How to Install Asterisk 18 and FreePBX 16 on Raspberry Pi OS Bullseye (64-bit & 32-bit)

Deploying Asterisk 18 LTS alongside the modern FreePBX 16 administrative interface on Raspberry Pi OS (Debian 11 Bullseye) enables a carrier-grade VoIP telephony platform on energy-efficient ARM hardware. This guide provides the complete setup for both 64-bit (aarch64) and 32-bit (armhf) architectures.

OS: Raspberry Pi OS Bullseye • Components: Asterisk 18.x, FreePBX 16, Node.js 14, MariaDB 10.5, Apache 2.4

1. Installing System Packages & PHP 7.4

sudo apt-get update && sudo apt-get upgrade -y
sudo apt-get install -y build-essential libssl-dev libncurses5-dev libnewt-dev   libxml2-dev libsqlite3-dev uuid-dev mariadb-server mariadb-client   libjansson-dev libedit-dev sox libsox-fmt-all git curl subversion

# Install PHP 7.4 modules required by FreePBX 16
sudo apt-get install -y apache2 php php-cli php-mysql php-pear php-gd php-mbstring   php-intl php-xml php-curl php-zip

2. Building Asterisk 18 LTS

cd /usr/src
sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18-current.tar.gz
sudo tar -zxf asterisk-18-current.tar.gz
cd asterisk-18.*

sudo contrib/scripts/get_mp3_source.sh
sudo contrib/scripts/install_prereq install

sudo ./configure --with-pjproject-bundled --with-jansson-bundled
sudo make menuselect.makeopts
sudo menuselect/menuselect --enable format_mp3 menuselect.makeopts
sudo make -j$(nproc)
sudo make install
sudo make config
sudo ldconfig

3. Installing FreePBX 16

cd /usr/src
sudo wget http://mirror.freepbx.org/modules/packages/freepbx/freepbx-16.0-latest.tgz
sudo tar -zxf freepbx-16.0-latest.tgz
cd freepbx
sudo ./install -n

TD-LTE Router In-Depth Comparison: AirMaster 3100V vs GreenPacket DT-350

Evaluating indoor TD-LTE desktop modems for Band 42 (3500 MHz) deployments requires comparing RF sensitivity, throughput stability under weak signal conditions, and hardware VoIP capabilities. Here is an empirical comparison between the AirMaster 3100V and the GreenPacket DT-350.

Tested Models: AirMaster 3100V (Firmware V2.0.0B31) • GreenPacket DT-350 (Unlocked Band 42 SIMs)

Key Findings and Benchmark Summary

Feature	AirMaster 3100V	GreenPacket DT-350
VoIP Hardware	No FXS RJ-11 port	Built-in SIP client & RJ-11 port
Wi-Fi Performance	802.11n 300Mbps	802.11n 300Mbps
Weak Signal Stability	Superior antenna gain (RSRP stability)	Moderate gain; benefits from external SMA antennas
Firmware Flexibility	Clean web management	Requires unlocking for full VoIP/APN menus

Recommendation: If telephone VoIP handset integration is required, the GreenPacket DT-350 is the clear choice. For raw RF reception in fringe coverage areas, the AirMaster 3100V delivers more consistent speeds.

How to Install Asterisk 18 and FreePBX 15 on Ubuntu (Raspberry Pi 4)

Running a full-featured IP-PBX telephony server on a Raspberry Pi 4 provides small offices and lab environments with an enterprise telephone system at negligible power consumption. This comprehensive guide covers compiling Asterisk 18 LTS with DAHDI/libpri and configuring the FreePBX 15 web administration portal on Ubuntu Server (ARM64).

Hardware: Raspberry Pi 4 (4GB / 8GB recommended) • OS: Ubuntu 20.04 LTS aarch64 • Stack: Asterisk 18 LTS, FreePBX 15, MariaDB, Node.js

1. Installing Build Prerequisites and MariaDB

sudo apt-get update && sudo apt-get upgrade -y
sudo apt-get install -y build-essential linux-headers-generic libssl-dev libncurses5-dev   libnewt-dev libxml2-dev linux-headers-$(uname -r) libsqlite3-dev uuid-dev   mariadb-server mariadb-client libjansson-dev libedit-dev sox libsox-fmt-all

# Configure database
sudo mysql_secure_installation

2. Compiling Asterisk 18 LTS

cd /usr/src
sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18-current.tar.gz
sudo tar -zxf asterisk-18-current.tar.gz
cd asterisk-18.*

sudo contrib/scripts/get_mp3_source.sh
sudo contrib/scripts/install_prereq install

sudo ./configure --with-pjproject-bundled --with-jansson-bundled
sudo make menuselect.makeopts
sudo menuselect/menuselect --enable format_mp3 menuselect.makeopts
sudo make -j$(nproc)
sudo make install
sudo make samples
sudo make config
sudo ldconfig

3. Installing FreePBX 15

Install Node.js, Apache, PHP modules, and run the FreePBX install script pointing to Asterisk:

cd /usr/src
sudo wget http://mirror.freepbx.org/modules/packages/freepbx/freepbx-15.0-latest.tgz
sudo tar -zxf freepbx-15.0-latest.tgz
cd freepbx
sudo ./install -n

How to Access Native SIP / VoIP Account Settings Across Different Android Versions

Many Android releases include a built-in native SIP client directly integrated into the system Dialer application, allowing users to make and receive VoIP calls over Wi-Fi and mobile data without installing third-party battery-draining apps. However, OEM custom skins (Samsung, Huawei, Asus, Xiaomi) frequently relocate the configuration menu.

Native SIP Client Path Reference by Android OS Version

• Android 4.2.2 (Jelly Bean):
  Dialer App → Menu → Settings → Internet Call → Accounts
• Android 5.0 / 5.1 (Lollipop):
  Phone App → Settings → Call Settings → Phone Account Settings → SIP Accounts
• Android 6.0 (Marshmallow) & 7.0 (Nougat):
  Phone App → Three Dots Menu → Settings → Calling Accounts → SIP Accounts
• Android 8.0 (Oreo) through Android 11:
  Phone App → Settings → Calls → Calling Accounts → SIP Accounts → Add (+)

Note: In Android 12 and newer, Google deprecated the legacy built-in SIP stack. For modern Android releases, standard standalone clients like Linphone, Grandstream Wave, or MizuDroid are recommended.

How to Configure VoIP SIP Settings on GreenPacket DT-350 4G Routers

The GreenPacket DT-350 TD-LTE indoor router includes an integrated hardware VoIP FXS port (RJ-11) for connecting analog telephone handsets directly to SIP servers (such as Asterisk, FreePBX, or telecom IMS gateways). However, on many telecom-branded firmwares (such as Irancell), the VoIP menu is hidden from standard administrative accounts.

Hardware: GreenPacket DT-350 TD-LTE Router • RJ-11 POTS Handset • SIP Protocol

1. Accessing Hidden VoIP Menus

Log in with the root/operator credentials (or inspect the web GUI source to reveal hidden navigation items under /voip_settings.html or /sip.html).

2. SIP Account Configuration

SIP Proxy / Registrar: YOUR.PBX.SERVER.IP
Port: 5060
SIP Transport: UDP
Outbound Proxy: (Optional, set if behind telecom NAT)
Account / Display Name: 101
Authentication Username: 101
Secret / Password: YOUR_SIP_SECRET
Codec Priority: G.711u (PCMU), G.711a (PCMA), G.729

Save and apply settings. Once registered, the Phone LED on the router faceplate illuminates solid green, and connected analog handsets will receive dial tone.