For AI agents: a documentation index is available at the root level at /llms.txt. Append /llms.txt to any URL for a page-level index, or .md for the markdown version of any page.
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](/docs/compatibility-api/rest/calls/create-a-call) or using
[Dial](/docs/compatibility-api/cxml/reference/voice/dial) with
[Number](/docs/compatibility-api/cxml/reference/voice/number),
[SIP](/docs/compatibility-api/cxml/reference/voice/sip), or
[Conference](/docs/compatibility-api/cxml/reference/voice/conference).
Use `StatusCallbackEvent` to receive every call progress event — not just the final status —
along with current call state and all parameters below.
Payload sent by SignalWire to your Voice Status Callback URL when the status of a call changes.
Use StatusCallbackEvent to receive every call progress event — not just the final status —
along with current call state and all parameters below.
Payload
The payload of this webhook request is an object.
AccountSidstringRequired
The unique ID of the project this call is associated with.
ApiVersionstringRequired
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.
AudioInAveragePtimeintegerRequired
The average time between packets for the inbound stream.
AudioInDtmfPacketCountintegerRequired
The number of DTMF packets in the inbound media stream.
AudioInFlushPacketCountintegerRequired
The number of incoming packets removed from the jitter buffer due to buffer overflow or underflow,
causing these packets to be discarded or lost.
AudioInJitterMaxVarianceintegerRequired
The maximum variance of jitter for incoming audio RTP packets.
AudioInJitterMinVarianceintegerRequired
The minimum variance of jitter for incoming audio RTP packets.
AudioInLargestJbSizeintegerRequired
The largest jitter buffer size recorded for incoming audio RTP packets during the session.
AudioInMosstringRequired
A mean opinion score on a scale of 1–5 that helps determine audio quality.
AudioInMediaPacketCountintegerRequired
The number of audio packets received in the inbound media stream.
AudioInSkipPacketCountintegerRequired
Increased with every packet not received at a given ptime.
Increases with every lost packet from the inbound stream.
AudioOutDtmfPacketCountintegerRequired
The total number of DTMF packets generated in the outbound media stream.
AudioOutMediaPacketCountintegerRequired
The number of audio packets sent in the outbound media stream.
CallbackSourcestringRequired
The source of the status callback.
CallSidstringRequired
A unique identifier for the call.
CallStatusenumRequired
The status of the call.
| Status | Description |
|--------|-------------|
| `initiated` | The initial dial has begun. |
| `ringing` | The call is ringing. |
| `in-progress` | The call was answered and is in progress. |
| `queued` | The call is ready and in line to initiate. |
| `failed` | The call could not be completed. Usually occurs when the phone number does not exist. |
| `busy` | The caller encountered a busy signal. |
| `no-answer` | The call ended without an answer. |
| `completed` | The call was answered and ended normally. |
| `canceled` | The REST API canceled the call while it was ringing or queued. |
DirectionenumRequired
The direction of the call.
- `outbound-dial`: calls launched through the Dial verb.
- `outbound-api`: calls launched through the REST API.
- `inbound`: inbound calls.
Allowed values:
FromstringRequired
The phone number that sent this call, in E.164 format.
SequenceNumberintegerRequired
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.
TimestampstringRequired
The timestamp, in RFC 2822 format, of when the event occurred.
TostringRequired
The phone number of the call recipient, in E.164 format.
CallDurationintegerOptional
The duration, in seconds, of the finished call. Only present on the completed event.
CallerNamestringOptional
The name of the caller. Only available if Caller ID lookup is enabled.
ForwardedFromstringOptional
The number this call was forwarded from.
ParentCallSidstringOptional
A unique identifier for the call that created this call.
RecordingDurationintegerOptional
The duration, in seconds, of the recording.
RecordingSidstringOptional
The unique identifier for the audio recording.
RecordingUrlstringOptional
The URL of the recorded audio call.
SipInviteResultPhrasestringOptional
A textual description of the result of the SIP INVITE request, indicating success or failure with additional details.
SipResultCodestringOptional
The numeric SIP response code received for the INVITE request, indicating the outcome of the call attempt. See the [list of SIP response codes](https://en.wikipedia.org/wiki/List_of_SIP_response_codes) for possible values.
Response
200
Webhook received
The status of the call.
Status
Description
initiated
The initial dial has begun.
ringing
The call is ringing.
in-progress
The call was answered and is in progress.
queued
The call is ready and in line to initiate.
failed
The call could not be completed. Usually occurs when the phone number does not exist.
busy
The caller encountered a busy signal.
no-answer
The call ended without an answer.
completed
The call was answered and ended normally.
canceled
The REST API canceled the call while it was ringing or queued.
The direction of the call.
outbound-dial: calls launched through the Dial verb.
outbound-api: calls launched through the REST API.
inbound: inbound calls.
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.
