Step 1: Choose Your VoIP Server Software
Asterisk is a popular choice for building VoIP systems due to its flexibility and wide range of features. Before proceeding, ensure you have the necessary hardware and network infrastructure to support your VoIP server.
Step 2: Prepare Your Hardware
Server Hardware
- Server: A reliable server with sufficient CPU, RAM, and storage to handle the expected call volume.
- Network : A robust network setup with sufficient bandwidth and low latency.
VoIP Devices
- VoIP Phones or Softphones: Devices or applications that will be used by clients to make and receive calls.
Step 3: Install the Operating System
Choose an OS
Asterisk can run on various operating systems, but it is commonly deployed on Linux distributions such as CentOS, Ubuntu, or Debian. For this guide, we will use Ubuntu.
Install the OS
Perform a clean installation of your chosen Linux distribution on your server hardware.
Step 4: Install VoIP Server Software (Asterisk)
Update the System
First, ensure your system is up to date:
sudo apt update && sudo apt upgrade
Install Dependencies
Install the necessary dependencies for Asterisk:
sudo apt install build-essential libxml2-dev libsqlite3-dev uuid-dev
Download and Install Asterisk
1. Download Asterisk:
cd /usr/src sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18-current.tar.gz
2. Extract the Tarball:
sudo tar -zxvf asterisk-18-current.tar.gz cd asterisk-18.*
3. Compile and Install Asterisk:
sudo ./configure sudo make sudo make install sudo make samples sudo make config sudo ldconfig
Start Asterisk
Start the Asterisk service and enable it to run on boot:
sudo systemctl start asterisk sudo systemctl enable asterisk
Step 5: Configure VoIP Server
Basic Configuration
SIP Configuration
Edit the sip.conf file to define SIP settings:
sudo nano /etc/asterisk/sip.conf
Add the following example configuration:
[general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes [1001] type=friend context=internal host=dynamic secret=password1
Extensions Configuration
Edit the extensions.conf file to define call routing and dial plans:
sudo nano /etc/asterisk/extensions.conf
Add the following example configuration:
[internal] exten => 1001,1,Dial(SIP/1001) exten => 1002,1,Dial(SIP/1002)
User Accounts
Create user accounts for each VoIP phone or softphone client by editing the 'sip.conf' and 'extensions.conf' files.
Step 6: Connect VoIP Phones
Softphones
Download and install softphone applications like Zoiper or Linphone on client devices.
Configuration
Use the account details created in 'sip.conf' to set up each client device.
Step 7: Test the VoIP Server
Make Test Calls
Ensure that clients can register with the server and make internal and external calls.
Check Logs
Monitor server logs for errors and resolve any issues.
Step 8: Secure Your VoIP Server
Firewall
Configure your firewall to allow VoIP traffic (e.g., port 5060 for SIP).
Fail2Ban
Install Fail2Ban to protect against brute-force attacks.
Encryption
Enable SRTP (Secure Real-time Transport Protocol) for encrypting voice traffic.
Step 9: Monitor and Maintain
Monitoring Tools
Use tools like Nagios or Zabbix to monitor server performance and call quality.
Regular Updates
Keep the OS and VoIP software updated with the latest security patches.
Optional: Advanced Features
Voicemail
Configure voicemail settings for users.
IVR
Set up Interactive Voice Response menus.
Call Recording
Enable call recording for quality control or legal purposes.