an issue here: https://github.com/iiab/iiab/issues_
If not, the link:tasks/freepbx.yml#L175-L187[configuration of /var/lib/php/asterisk_sessions/] might be made conditional upon `when: not pbx_use_apache`
. The link:tasks/freepbx.yml#L214-L221[installation of /etc/odbc.ini] for CDR (Call Detail Records) database `asteriskcdrdb` might benefit from compiling the ODBC driver for aarch64, per https://mghadam.blogspot.com/2021/03/install-asterisk-18-freepbx-15-on.html ?
See the output of `asterisk -rx "cdr show status"` as mentioned at https://github.com/iiab/iiab/pull/2938#issuecomment-898693126[#2938] and https://github.com/iiab/iiab/pull/2942[PR #2942].
Node.js applications like Asterisk/FreePBX, Node-RED and Sugarizer won't work on Raspberry Pi Zero W (ARMv6) if you installed Node.js while on RPi 3, 3 B+ (ARMv7) or RPi 4 (ARMv8). If necessary, run `apt remove nodejs` or `apt purge nodejs` then `rm /etc/apt/sources.list.d/nodesource.list; apt update` then (https://nodered.org/docs/hardware/raspberrypi#swapping-sd-cards[attempt!]) to link:../nodejs/tasks/main.yml[install Node.js] _on the Raspberry Pi Zero W itself_ (a better approach than "cd /opt/iiab/iiab; ./runrole nodejs" is to try `apt install nodejs` or try installing the tar file mentioned at https://github.com/iiab/iiab/issues/2082#issuecomment-569344617[#2082]). You might also need `apt install npm`. Whatever versions of Node.js and npm you install, make sure `/etc/iiab/iiab_state.yml` contains the line `nodejs_installed: True` (add it if nec!) Finally, proceed to install Asterisk/FreePBX, Node-RED and/or Sugarizer. https://github.com/iiab/iiab/issues/1799[#1799]
As of 2019-02-14, "systemctl restart freepbx" failed more than 50% of the time when run on a https://wiki.iiab.io/go/FAQ#What_services_.28IIAB_apps.29_are_suggested_during_installation%3F[LARGE-sized] install of IIAB 6.7 on RPi 3 or RPi 3 B+.
It is possible that FreePBX restarts much more reliably when run on a SMALL-sized install of IIAB? Please https://wiki.iiab.io/go/FAQ#What_are_the_best_places_for_community_support%3F[contact us] if you can assist here in any way: https://github.com/iiab/iiab/issues/1493[#1493]
In February 2019, this https://github.com/iiab/iiab/tree/master/roles/pbx[roles/pbx] IIAB https://github.com/iiab/iiab/wiki/IIAB-Contributors-Guide#ansible[Ansible playbook] was originally inspired by Yannik Sembritzki's https://github.com/Yannik/ansible-role-asterisk[Asterisk] and https://github.com/Yannik/ansible-role-freepbx[FreePBX] Ansible work from 2017-2019.
