JavaScript

import Telnyx from 'telnyx';

const client = new Telnyx({
  apiKey: 'My API Key',
});

const response = await client.calls.dial({
  connection_id: '7267xxxxxxxxxxxxxx',
  from: '+18005550101',
  to: '+18005550100 or sip:username@sip.telnyx.com',
});

console.log(response.data);

{
  "data": {
    "call_control_id": "v3:MdI91X4lWFEs7IgbBEOT9M4AigoY08M0WWZFISt1Yw2axZ_IiE4pqg",
    "call_leg_id": "2dc6fc34-f9e0-11ea-b68e-02420a0f7768",
    "call_session_id": "2dc1b3c8-f9e0-11ea-bc5a-02420a0f7768",
    "client_state": "aGF2ZSBhIG5pY2UgZGF5ID1d",
    "is_alive": false,
    "call_duration": 50,
    "record_type": "call",
    "recording_id": "d7e9c1d4-8b2a-4b8f-b3a7-9a671c9e9b0a",
    "start_time": "2019-01-23T18:10:02.574Z",
    "end_time": "2019-01-23T18:11:52.574Z"
  }
}

Dial

Dial a number or SIP URI from a given connection. A successful response will include a call_leg_id which can be used to correlate the command with subsequent webhooks.

Expected Webhooks:

call.initiated
call.answered or call.hangup
call.machine.detection.ended if answering_machine_detection was requested
call.machine.greeting.ended if answering_machine_detection was requested to detect the end of machine greeting
call.machine.premium.detection.ended if answering_machine_detection=premium was requested
call.machine.premium.greeting.ended if answering_machine_detection=premium was requested and a beep was detected
streaming.started, streaming.stopped or streaming.failed if stream_url was set

When the record parameter is set to record-from-answer, the response will include a recording_id field.

POST

calls

JavaScript

import Telnyx from 'telnyx';

const client = new Telnyx({
  apiKey: 'My API Key',
});

const response = await client.calls.dial({
  connection_id: '7267xxxxxxxxxxxxxx',
  from: '+18005550101',
  to: '+18005550100 or sip:username@sip.telnyx.com',
});

console.log(response.data);

{
  "data": {
    "call_control_id": "v3:MdI91X4lWFEs7IgbBEOT9M4AigoY08M0WWZFISt1Yw2axZ_IiE4pqg",
    "call_leg_id": "2dc6fc34-f9e0-11ea-b68e-02420a0f7768",
    "call_session_id": "2dc1b3c8-f9e0-11ea-bc5a-02420a0f7768",
    "client_state": "aGF2ZSBhIG5pY2UgZGF5ID1d",
    "is_alive": false,
    "call_duration": 50,
    "record_type": "call",
    "recording_id": "d7e9c1d4-8b2a-4b8f-b3a7-9a671c9e9b0a",
    "start_time": "2019-01-23T18:10:02.574Z",
    "end_time": "2019-01-23T18:11:52.574Z"
  }
}

Authorizations

Authorization

string

header

required

Bearer authentication header of the form Bearer <token>, where <token> is your auth token.

Body

application/json

Call request

required

The DID or SIP URI to dial out to. Multiple DID or SIP URIs can be provided using an array of strings

Example:

"+18005550100 or sip:username@sip.telnyx.com"

from

string

required

The from number to be used as the caller id presented to the destination (to number). The number should be in +E164 format.

Example:

"+18005550101"

connection_id

string

required

The ID of the Call Control App (formerly ID of the connection) to be used when dialing the destination.

from_display_name

string

The from_display_name string to be used as the caller id name (SIP From Display Name) presented to the destination (to number). The string should have a maximum of 128 characters, containing only letters, numbers, spaces, and -_~!.+ special characters. If ommited, the display name will be the same as the number in the from field.

Example:

"Company Name"

audio_url

string

The URL of a file to be played back to the callee when the call is answered. The URL can point to either a WAV or MP3 file. media_name and audio_url cannot be used together in one request.

