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Asterisk is a powerful open-source platform that enables developers to create voice applications such as VoIP systems, conference bridges, and call centers. In this tutorial, we’ll guide you through building a web-to-phone solution using Asterisk. This system allows users to make phone calls directly from a web interface.
Before we begin, make sure you have the following:
If Asterisk is not already installed on your server, follow these steps:
sudo apt update && sudo apt upgrade -y
sudo apt install wget build-essential subversion libjansson-dev libxml2-dev libncurses5-dev
cd /usr/src
sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20-current.tar.gz
sudo tar -zxvf asterisk-20-current.tar.gz
cd asterisk-20*/
sudo ./configure
sudo make
sudo make install
sudo make samples
sudo make config
sudo systemctl start asterisk
sudo systemctl enable asterisk
Check the installation:
sudo asterisk -rvv
WebRTC is essential for enabling web-to-phone communication.
sip.conf
Edit the SIP configuration file:
sudo nano /etc/asterisk/sip.conf
Add the following:
[general]
context=default
udpbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=udp,tcp,ws,wss
[webrtc_client]
type=friend
host=dynamic
secret=your_password
context=from-internal
encryption=yes
avpf=yes
force_avp=yes
dtlsenable=yes
dtlsverify=fingerprint
dtlsrekey=60
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem
dtlssetup=actpass
Install OpenSSL and generate certificates for DTLS (required for WebRTC):
sudo mkdir /etc/asterisk/keys
cd /etc/asterisk/keys
sudo openssl genrsa -out asterisk.key 2048
sudo openssl req -new -x509 -key asterisk.key -out asterisk.crt -days 365
sudo cat asterisk.crt asterisk.key > asterisk.pem
extensions.conf
Edit the dial plan file:
bashCopy codesudo nano /etc/asterisk/extensions.conf
Add a simple plan to route calls:
[from-internal]
exten => _X.,1,Dial(SIP/${EXTEN}@your_sip_provider)
exten => _X.,n,Hangup
Replace your_sip_provider
with your VoIP provider’s SIP trunk name.
Use a WebRTC-compatible library like SIP.js to create a web interface.
<!DOCTYPE html>
<html>
<head>
<title>Web to Phone</title>
<script src="https://sipjs.com/releases/0.15.10/sip.min.js"></script>
</head>
<body>
<h1>Web-to-Phone</h1>
<input type="text" id="phone-number" placeholder="Enter Phone Number">
<button id="call-btn">Call</button>
<script>
const userAgent = new SIP.UserAgent({
uri: 'sip:webrtc_client@your-server-ip',
transportOptions: {
wsServers: 'wss://your-server-ip:8089/ws'
},
authorizationUsername: 'webrtc_client',
authorizationPassword: 'your_password'
});
userAgent.start();
document.getElementById('call-btn').addEventListener('click', () => {
const phoneNumber = document.getElementById('phone-number').value;
userAgent.invite('sip:' + phoneNumber + '@your-server-ip');
});
</script>
</body>
</html>
Replace your-server-ip
with your public server IP or domain.
Edit the http.conf
file to enable the Asterisk WebSocket server:
bashCopy codesudo nano /etc/asterisk/http.conf
Add the following:
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8089
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlsprivatekey=/etc/asterisk/keys/asterisk.pem
Restart Asterisk:
sudo systemctl restart asterisk
python3 -m http.server
sudo asterisk -rvv
Asterisk, combined with WebRTC, is a powerful tool for creating a web-to-phone system. By following the steps outlined in this blog, you can set up a basic solution that bridges web interfaces and traditional phone systems.
Feel free to expand on this by integrating features like voicemail, IVR, or SMS.
Happy coding!
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