📞 Asterisk PBX Integration

Advanced telephony features for your YateBTS setup

🎯 Overview

Asterisk PBX is a powerful open-source telephony platform that provides advanced call management, routing, and telephony features. When integrated with your YateBTS setup, it enables sophisticated call handling, voicemail, conferencing, and more.

Why Integrate Asterisk PBX?

  • Advanced Call Management - Sophisticated call routing and handling
  • Voicemail System - Professional voicemail with email notifications
  • Conference Calling - Multi-party conference capabilities
  • Call Recording - Record calls for quality assurance
  • IVR Systems - Interactive Voice Response menus
  • Integration - Seamless integration with YateBTS

System Requirements

Hardware

  • Raspberry Pi 4 with 8GB RAM
  • 2TB M.2 SSD (recommended)
  • BladeRF Mini A4 SDR device
  • GeeekPi DeskPi Lite case

Software

  • Ubuntu 22.04 LTS
  • YateBTS (already installed)
  • Asterisk 21.x (to be installed)
  • FreePBX (optional web interface)

🚀 Installation

Follow these steps to install Asterisk PBX on your Raspberry Pi 4 8GB system.

Step 1: Update System

# Update package lists sudo apt update && sudo apt upgrade -y # Install required dependencies sudo apt install -y build-essential wget curl libnewt-dev libssl-dev libncurses5-dev subversion libsqlite3-dev libjansson-dev libxml2-dev uuid-dev libsrtp2-dev

Step 2: Download and Compile Asterisk

# Create asterisk user sudo useradd -m -s /bin/bash asterisk sudo usermod -aG audio,dialout asterisk # Download Asterisk source cd /usr/src sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21-current.tar.gz sudo tar -xzf asterisk-21-current.tar.gz cd asterisk-21.* # Configure and compile sudo ./configure --libdir=/usr/lib/arm-linux-gnueabihf sudo make menuselect.makeopts sudo make -j4 sudo make install sudo make config sudo ldconfig

Step 3: Install Asterisk Modules

# Install additional modules sudo make samples sudo make install-logrotate # Set permissions sudo chown -R asterisk:asterisk /var/lib/asterisk sudo chown -R asterisk:asterisk /var/log/asterisk sudo chown -R asterisk:asterisk /var/spool/asterisk sudo chown -R asterisk:asterisk /etc/asterisk

Step 4: Enable and Start Services

# Enable Asterisk service sudo systemctl enable asterisk sudo systemctl start asterisk # Verify installation sudo systemctl status asterisk sudo asterisk -r -x "core show version"

⚠️ Important Notes

  • Compilation may take 30-60 minutes on Raspberry Pi 4
  • Ensure adequate cooling during compilation
  • Monitor system temperature with vcgencmd measure_temp
  • Consider using a fan or heatsink for extended operation

⚙️ Configuration

Configure Asterisk PBX for optimal performance with your YateBTS setup.

Basic Configuration

# Edit main configuration sudo nano /etc/asterisk/asterisk.conf # Add these settings: [options] verbose = 3 debug = 1 alwaysfork = yes nofork = no quiet = no timestamp = yes execincludes = yes console = yes highpriority = yes initcrypto = yes nocolor = no dontwarn = no dumpcore = yes languageprefix = yes systemname = RFS-Portable-BTS maxcalls = 1000 maxload = 0.9 maxfiles = 1000 minmemfree = 1 cache_record_files = yes record_cache_dir = /tmp transmit_silence = yes

YateBTS Integration Configuration

# Create YateBTS channel configuration sudo nano /etc/asterisk/yatebts.conf # Add YateBTS channel configuration: [yatebts] type=friend host=127.0.0.1 port=5038 username=yatebts secret=yatebts123 context=from-yatebts dtmfmode=rfc2833 canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm

Dialplan Configuration

# Edit dialplan sudo nano /etc/asterisk/extensions.conf # Add basic dialplan: [globals] CONSOLE = Console/dsp IAXINFO = guest TRUNK = DAHDI/G2 TRUNKMSD = 1 [from-yatebts] ; Incoming calls from YateBTS exten => _X.,1,NoOp(Incoming call from YateBTS: ${CALLERID(num)}) exten => _X.,n,Answer() exten => _X.,n,Playback(hello-world) exten => _X.,n,Hangup() [internal] ; Internal extensions exten => 100,1,Dial(SIP/100) exten => 101,1,Dial(SIP/101) exten => 102,1,Dial(SIP/102) [outbound] ; Outbound calls exten => _9NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@yatebts) exten => _9NXXNXXXXXX,n,Hangup()

🌟 Features

Asterisk PBX provides a comprehensive set of telephony features for your YateBTS setup.