Example:

"http://example.com/message.wav"

media_name

string

The media_name of a file to be played back to the callee when the call is answered. The media_name must point to a file previously uploaded to api.telnyx.com/v2/media by the same user/organization. The file must either be a WAV or MP3 file.

Example:

"my_media_uploaded_to_media_storage_api"

preferred_codecs

string

The list of comma-separated codecs in a preferred order for the forked media to be received.

Example:

"G722,PCMU,PCMA,G729,OPUS,VP8,H264"

timeout_secs

integer<int32>

default:30

The number of seconds that Telnyx will wait for the call to be answered by the destination to which it is being called. If the timeout is reached before an answer is received, the call will hangup and a call.hangup webhook with a hangup_cause of timeout will be sent. Minimum value is 5 seconds. Maximum value is 600 seconds.

Example:

60

time_limit_secs

integer<int32>

default:14400

Sets the maximum duration of a Call Control Leg in seconds. If the time limit is reached, the call will hangup and a call.hangup webhook with a hangup_cause of time_limit will be sent. For example, by setting a time limit of 120 seconds, a Call Leg will be automatically terminated two minutes after being answered. The default time limit is 14400 seconds or 4 hours and this is also the maximum allowed call length.

Required range: 30 <= x <= 14400

Example:

600

answering_machine_detection

enum<string>

default:disabled

Enables Answering Machine Detection. Telnyx offers Premium and Standard detections. With Premium detection, when a call is answered, Telnyx runs real-time detection and sends a call.machine.premium.detection.ended webhook with one of the following results: human_residence, human_business, machine, silence or fax_detected. If we detect a beep, we also send a call.machine.premium.greeting.ended webhook with the result of beep_detected. If we detect a beep before call.machine.premium.detection.ended we only send call.machine.premium.greeting.ended, and if we detect a beep after call.machine.premium.detection.ended, we send both webhooks. With Standard detection, when a call is answered, Telnyx runs real-time detection to determine if it was picked up by a human or a machine and sends an call.machine.detection.ended webhook with the analysis result. If greeting_end or detect_words is used and a machine is detected, you will receive another call.machine.greeting.ended webhook when the answering machine greeting ends with a beep or silence. If detect_beep is used, you will only receive call.machine.greeting.ended if a beep is detected.

Available options:

premium,

detect,

detect_beep,

detect_words,

greeting_end,

disabled

answering_machine_detection_config

object

Optional configuration parameters to modify 'answering_machine_detection' performance.

Show child attributes

answering_machine_detection_config.total_analysis_time_millis

integer<int32>

default:3500

Maximum timeout threshold for overall detection.

Example:

5000

answering_machine_detection_config.after_greeting_silence_millis

integer<int32>

default:800

Silence duration threshold after a greeting message or voice for it be considered human.

Example:

1000

answering_machine_detection_config.between_words_silence_millis

integer<int32>

default:50

Maximum threshold for silence between words.

Example:

100

answering_machine_detection_config.greeting_duration_millis

integer<int32>

default:3500

Maximum threshold of a human greeting. If greeting longer than this value, considered machine.

Example:

1500

answering_machine_detection_config.initial_silence_millis

integer<int32>

default:3500

If initial silence duration is greater than this value, consider it a machine.

Example:

1800

answering_machine_detection_config.maximum_number_of_words

integer<int32>

default:5

If number of detected words is greater than this value, consder it a machine.

Example:

3

answering_machine_detection_config.maximum_word_length_millis

integer<int32>

default:3500

If a single word lasts longer than this threshold, consider it a machine.

Example:

2000

answering_machine_detection_config.silence_threshold

integer<int32>

default:256

Minimum noise threshold for any analysis.

Example:

512

answering_machine_detection_config.greeting_total_analysis_time_millis

integer<int32>

default:5000

If machine already detected, maximum timeout threshold to determine the end of the machine greeting.

