Asterisk IP Telephony Integration

Asterisk IP Telephony Integration on Website

Asterisk is an open-source IP PBX platform, the de-facto standard for corporate telephony. Unlike cloud providers (Mango, Zadarma), Asterisk is deployed on your own server — full control over data, routing, and call costs.

Integration Architecture

Browser (WebRTC/SIP.js) ←→ Asterisk (Kamailio/nginx + WebSocket)
                                    ↓
                              AMI (Asterisk Manager Interface)
                                    ↓
                              Backend API (PHP/Go/Node)
                                    ↓
                              Database + Redis (events)

Asterisk Manager Interface (AMI) — TCP connection to port 5038, through which backend receives real-time events (incoming call, answer, completion) and sends commands (originate, hangup).

Connecting to AMI

// Low-level TCP connection to AMI
$socket = fsockopen(env('ASTERISK_HOST'), 5038, $errno, $errstr, 5);
fwrite($socket, "Action: Login\r\nUsername: api_user\r\nSecret: api_secret\r\n\r\n");

// Receiving events (polling in background process)
while (!feof($socket)) {
    $line = fgets($socket);
    // Parse events: Event: Newchannel, Event: Hangup, etc.
}

In practice, it's more convenient to use a library: optika-si/phpagi or pami/pami (PAMI — PHP Asterisk Manager Interface).

PAMI — Working with Events

// composer require marcelog/pami

$client = new \PAMI\Client\Impl\ClientImpl([
    'host'     => env('ASTERISK_HOST'),
    'port'     => 5038,
    'username' => 'api_user',
    'secret'   => 'api_secret',
    'connect_timeout' => 10
]);

$client->open();

// Listen for incoming calls
$client->registerEventListener(function(\PAMI\Message\Event\EventMessage $event) {
    if ($event->getName() === 'Newchannel') {
        $callerNum = $event->getKey('CallerIDNum');
        $channel   = $event->getKey('Channel');
        // Find customer, notify manager via Redis pub/sub
        Redis::publish('incoming_call', json_encode([
            'caller' => $callerNum,
            'channel' => $channel,
            'timestamp' => now()->toIso8601String()
        ]));
    }
    if ($event->getName() === 'Hangup') {
        $duration = $event->getKey('Duration');
        // Save call to database
    }
});

while (true) {
    $client->process();
    usleep(100000); // 100ms
}

This process runs as a supervisor daemon, always maintaining connection with AMI.

Click-to-Call via Originate

// Initiate call: connect agent with customer
$client->send(new \PAMI\Message\Action\OriginateAction(
    "SIP/{$agentExtension}",  // agent's channel
    "exten,{$customerPhone},default",  // where to call after agent answers
    [
        'Timeout'    => 30000,
        'CallerID'   => '"My Store" <80001234567>',
        'Variable'   => "ORDER_ID={$orderId}",
        'Async'      => true
    ]
));

Asterisk first calls the agent, then after answer connects them to the customer — this is more convenient for the agent than dialing the number themselves.

ARI (Asterisk REST Interface)

Modern approach — ARI (Asterisk REST Interface) via WebSocket. Allows building complex call handling scenarios as code:

// Node.js + ari-client
const ari = require('ari-client');

ari.connect('http://asterisk:8088', 'user', 'secret', function(err, client) {
    client.on('StasisStart', function(event, channel) {
        const callerNum = channel.caller.number;
        // Find customer, play personalized greeting
        channel.play({media: `sound:welcome-${customerId}`});
    });
    client.start('my-stasis-app');
});

WebRTC via Asterisk + Nginx

For browser calls Asterisk must work with WebSocket TLS. Configuration:

; /etc/asterisk/http.conf
[general]
enabled=yes
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/etc/ssl/asterisk/asterisk.crt
tlsprivatekey=/etc/ssl/asterisk/asterisk.key

; /etc/asterisk/pjsip.conf
[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0:8089

Call Recording

Asterisk records calls with MixMonitor. Files are stored in /var/spool/asterisk/monitor/. A script moves them to S3 and updates call_records.recording_url.

Development timeline: 3–5 weeks for full integration: deploying Asterisk, AMI daemon, click-to-call, WebRTC in browser and call history.