Astproxy is a proxy between Asterisk PBX system and your application. It provides the possibility to send commands and to receive events from Asterisk offering an abstraction layer of data to always remain independent from the specific version of the PBX and to offer the possibiity to extend its functionalities to even support other type of PBX systems.
Node.js v10 LTS (10.19.0) or later.
This is a Node.js module available through the npm registry.
Before installing, download and install Node.js.
Install with npm:
npm i astproxy --save
- Install the executable:
npm install astproxy- Include the library:
const astproxy = require('astproxy');- Configure
astproxy
astproxy uses Asterisk AMI to communicate with Asterisk, so it needs some information to connect to it:
astproxy.config({
port: 5038,
host: 'localhost',
username: '<USERNAME>',
password: '<PASSWORD>',
reconnect: true,
reconnect_after: 3000
});The credentials are usually present into /etc/asterisk/manager.conf.
- Register for the event
ready
The event ready is emmited when the connection with Asterisk has been successfully established. Before using the astproxy you always have to register for this event:
let ready = () => {
// astproxy is ready to be used
// your code here
};
astproxy.on('ready', ready);- Start the module:
astproxy.start();- Configuration by JSON files
- Send command to Asterisk
- Receive events from Asterisk
- Provides all the JSON data about:
extensionsqueuesparkingscall conversations
- Runtime reloading
An example of use cases could be:
- originate a new call, answer to an incoming call, hangup a call
- obtain all the extensions status: online, busy, ringing, offline, dnd, ...
- obtain information about all phone conversations
These are the configuration options used by astproxy to connect to Asterisk AMI:
- port: port number for Asterisk AMI (default
5038) - host: host of Asterisk (default
localhost) - username: username of Asterisk AMI user (default
username) - password: password of Asterisk AMI user (default
password) - reconnect: do you want the ami to reconnect if the connection is dropped (default
false) - reconnect_after: how long to wait to reconnect, in miliseconds (default
3000) - prefix: a prefix number to be used for all phone calls (optional)
Example:
const astproxy = require('astproxy');
astproxy.config({
port: 5038,
host: 'localhost',
username: 'admin',
password: '0123456789',
reconnect: true,
reconnect_after: 3000
});The credentials are usually contained into /etc/asterisk/manager.conf.
You can obtain the events list with astproxy.EVENTS.
ready: is emitted once the asterisk connection has been established. You always have to register for this event before doing any operation.
const astproxy = require('astproxy');
astproxy.config({
port: 5038,
host: 'localhost',
username: '<USERNAME>',
password: '<PASSWORD>',
reconnect: true,
reconnect_after: 3000
});
astproxy.on('ready', () => {}); // or astproxy.on(asproxy.EVENTS.EVT_READY, () => {});
astproxy.start();extenHangupextenDialing
Data about all the extensions:
let extensions = astproxy.getExtensions();Example of returned data:
{
"2001": { // extension identifier
"ip": "192.168.5.163", // the ip address of the phone registered with this extension
"cf": "221", // the call forward status. If it's disabled, it is an empty string
"cfb": "221", // the call forward on busy status. If it's disabled, it is an empty string
"cfu": "221", // the call forward on unavailable status. If it's disabled, it is an empty string
"dnd": false, // it's true if the don't disturb is active
"cfVm": "", // the call forward to voicemail status. If it's disabled, it is an empty string
"cfbVm": "", // the call forward on busy to voicemail status. If it's disabled, it is an empty string
"cfuVm": "", // the call forward on unavailable to voicemail status. If it's disabled, it is an empty string
"port": "5062",
"name": "Alessandro",
"exten": "214",
"status": "online", // the status can be: "dnd", "busy", "online", "onhold", "offline", "ringing", "busy_ringing"
"context": "from-internal", // the context
"useWebsocket": false, // if the extension use websocket
"sipuseragent": "Twinkle/1.4.2",
"conversations": { // the keys are the conversation identifiers
"SIP/214-000002f4>SIP/209-000002f5": {
"id": "SIP/214-000002f4>SIP/209-000002f5",
"owner": "214",
"chDest": { ... }, // the source channel of the call
"queueId": "401", // the queue identifier if the conversation has gone through a queue
"chSource": { ... }, // the destination channel of the call
"duration": 26,
"recording": "false", // it's "true" or "mute" if the conversation is recording, "false" otherwise
"direction": "in",
"inConference": true, // if the conversation involves a meetme conference
"throughQueue": true, // if the call has gone through a queue
"counterpartNum": "209",
"counterpartName": "user"
}
}
},
...
}let callCb = (err, resp) => {
// callback function
};
astproxy.call({
endpointId: '2001', // the extension identifier: the caller
to: '2002', // the destination number of the call
cb: callCb // the callback function
});
React
Typescript
Django
Laravel
Facebook
Microsoft
Google
Alibaba
D3
Tencent