Example:

7500

answering_machine_detection_config.greeting_silence_duration_millis

integer<int32>

default:1500

If machine already detected, maximum threshold for silence between words. If exceeded, the greeting is considered ended.

Example:

2000

conference_config

object

Optional configuration parameters to dial new participant into a conference.

Show child attributes

conference_config.id

string<uuid>

Conference ID to be joined

Example:

"0ccc7b54-4df3-4bca-a65a-3da1ecc777f0"

conference_config.conference_name

string

Conference name to be joined

Example:

"telnyx-conference"

conference_config.early_media

boolean

default:true

Controls the moment when dialled call is joined into conference. If set to true user will be joined as soon as media is available (ringback). If false user will be joined when call is answered. Defaults to true

Example:

false

conference_config.end_conference_on_exit

boolean

Whether the conference should end and all remaining participants be hung up after the participant leaves the conference. Defaults to "false".

Example:

true

conference_config.soft_end_conference_on_exit

boolean

Whether the conference should end after the participant leaves the conference. NOTE this doesn't hang up the other participants. Defaults to "false".

Example:

true

conference_config.hold

boolean

Whether the participant should be put on hold immediately after joining the conference. Defaults to "false".

Example:

true

conference_config.hold_audio_url

string

The URL of a file to be played to the participant when they are put on hold after joining the conference. hold_media_name and hold_audio_url cannot be used together in one request. Takes effect only when "start_conference_on_create" is set to "false". This property takes effect only if "hold" is set to "true".

Example:

"http://example.com/message.wav"

conference_config.hold_media_name

string

The media_name of a file to be played to the participant when they are put on hold after joining the conference. The media_name must point to a file previously uploaded to api.telnyx.com/v2/media by the same user/organization. The file must either be a WAV or MP3 file. Takes effect only when "start_conference_on_create" is set to "false". This property takes effect only if "hold" is set to "true".

Example:

"my_media_uploaded_to_media_storage_api"

conference_config.mute

boolean

Whether the participant should be muted immediately after joining the conference. Defaults to "false".

Example:

true

conference_config.start_conference_on_enter

boolean

Whether the conference should be started after the participant joins the conference. Defaults to "false".

Example:

true

conference_config.start_conference_on_create

boolean

Whether the conference should be started on creation. If the conference isn't started all participants that join are automatically put on hold. Defaults to "true".

Example:

false

conference_config.supervisor_role

enum<string>

Sets the joining participant as a supervisor for the conference. A conference can have multiple supervisors. "barge" means the supervisor enters the conference as a normal participant. This is the same as "none". "monitor" means the supervisor is muted but can hear all participants. "whisper" means that only the specified "whisper_call_control_ids" can hear the supervisor. Defaults to "none".

Available options:

barge,

monitor,

none,

whisper

Example:

"whisper"

conference_config.whisper_call_control_ids

string[]

Array of unique call_control_ids the joining supervisor can whisper to. If none provided, the supervisor will join the conference as a monitoring participant only.

Example:

[
  "v2:Sg1xxxQ_U3ixxxyXT_VDNI3xxxazZdg6Vxxxs4-GNYxxxVaJPOhFMRQ",
  "v2:qqpb0mmvd-ovhhBr0BUQQn0fld5jIboaaX3-De0DkqXHzbf8d75xkw"
]

conference_config.beep_enabled

enum<string>

Whether a beep sound should be played when the participant joins and/or leaves the conference. Can be used to override the conference-level setting.

Available options:

always,

never,

on_enter,

on_exit

Example:

"on_exit"

custom_headers

Custom SIP Header · object[]

Custom headers to be added to the SIP INVITE.

Show child attributes

custom_headers.name

string

required

The name of the header to add.

Example:

"head_1"

custom_headers.value

string

required

The value of the header.

