Complete Guide: Preparing Asterisk for ARI + Node.js + WebRTC + SIP + AMI Integration

📘 Overview

This guide walks you through setting up Asterisk (v16 or higher) for modern telephony development using the Asterisk REST Interface (ARI) with Node.js, PJSIP for SIP/WebRTC clients, and optional AMI (Asterisk Manager Interface) for monitoring.

By the end, you’ll have:

• A working Asterisk system with ARI and AMI enabled
• Support for both SIP and WebRTC endpoints
• A functional dialplan routing calls into your Node.js app
• A simple Node.js example using node-ari-client

---

⚙️ 1. Prerequisites

Before you start, ensure you have:

• Asterisk installed (apt install asterisk or source build)
• Node.js (v16+)
• At least one SIP softphone (e.g., Zoiper, Linphone)
• Optional: a browser WebRTC client (SIP.js, SIPML5)
• Proper firewall/NAT configuration for SIP, RTP, and WebSocket ports

---

🌐 2. Configure HTTP for ARI & WebRTC

File: /etc/asterisk/http.conf

[general]
enabled       = yes
bindaddr      = 0.0.0.0
bindport      = 8088

; TLS (HTTPS/WSS) for WebRTC
tlsenable     = yes
tlsbindaddr   = 0.0.0.0:8089
tlscertfile   = /etc/asterisk/keys/asterisk.crt
tlsprivatekey = /etc/asterisk/keys/asterisk.key

Notes:

• Create self-signed certs for testing:

cd /etc/asterisk/keys
ast_tls_cert -C yourdomain.com -O "Asterisk" -d /etc/asterisk/keys

Port 8089 (TLS) is used for secure WebSocket (wss://) signaling in WebRTC.

🔑 3. Enable ARI (REST Interface)

File: /etc/asterisk/ari.conf

[general]
enabled = yes
pretty = yes
allowed_origins = http://localhost:3000, https://myapp.example.com

[ariuser]
type = user
read_only = no
password = strongpassword

Notes:

• allowed_origins defines which web apps can connect (CORS).
• Credentials here are used by your Node.js app.

📡 4. Enable AMI (Manager Interface)

File: /etc/asterisk/manager.conf

[general]
enabled = yes
webenabled = yes
port = 5038
bindaddr = 127.0.0.1  ; restrict to localhost for security

[amiuser]
secret = amisecret
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user

Notes:

• AMI is optional, mainly used for monitoring and legacy automation.
• Limit AMI access to localhost or secure network only.

5. Configure SIP & WebRTC Endpoints (PJSIP)

File: /etc/asterisk/pjsip.conf

[global]
type = global
user_agent = MyAsteriskPBX/20.0

; TRANSPORTS
[transport-udp]
type     = transport
protocol = udp
bind     = 0.0.0.0:5060

[transport-wss]
type     = transport
protocol = wss
bind     = 0.0.0.0:8089

; === WebRTC Client (1000) ===
[1000_auth]
type = auth
auth_type = userpass
username = 1000
password = 1000pass

[1000_aor]
type = aor
max_contacts = 1

[1000_endpoint]
type = endpoint
transport = transport-wss
context = webrtc-context
disallow = all
allow = opus,ulaw,alaw
ice_support = yes
rtcp_mux = yes
rtp_symmetric = yes
media_use_received_transport = yes
dtls_auto_generate_cert = yes
dtls_setup = actpass
aors = 1000_aor
auth = 1000_auth
direct_media = no

; === SIP Phones (2000 & 2001) ===
[2000_auth]
type = auth
auth_type = userpass
username = 2000
password = 2000pass

[2000_aor]
type = aor
max_contacts = 2

[2000_endpoint]
type = endpoint
transport = transport-udp
context = internal
disallow = all
allow = ulaw,alaw
aors = 2000_aor
auth = 2000_auth
direct_media = no

[2001_auth]
type = auth
auth_type = userpass
username = 2001
password = 2001pass

[2001_aor]
type = aor
max_contacts = 2

[2001_endpoint]
type = endpoint
transport = transport-udp
context = internal
disallow = all
allow = ulaw,alaw
aors = 2001_aor
auth = 2001_auth
direct_media = no

Notes:

• transport-wss allows WebRTC clients over wss://.
• Codecs opus, ulaw, and alaw ensure broad compatibility.
• Each endpoint maps to a specific context in your dialplan

6. Dialplan Configuration

File: /etc/asterisk/extensions.conf

[internal]
exten => 2000,1,NoOp(2000 Calling 2001)
    same => n,Dial(PJSIP/2001,20)
    same => n,Hangup()

exten => 2001,1,NoOp(2001 Calling 2000)
    same => n,Dial(PJSIP/2000,20)
    same => n,Hangup()

; Route call into ARI app
exten => 4000,1,NoOp(SIP phone enters ARI)
    same => n,Answer()
    same => n,Stasis(my_app)
    same => n,Hangup()

[webrtc-context]
exten => 3000,1,NoOp(WebRTC call into ARI)
    same => n,Stasis(my_app)
    same => n,Hangup()

Explanation:SIP phones can call each other on 2000 and 2001.Dialing 4000 or 3000 routes the call into the ARI app my_app.💻 7. Node.js ARI Application ExampleFile: app.js

'use strict';
const ari = require('ari-client');

const url = 'http://localhost:8088';
const user = 'ariuser';
const pass = 'strongpassword';

ari.connect(url, user, pass, clientLoaded);

function clientLoaded(err, client) {
  if (err) throw err;

  client.on('StasisStart', (event, channel) => {
    console.log(`Channel ${channel.name} entered Stasis`);
    channel.answer()
      .then(() => channel.play({ media: 'sound:demo-congrats' }))
      .then(() => channel.hangup())
      .catch(console.error);
  });

  client.start('my_app');
}

Run the app

npm init -y
npm install ari-client
node app.js

Test:

• From your SIP/WebRTC client, dial 3000 or 4000.
• The Node.js app answers, plays “demo-congrats”, then hangs up.

---

🔐 8. Security Tips

• Use strong passwords for ARI/AMI/PJSIP.
• Restrict network access (use 127.0.0.1 where possible).
• Enable TLS for WebRTC/WSS.
• Use firewalls or iptables to restrict ports.
• For production, use Nginx as a reverse proxy for ARI/WebRTC signaling.

---

🚀 9. Verify Setup

Run the following in Asterisk CLI:

Component	Purpose	Config File
HTTP / WebSocket	REST + WebRTC Transport	/etc/asterisk/http.conf
ARI	Node.js REST Control	/etc/asterisk/ari.conf
AMI	Legacy Manager Interface	/etc/asterisk/manager.conf
PJSIP	SIP & WebRTC Endpoints	/etc/asterisk/pjsip.conf
Dialplan	Call Logic & Routing	/etc/asterisk/extensions.conf
Node.js App	Call Control via ARI	app.js