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.
Dial Action Request HTTP Parameters
The Restcomm parameters passed to your application in its asynchronous request to the Dial action URL. The full list of parameters can be found in the RCML Voice Request documentation.
The Status Callback request also passes these additional 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. |
DialRingDuration |
The duration in seconds of the dialed call ringing time. 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 |
ReferTarget |
|
Transferor |
|
Transferee |
|
FailureReason (optional) |
Provides more information about failure reason where Dial Number/Sip/Client. This parameter is optional and will only be present when call status is Failed. |
SipHeader_* |
A set of optional field with custom SIP headers that are present in INVITE SIP request or in a last SIP response |
DialSipCallId |
Optional parameter: sipCallId from a response in a last SIP response if a last SIP response code >=400 |
DialSipResponseCode |
Optional parameter: statusCode in a last SIP response if a last SIP response code >=400 |
DialSipHeader_* |
A set of optional field with custom SIP headers that are present in a last SIP response |
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>
Examples
For examples of how to use the <Dial> verb see below.
Dial a phone number and record the call.
<Response>
<Dial record="true">1-444-555-6666</Dial>
</Response>
Dial a phone number with set callerId and record the call.
<Response>
<Dial callerId="15556667777" record="true">1-444-555-6666</Dial>
</Response>
Sequential Dial
You may want dial phone numbers, clients, SIP URIs and conference rooms in a sequence. In that case, you can specify a timeout limit during which a single dial will last. Optionally you can set a <Pause> between the dials.
In the example below, every dial lasts for 30 seconds. In addition, there is a pause of 4 seconds in between. If the first dialed party does not pick up, the sequence continues to the second one and so on, till one of the parties answers the call.
<Response>
<Say>This is a sequential Dial example.</Say>
<Dial timeout="30">
<Number>12223334444</Number>
</Dial>
<Pause length="4"/>
<Dial timeout="30">
<Number>15556667777</Number>
</Dial>
<Pause length="4"/>
<Dial>
<Client>jane</Client>
</Dial>
</Response>