Example:

"val_1"

Example:

[
  { "name": "head_1", "value": "val_1" },
  { "name": "head_2", "value": "val_2" }
]

billing_group_id

string<uuid>

Use this field to set the Billing Group ID for the call. Must be a valid and existing Billing Group ID.

Example:

"f5586561-8ff0-4291-a0ac-84fe544797bd"

client_state

string

Use this field to add state to every subsequent webhook. It must be a valid Base-64 encoded string.

Example:

"aGF2ZSBhIG5pY2UgZGF5ID1d"

command_id

string

Use this field to avoid duplicate commands. Telnyx will ignore others Dial commands with the same command_id.

Example:

"891510ac-f3e4-11e8-af5b-de00688a4901"

link_to

string

Use another call's control id for sharing the same call session id

Example:

"ilditnZK_eVysupV21KzmzN_sM29ygfauQojpm4BgFtfX5hXAcjotg=="

bridge_intent

boolean

default:false

Indicates the intent to bridge this call with the call specified in link_to. When bridge_intent is true, link_to becomes required and the from number will be overwritten by the from number from the linked call.

Example:

true

bridge_on_answer

boolean

default:false

Whether to automatically bridge answered call to the call specified in link_to. When bridge_on_answer is true, link_to becomes required.

Example:

true

park_after_unbridge

string

If supplied with the value self, the current leg will be parked after unbridge. If not set, the default behavior is to hang up the leg. When park_after_unbridge is set, link_to becomes required.

Example:

"self"

media_encryption

enum<string>

default:disabled

Defines whether media should be encrypted on the call.

Available options:

disabled,

SRTP,

DTLS

sip_auth_username

string

SIP Authentication username used for SIP challenges.

sip_auth_password

string

SIP Authentication password used for SIP challenges.

sip_headers

SIP Header · object[]

SIP headers to be added to the SIP INVITE request. Currently only User-to-User header is supported.

Show child attributes

sip_headers.name

enum<string>

required

The name of the header to add.

Available options:

User-to-User

Example:

"User-to-User"

sip_headers.value

string

required

The value of the header.

Example:

"value"

Example:

[
  { "name": "User-to-User", "value": "value" }
]

sip_transport_protocol

enum<string>

default:UDP

Defines SIP transport protocol to be used on the call.

Available options:

UDP,

TCP,

TLS

sound_modifications

Sound modifications · object

Use this field to modify sound effects, for example adjust the pitch.

Show child attributes

sound_modifications.pitch

number<double>

Set the pitch directly, value should be > 0, default 1 (lower = lower tone)

Example:

0.8

sound_modifications.semitone

number<double>

Adjust the pitch in semitones, values should be between -14 and 14, default 0

Example:

-10

sound_modifications.octaves

number<double>

Adjust the pitch in octaves, values should be between -1 and 1, default 0

Example:

-0.5

sound_modifications.track

string

default:outbound

The track to which the sound modifications will be applied. Accepted values are inbound or outbound

Example:

"inbound"

Example:

{
  "pitch": "0.8",
  "semitone": -2,
  "octaves": 0.1,
  "track": "both"
}

stream_url

string

The destination WebSocket address where the stream is going to be delivered.

Example:

"wss://www.example.com/websocket"

stream_track

enum<string>

default:inbound_track

Specifies which track should be streamed.

Available options:

inbound_track,

outbound_track,

both_tracks

Example:

"both_tracks"

stream_codec

enum<string>

default:default

Specifies the codec to be used for the streamed audio. When set to 'default' or when transcoding is not possible, the codec from the call will be used.

Available options:

PCMU,

PCMA,

G722,

OPUS,

AMR-WB,

L16,

default

Example:

"PCMA"

stream_bidirectional_mode

enum<string>

default:mp3

Configures method of bidirectional streaming (mp3, rtp).

Available options:

mp3,

rtp

Example:

"rtp"

stream_bidirectional_codec

enum<string>

default:PCMU

Indicates codec for bidirectional streaming RTP payloads. Used only with stream_bidirectional_mode=rtp. Case sensitive.