📞 Call Management

  • Call routing and forwarding
  • Call transfer and hold
  • Call waiting and caller ID
  • Call recording and monitoring

📧 Voicemail System

  • Professional voicemail
  • Email notifications
  • Voicemail to email
  • Web-based access

🎤 Conference Calling

  • Multi-party conferences
  • Conference recording
  • Moderator controls
  • PIN-protected rooms

🎛️ IVR Systems

  • Interactive menus
  • Custom prompts
  • DTMF handling
  • Call routing

📊 Call Analytics

  • Call detail records
  • Usage statistics
  • Performance monitoring
  • Billing integration

🔒 Security Features

  • Call authentication
  • Access control
  • Encryption support
  • Firewall integration

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Advanced Features

Feature Description Use Case
Call Queues Manage incoming call queues with agents Customer service, support centers
Follow Me Ring multiple devices simultaneously Mobile workforce, remote workers
Call Parking Park calls for retrieval from other extensions Office environments, call centers
Music on Hold Play music while callers wait Professional call handling
Call Screening Screen incoming calls before answering Privacy protection, spam filtering

🔗 YateBTS Integration

Integrate Asterisk PBX with your existing YateBTS setup for seamless operation.

Integration Architecture

System Architecture

YateBTS → Asterisk PBX → End Users

  • YateBTS handles GSM radio interface
  • Asterisk PBX manages call routing and features
  • Users get advanced telephony capabilities

Configuration Steps

# 1. Configure YateBTS to work with Asterisk sudo nano /usr/local/etc/yate/ysipchan.conf # Add Asterisk integration: [general] port=5060 bind=127.0.0.1 context=from-yatebts register=asterisk:[email protected]:5060 # 2. Configure Asterisk SIP channel sudo nano /etc/asterisk/sip.conf # Add YateBTS peer: [yatebts] type=friend host=127.0.0.1 port=5060 username=yatebts secret=yatebts123 context=from-yatebts dtmfmode=rfc2833 canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm

Testing Integration

# Test YateBTS connection sudo asterisk -r -x "sip show peers" # Test call routing sudo asterisk -r -x "dialplan show from-yatebts" # Monitor calls sudo asterisk -r -x "core show channels"

✅ Integration Checklist

  • YateBTS is running and accessible
  • Asterisk PBX is installed and configured
  • SIP channels are properly configured
  • Dialplan routes calls correctly
  • Test calls are working

🔧 Troubleshooting

Common issues and solutions for Asterisk PBX integration.

Common Issues

🔌 Connection Issues

  • Check SIP peer status
  • Verify network connectivity
  • Check firewall settings
  • Review log files

📞 Call Quality

  • Check codec compatibility
  • Monitor system resources
  • Verify audio settings
  • Test with different devices

⚡ Performance

  • Monitor CPU usage
  • Check memory consumption
  • Review disk I/O
  • Optimize configuration

🔒 Security

  • Update authentication
  • Review access controls
  • Check encryption settings
  • Monitor for attacks

Diagnostic Commands

# Check Asterisk status sudo systemctl status asterisk # View Asterisk logs sudo tail -f /var/log/asterisk/messages # Check SIP peers sudo asterisk -r -x "sip show peers" # Check dialplan sudo asterisk -r -x "dialplan show" # Monitor calls sudo asterisk -r -x "core show channels" # Check system resources htop df -h free -h

Performance Optimization

# Optimize Asterisk configuration sudo nano /etc/asterisk/asterisk.conf # Add performance settings: [options] maxcalls = 500 maxload = 0.8 maxfiles = 1000 minmemfree = 2 cache_record_files = yes transmit_silence = yes # Optimize system echo 'vm.swappiness=10' | sudo tee -a /etc/sysctl.conf echo 'vm.vfs_cache_pressure=50' | sudo tee -a /etc/sysctl.conf sudo sysctl -p