Dial
Dial
The <Dial> verb connects the current caller to another phone. If the called party picks up, the two parties are connected and can communicate until one hangs up. If the called party does not pick up, if a busy signal is received, or if the number doesn’t exist, the dial verb will finish.
Dial Attributes
| Name | Allowed Values | Default Value |
|---|---|---|
action |
relative or absolute URL |
no default for <Dial> |
method |
GET, POST |
POST |
timeout |
positive integer in seconds |
30 seconds |
timeLimit |
positive integer (seconds) |
14400 seconds (4 hours) |
callerId |
a valid phone number, or client identifier if you are dialing a <Client>. |
Caller’s callerId |
record |
true, false |
false |
dialMusic |
relative or absolute URL |
no default |
-
action. The 'action' attribute takes a URL as an argument. When the dialed call ends, Restcomm will make a GET or POST request to this URL including the parameters below. If you provide an 'action' URL, Restcomm will continue the current call after the dialed party has hung up, using the RCML received in your response to the 'action' URL request. Any RCML verbs occuring after a <Dial> which specifies an 'action' attribute are unreachable. If no 'action' is provided, <Dial> will finish and Restcomm will move on to the next RCML verb in the document. If there is no next verb, Restcomm will end the phone call.
Request Parameters
| Parameter | Description |
|---|---|
DialCallStatus |
The outcome of the <Dial> attempt. See the DialCallStatus section below for details. |
DialCallSid |
The call sid of the new call leg. This parameter is not sent after dialing a conference. |
DialCallDuration |
The duration in seconds of the dialed call. This parameter is not sent after dialing a conference. |
RecordingUrl |
The URL of the recorded audio. This parameter is only sent if record="true" is set on the |
DialCallStatus Values
| Parameter | Description |
|---|---|
completed |
The called party answered the call and was connected to the caller. |
busy |
Restcomm received a busy signal when trying to connect to the called party. |
no-answer |
The called party did not pick up before the timeout period passed. |
failed |
Restcomm was unable to route to the given phone number. This is frequently caused by dialing a properly formated but non-existent phone number. |
canceled |
The call was canceled via the REST API before it was answered. |
-
method. The 'method' attribute takes the value 'GET' or 'POST'. This tells Restcomm whether to request the 'action' URL via HTTP GET or POST. This attribute is modeled after the HTML form 'method' attribute.
-
timeout. The 'timeout' attribute sets the limit in seconds that <Dial> waits for the called party to answer the call.
-
timelimit. The 'timeLimit' attribute sets the maximum duration of the <Dial> in seconds.
-
callerId. The 'callerId' attribute lets you specify the caller ID that will appear to the called party when Restcomm calls. By default, when you put a <Dial> in your RCML response to Restcomm's inbound call request, the caller ID that the dialed party sees is the inbound caller’s caller ID. If you are dialing to a <Client>, you can set a client identifier as the callerId attribute. For instance, if you’ve set up a client for incoming calls and you are dialing to it, you could set the callerId attribute to client:thomas.
-
record. The 'record' attribute lets you specify whether the call will be recorded or not. By default, the call is not recorded. If you set the attribute to 'true' Restcomm will start recording when the two calls are bridged
-
dialMusic. The 'dialMusic' attribute lets you specify a custom RingBack tone to be played. The attribute value is a valid URL pointing to WAV file
| The current release of Restcomm doesn’t support recording for Dial Conference. This feature will be available in the next release |
-
Nesting. You can nest the following nouns within <Dial>: <Number>, <Client>, <Conference>