Available options:

PCMU,

PCMA,

G722,

OPUS,

AMR-WB,

L16

Example:

"G722"

stream_bidirectional_target_legs

enum<string>

default:opposite

Specifies which call legs should receive the bidirectional stream audio.

Available options:

both,

self,

opposite

Example:

"both"

stream_bidirectional_sampling_rate

enum<integer>

default:8000

Audio sampling rate.

Available options:

8000,

16000,

22050,

24000,

48000

Example:

16000

stream_establish_before_call_originate

boolean

default:false

Establish websocket connection before dialing the destination. This is useful for cases where the websocket connection takes a long time to establish.

Example:

true

send_silence_when_idle

boolean

default:false

Generate silence RTP packets when no transmission available.

Example:

true

webhook_url

string

Use this field to override the URL for which Telnyx will send subsequent webhooks to for this call.

Example:

"https://www.example.com/server-b/"

webhook_url_method

enum<string>

default:POST

HTTP request type used for webhook_url.

Available options:

POST,

GET

Example:

"GET"

record

enum<string>

Start recording automatically after an event. Disabled by default.

Available options:

record-from-answer

Example:

"record-from-answer"

record_channels

enum<string>

default:dual

Defines which channel should be recorded ('single' or 'dual') when record is specified.

Available options:

single,

dual

Example:

"single"

record_format

enum<string>

default:mp3

Defines the format of the recording ('wav' or 'mp3') when record is specified.

Available options:

wav,

mp3

Example:

"wav"

record_max_length

integer<int32>

default:0

Defines the maximum length for the recording in seconds when record is specified. The minimum value is 0. The maximum value is 43200. The default value is 0 (infinite).

Example:

1000

record_timeout_secs

integer<int32>

default:0

The number of seconds that Telnyx will wait for the recording to be stopped if silence is detected when record is specified. The timer only starts when the speech is detected. Please note that call transcription is used to detect silence and the related charge will be applied. The minimum value is 0. The default value is 0 (infinite).

Example:

100

record_track

enum<string>

default:both

The audio track to be recorded. Can be either both, inbound or outbound. If only single track is specified (inbound, outbound), channels configuration is ignored and it will be recorded as mono (single channel).

Available options:

both,

inbound,

outbound

Example:

"outbound"

record_trim

enum<string>

When set to trim-silence, silence will be removed from the beginning and end of the recording.

Available options:

trim-silence

Example:

"trim-silence"

record_custom_file_name

string

The custom recording file name to be used instead of the default call_leg_id. Telnyx will still add a Unix timestamp suffix.

Required string length: 1 - 40

Example:

"my_recording_file_name"

supervise_call_control_id

string

The call leg which will be supervised by the new call.

Example:

"v3:MdI91X4lWFEs7IgbBEOT9M4AigoY08M0WWZFISt1Yw2axZ_IiE4pqg"

supervisor_role

enum<string>

default:barge

The role of the supervisor call. 'barge' means that supervisor call hears and is being heard by both ends of the call (caller & callee). 'whisper' means that only supervised_call_control_id hears supervisor but supervisor can hear everything. 'monitor' means that nobody can hear supervisor call, but supervisor can hear everything on the call.

Available options:

barge,

whisper,

monitor

enable_dialogflow

boolean

default:false

Enables Dialogflow for the current call. The default value is false.

Example:

true

dialogflow_config

Dialogflow Config · object

Show child attributes

dialogflow_config.analyze_sentiment

boolean

default:false

Enable sentiment analysis from Dialogflow.

Example:

true

dialogflow_config.partial_automated_agent_reply

boolean

default:false

Enable partial automated agent reply from Dialogflow.

Example:

true

transcription

boolean

default:false

Enable transcription upon call answer. The default value is false.

