asterisk 0.2.1

Idiomatic bindings to Asterisk's REST interface, allowing calls to be managed in Dart.

Asterisk

Asterisk is "an open-source framework for building communications applications". package:asterisk is a package for writing these communication applications in Dart!

Asterisk allows building powerful communication systems. By providing an abstraction over channels, bridges and endpoints and supporting a wide range of protocols including VoIP (through SIP and RTP), it is a great choice for all kinds of applications dealing with phone calls. How calls are handled in Asterisk is defined in a dialplan, a text file which can get complex to write and maintain in growing setups. And while Asterisk supports a large number of builtin applications, these are sometimes hard to setup and understand, or don't support the feature you need.

Fortunately, Asterisk contains a RESTful interface, which can be used to interact with calls. This enables you to write your own communication apps in a language you already know.

Features

package:asterisk provides high-level Dart bindings to both the Asterisk REST interface as well as the web-socket notification mechanism for events.

Here are just some of the things you can build with this package:

• An automated voicemail system.
• Voice-based timers.
• Call-Roulette!
• Your very own telephone conference setup.
• A customer service system, with hold queues and everything.

Some of these have already been implemented in examples/ - go check them out!

However, note that this package doesn't support real-time interaction with calls. You can record calls and play sounds, but this isn't a softphone. Of course, you could use Flutter WebRTC to write a voice-enabled Flutter app that can be called through an Asterisk managed by this package.

Getting started

Using this package requires a running Asterisk server. To interact with the actual phone network, you need a VoIP provider that server can talk to.

Local testing

For testing, you can emulate a local phone network, placing calls in a web browser thanks to WebRTC. To get started, run

docker run -ti --network host ghcr.io/simolus3/asterisk_demo:latest

Starting asterisk will print a lot of errors as it tires to load a lot of modules we don't need - once it prints "Asterisk Ready.", the system is ready to go.

Visit http://localhost:8088/. When opening the website for the first time, a settings screen will be shown. You need to enter a SIP username under "Account" - the server will accept 201, 202, 203, 204 or 205. They all use demo as their passsword and are reachable by dialing their account number.

Next, run an example to call:

dart run example/main.dart

All examples are reachable in the demo server by dialing 1.

Usage

This simple Asterisk application that accepts incoming calls, announces the caller id on the channel and then hangs up:

import 'package:asterisk/asterisk.dart';

void main() async {
  final asterisk = Asterisk(
    baseUri: Uri.parse('http://localhost:8088'),
    applicationName: 'demo',
    username: 'demoapp',
    password: 'demo',
  );
  await for (final incoming in asterisk.stasisStart) {
    _announceId(incoming.channel);
  }
}

Future<void> _announceId(LiveChannel channel) async {
  print('Has incoming call from ${channel.channel.caller}');

  await channel.answer();
  await Future.delayed(const Duration(seconds: 1));

  final playback = await channel
      .play(sources: [MediaSource.digits(channel.channel.caller.number)]);
  // Wait for the playback to finish.
  await playback.events.where((e) => e is PlaybackFinished).first;

  if (!channel.isClosed) {
    await channel.hangUp();
  }
}