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.
Retrieves one call by SID so you can inspect a known call’s progress and outcome. Use List all Calls to discover SIDs. The Calling API is command-oriented and uses its own call identifiers.
Permissions
The API token used to authenticate must have the following scope(s) enabled to make a successful request: Voice.
SignalWire Basic Authentication using Project ID and API Token.
The client sends HTTP requests with the Authorization header containing
the word Basic followed by a space and a base64-encoded string of project_id:token.
The project ID will be used as the username and the API token as the password.
Example:
Authorization: Basic base64(project_id:token)
Path parameters
AccountSidstringRequiredformat: "uuid"
The Project ID that uniquely identifies the Account.
SidstringRequiredformat: "uuid"
The unique identifier for the call.
Response
The request has succeeded.
sidstringformat: "uuid"
The unique identifier for the call.
account_sidstringformat: "uuid"
The unique identifier for the account that created this call.
date_createdstring
The date, in RFC 2822 GMT format, this call was created.
date_updatedstring
The date, in RFC 2822 GMT format, this call was updated.
parent_call_sidstring or nullformat: "uuid"
The unique identifier for the call that created this call.
tostring
The address that received the call.
formatted_tostring
The formatted number that received the call.
to_formattedstring
The formatted number that received the call. Alias for formatted_to.
fromstring
The address that initiated the call.
formatted_fromstring
The formatted number that initiated the call.
from_formattedstring
The formatted number that initiated the call. Alias for formatted_from.
phone_number_sidstring or nullformat: "uuid"
The unique identifier for the phone number.
statusenum
The status of the call.
start_timestring or null
The time, in RFC 2822 GMT format, the call started ringing. This is before the call was answered. Null until the call starts.
end_timestring or null
The time, in RFC 2822 GMT format, on which the call was terminated. Null until the call ends.
durationinteger
The duration, in seconds, of the call: end_time minus start_time, rounded up to the next whole second.
It includes ringing time, so it’s longer than the time the call was connected and usually longer than a
recording of the call. For a call still in progress, it’s the time elapsed so far; before the call starts, it’s 0.
pricefloat or null
The charge for the call.
price_unitstring
The currency, in ISO 4127 format, for the price of the call.
directionenum
The direction of the call.
inbound: inbound calls.
outbound-api: calls launched through the REST API.
outbound-dial: calls launched through the Dial verb.
Allowed values:
answered_byenum or null
Who/what the call was answered by.
Allowed values:
api_versionstring
The version of the SignalWire API.
forwarded_fromstring or null
The number this call was forwarded from. Always null.
caller_namestring or null
The caller name. Always null.
uristring
The URI for the call.
subresource_urisobject
A Map of available sub-resources.
annotationstring or null
The annotation for the call. Always null.
group_sidstring or null
The group SID for the call. Always null.
audio_in_mosstring or null
The estimated Mean Opinion Score (MOS) for inbound audio quality, based on the share of incoming audio packets that arrived without loss or delay. It ranges from about 1 to 4.5, the score for a call with no lost or late packets. Returned as a decimal string. Null if the call has no quality measurement.
sip_result_codestring or null
The SIP result code for the call.
audio_rtt_avgdouble or null
The average round-trip time for audio in milliseconds, in steps of 10.
audio_rtt_mindouble or null
The minimum round-trip time for audio in milliseconds, in steps of 10.
audio_rtt_maxdouble or null
The maximum round-trip time for audio in milliseconds, in steps of 10.
audio_in_jitter_mindouble or null
The smallest variance measured in the time between incoming audio packets, in milliseconds squared.
audio_in_jitter_maxdouble or null
The largest variance measured in the time between incoming audio packets, in milliseconds squared.
audio_out_jitter_mindouble or null
The minimum outbound audio jitter in milliseconds.
audio_out_jitter_maxdouble or null
The maximum outbound audio jitter in milliseconds.
audio_out_jitter_avgdouble or null
The average outbound audio jitter in milliseconds.
audio_in_media_packet_countinteger or null
The number of audio packets received in the inbound media stream.
audio_out_packet_countinteger or null
Not populated. Always null; use audio_out_media_packet_count for the number of audio packets sent.
audio_out_media_packet_countinteger or null
The number of audio packets sent in the outbound media stream.
audio_out_lostinteger or null
The number of outbound audio packets lost.
audio_in_mean_intervaldouble or null
The mean interval between inbound audio packets in milliseconds.
audio_in_dtmf_packet_countinteger or null
The number of DTMF packets in the inbound media stream.
audio_out_dtmf_packet_countinteger or null
The number of DTMF packets in the outbound media stream.
audio_in_skip_packet_countinteger or null
The number of inbound audio intervals filled with comfort noise because no audio packet was available. It grows with lost or late packets and during DTMF digits.
audio_in_flush_packet_countinteger or null
The number of queued inbound packets discarded when the media stream was flushed, for example when the call was bridged or unbridged.
audio_in_largest_jb_sizeinteger or null
Not populated. Always 0 when the call has quality statistics.
