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v36

latestOpenAPI 3.1.0raw.githubusercontent.com2026-05-1279125449.6 KB
Calls

Voice status callback

Payload sent by SignalWire to your Voice Status Callback URL when the status of a call changes.

Configure this callback using the StatusCallback and StatusCallbackEvent parameters when creating a call via the API or using Dial with Number, SIP, or Conference.

Use StatusCallbackEvent to receive every call progress event — not just the final status — along with current call state and all parameters below.

postWebhookvoiceStatusCallback

Payload

AccountSidstring required

The unique ID of the project this call is associated with.

ApiVersionstring required

The version of the SignalWire API. Incoming calls use the API version placed on the number called. Outgoing calls use the version of the REST API request.

AudioInAveragePtimeinteger required

The average time between packets for the inbound stream.

AudioInDtmfPacketCountinteger required

The number of DTMF packets in the inbound media stream.

AudioInFlushPacketCountinteger required

The number of incoming packets removed from the jitter buffer due to buffer overflow or underflow, causing these packets to be discarded or lost.

AudioInJitterMaxVarianceinteger required

The maximum variance of jitter for incoming audio RTP packets.

AudioInJitterMinVarianceinteger required

The minimum variance of jitter for incoming audio RTP packets.

AudioInLargestJbSizeinteger required

The largest jitter buffer size recorded for incoming audio RTP packets during the session.

AudioInMosstring required

A mean opinion score on a scale of 1–5 that helps determine audio quality.

AudioInMediaPacketCountinteger required

The number of audio packets received in the inbound media stream.

AudioInSkipPacketCountinteger required

Increased with every packet not received at a given ptime. Increases with every lost packet from the inbound stream.

AudioOutDtmfPacketCountinteger required

The total number of DTMF packets generated in the outbound media stream.

AudioOutMediaPacketCountinteger required

The number of audio packets sent in the outbound media stream.

CallbackSourcestring required

The source of the status callback.

CallDurationinteger

The duration, in seconds, of the finished call. Only present on the completed event.

CallerNamestring

The name of the caller. Only available if Caller ID lookup is enabled.

CallSidstring required

A unique identifier for the call.

CallStatus'initiated' | 'ringing' | 'in-progress' | 'queued' | 'failed' | 'busy' | 'no-answer' | 'completed' | 'canceled' required

The status of the call.

StatusDescription
initiatedThe initial dial has begun.
ringingThe call is ringing.
in-progressThe call was answered and is in progress.
queuedThe call is ready and in line to initiate.
failedThe call could not be completed. Usually occurs when the phone number does not exist.
busyThe caller encountered a busy signal.
no-answerThe call ended without an answer.
completedThe call was answered and ended normally.
canceledThe REST API canceled the call while it was ringing or queued.
Direction'outbound-dial' | 'outbound-api' | 'inbound' required

The direction of the call.

  • outbound-dial: calls launched through the Dial verb.
  • outbound-api: calls launched through the REST API.
  • inbound: inbound calls.
ForwardedFromstring

The number this call was forwarded from.

Fromstring required

The phone number that sent this call, in E.164 format.

ParentCallSidstring

A unique identifier for the call that created this call.

RecordingDurationinteger

The duration, in seconds, of the recording.

RecordingSidstring

The unique identifier for the audio recording.

RecordingUrlstring

The URL of the recorded audio call.

SequenceNumberinteger required

The order in which events occur, starting at 0. Although events are fired in order, they each take time and may not appear in the order you expect.

SipInviteResultPhrasestring

A textual description of the result of the SIP INVITE request, indicating success or failure with additional details.

SipResultCodestring

The numeric SIP response code received for the INVITE request, indicating the outcome of the call attempt. See the list of SIP response codes for possible values.

Timestampstring required

The timestamp, in RFC 2822 format, of when the event occurred.

Tostring required

The phone number of the call recipient, in E.164 format.

Example payload

{
  "AccountSid": "b3877c40-da60-4998-90ad-b792e98472af",
  "ApiVersion": "2010-04-01",
  "AudioInAveragePtime": 20,
  "AudioInFlushPacketCount": 2,
  "AudioInJitterMaxVariance": 8,
  "AudioInJitterMinVariance": 1,
  "AudioInLargestJbSize": 4,
  "AudioInMos": "4.3",
  "AudioInMediaPacketCount": 1842,
  "AudioInSkipPacketCount": 1,
  "AudioOutMediaPacketCount": 1840,
  "CallbackSource": "call-progress-events",
  "CallDuration": 94,
  "CallerName": "Jane Smith",
  "CallSid": "b3877c40-da60-4998-90ad-b792e98472ca",
  "CallStatus": "completed",
  "Direction": "outbound-api",
  "ForwardedFrom": "+15550001111",
  "From": "+15551234567",
  "ParentCallSid": "b3877c40-da60-4998-90ad-b792e98472pa",
  "RecordingDuration": 94,
  "RecordingSid": "b3877c40-da60-4998-90ad-b792e984re01",
  "RecordingUrl": "https://example.signalwire.com/api/laml/2010-04-01/Accounts/b3877c40-da60-4998-90ad-b792e98472af/Recordings/b3877c40-da60-4998-90ad-b792e984re01",
  "SequenceNumber": 4,
  "SipInviteResultPhrase": "OK",
  "SipResultCode": "200",
  "Timestamp": "Thu, 04 Apr 2024 18:32:15 +0000",
  "To": "+15559876543"
}

Response

Webhook received