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Tip | ||
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| ||
http_port settings are determined by your issued configuration - see the configuration document for more information |
Create Call:
RESTful API |
| WebRTC | ||||||||
---|---|---|---|---|---|---|---|---|---|---|
Code Block | ||||||||||
|
| uri
| '<ext>@<server_url>'
| wsServers
| ['ws://<server_url>:8088/ws']
| authorizationUser: '<ext>',
password : '<ext_password>',
hackIpInContact : true
});
// Call options
var options = {
media: {
constraints: {
audio: true,
video: false
},
render: {
remote: {
audio: document.getElementById('example-audio-tag'),
},
},
},
extraHeaders : [ #SIP headers for Click2Dial Settings
'X-Platforma-Outbound-ID:<outbound_id>'
| 'X-Platforma-Call-Ref:
| '
]
| ;
// Make the call
var session = userAgent.invite('sip:<callee>@<server_url>', options);
| Agent
| <ext_password> Agents Password for the Extension
| , (in the dialer this is the contact_ref, but call_ref is kept for backwards compatibility)
WebRTC | From v3.0.0
From v4.0.0 The Sip Headers changed to X-Teleforge-* |
Values:
<ext> | The agent extension number or the first number connected to the call
|
---|---|
<callee> | The number to dial: One of the following:
|
<outbound_id> | Number to use as caller ID: One of the following:
|
<custom_call_ref> | Your custom call reference specified for this call Alphanumeric and .-_~: Max length 230 characters, thereafter they are truncated Default: if none is provided, then one is generated, although it serves no real purpose |