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.
Payload
The unique ID of the project this call is associated with.
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.
The average time between packets for the inbound stream.
The number of DTMF packets in the inbound media stream.
The number of incoming packets removed from the jitter buffer due to buffer overflow or underflow, causing these packets to be discarded or lost.
The maximum variance of jitter for incoming audio RTP packets.
The minimum variance of jitter for incoming audio RTP packets.
The largest jitter buffer size recorded for incoming audio RTP packets during the session.
A mean opinion score on a scale of 1–5 that helps determine audio quality.
The number of audio packets received in the inbound media stream.
Increased with every packet not received at a given ptime. Increases with every lost packet from the inbound stream.
The total number of DTMF packets generated in the outbound media stream.
The number of audio packets sent in the outbound media stream.
The source of the status callback.
The duration, in seconds, of the finished call. Only present on the completed event.
The name of the caller. Only available if Caller ID lookup is enabled.
A unique identifier for the call.
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 number this call was forwarded from.
The phone number that sent this call, in E.164 format.
A unique identifier for the call that created this call.
The duration, in seconds, of the recording.
The unique identifier for the audio recording.
The URL of the recorded audio call.
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.
A textual description of the result of the SIP INVITE request, indicating success or failure with additional details.
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.
The timestamp, in RFC 2822 format, of when the event occurred.
The phone number of the call recipient, in E.164 format.
Response
Webhook received