How to config /phone/scripts/config.js
// 주식회사 얼쑤팩토리의 makecall.io 신청 후 발급받은 정보를 입력합니다.
var user = {
// SIP 사용자ID
"User" : "sipuser",
// SIP 비밀번호
"Pass" : "yourpassword",
// SIP Auth Realm
"Realm" : "realm.olssoo.com",
// 발신자 표시이름
"Display" : "olssoo",
// WebSocket 서버 URL
"WSServer" : "wss://wss.olssoo.com:8443"
};
A Javascript SIP client based on SIP.js.
ctxSip is a Javascript based SIP client that uses WebRTC and WebSockets to connect to your SIP server. The UI is designed to be launched as a popup from within your application. Works well with Kazoo from 2600hz
- Audio only, Hold / Resume, Mute, multiple call support.
- No plugins required, Works with WebSocket / WebRTC enabled browsers. (Firefox & Chrome.)
- Call log is saved to localStorage.
- Intuitive interface makes it easy for users.
- Easy to configure and integrate into your project.
- MIT licensed.
You will need a sip account on a server that supports SIP over websockets. This has been tested with Kamailio in front of Freeswitch.
- Clone this project.
- Copy
phone/scripts/config-sample.js
tophone/scripts/config.js
- Edit
phone/scripts/config.js
to reflect your sip account. - In your application HMTL, create a document and add the following code:
<a href="phone" id="launchPhone">Launch</a>
<script>
var url = '/phone',
features = 'menubar=no,location=no,resizable=no,scrollbars=no,status=no,addressbar=no,width=320,height=480';
$('#launchPhone').on('click', function(event) {
event.preventDefault();
// This is set when the phone is open and removed on close
if (!localStorage.getItem('ctxPhone')) {
window.open(url, 'ctxPhone', features);
return false;
} else {
window.alert('Phone already open.');
}
});
</script>
SSL connections for are required for this to work!
ctxSip uses:
Tested on:
Translate some menus and titles