Example:

true

transcription_config

Transcription start request · object

Show child attributes

transcription_config.transcription_engine

enum<string>

default:Google

Engine to use for speech recognition. Legacy values A - Google, B - Telnyx are supported for backward compatibility.

Available options:

Google,

Telnyx,

Deepgram,

A,

B

Example:

"Google"

transcription_config.transcription_engine_config

Transcription engine Google config · object

Transcription engine Google config
Transcription engine Telnyx config
Transcription engine Deepgram config
Transcription engine A config
Transcription engine B config

Show child attributes

transcription_config.transcription_engine_config.transcription_engine

enum<string>

Engine identifier for Google transcription service

Available options:

Google

transcription_config.transcription_engine_config.language

enum<string>

default:en

Language to use for speech recognition

Available options:

af,

sq,

am,

ar,

hy,

az,

eu,

bn,

bs,

bg,

my,

ca,

yue,

zh,

hr,

cs,

da,

nl,

en,

et,

fil,

fi,

fr,

gl,

ka,

de,

el,

gu,

iw,

hi,

hu,

is,

id,

it,

ja,

jv,

kn,

kk,

km,

ko,

lo,

lv,

lt,

mk,

ms,

ml,

mr,

mn,

ne,

no,

fa,

pl,

pt,

pa,

ro,

ru,

rw,

sr,

si,

sk,

sl,

ss,

st,

es,

su,

sw,

sv,

ta,

te,

th,

tn,

tr,

ts,

uk,

ur,

uz,

ve,

vi,

xh,

zu

Example:

"en"

transcription_config.transcription_engine_config.interim_results

boolean

default:false

Whether to send also interim results. If set to false, only final results will be sent.

Example:

true

transcription_config.transcription_engine_config.enable_speaker_diarization

boolean

default:false

Enables speaker diarization.

Example:

true

transcription_config.transcription_engine_config.min_speaker_count

integer<int32>

default:2

Defines minimum number of speakers in the conversation.

Example:

4

transcription_config.transcription_engine_config.max_speaker_count

integer<int32>

default:6

Defines maximum number of speakers in the conversation.

Example:

4

transcription_config.transcription_engine_config.profanity_filter

boolean

default:false

Enables profanity_filter.

Example:

true

transcription_config.transcription_engine_config.use_enhanced

boolean

default:false

Enables enhanced transcription, this works for models phone_call and video.

Example:

true

transcription_config.transcription_engine_config.model

enum<string>

The model to use for transcription.

Available options:

latest_long,

latest_short,

command_and_search,

phone_call,

video,

default,

medical_conversation,

medical_dictation

transcription_config.transcription_engine_config.hints

string[]

Hints to improve transcription accuracy.

Example:

[]

transcription_config.transcription_engine_config.speech_context

object[]

Speech context to improve transcription accuracy.

Show child attributes

transcription_config.transcription_engine_config.speech_context.phrases

string[]

Example:

[]

transcription_config.transcription_engine_config.speech_context.boost

number

default:1

Boost factor for the speech context.

Required range: 0 <= x <= 20

Example:

1

transcription_config.client_state

string

Use this field to add state to every subsequent webhook. It must be a valid Base-64 encoded string.

Example:

"aGF2ZSBhIG5pY2UgZGF5ID1d"

transcription_config.transcription_tracks

string

default:inbound

Indicates which leg of the call will be transcribed. Use inbound for the leg that requested the transcription, outbound for the other leg, and both for both legs of the call. Will default to inbound.

Example:

"both"

transcription_config.command_id

string

Use this field to avoid duplicate commands. Telnyx will ignore any command with the same command_id for the same call_control_id.

Example:

"891510ac-f3e4-11e8-af5b-de00688a4901"

Example:

{
  "language": "en",
  "client_state": "aGF2ZSBhIG5pY2UgZGF5ID1d",
  "command_id": "891510ac-f3e4-11e8-af5b-de00688a4901"
}

sip_region

enum<string>

default:US

Defines the SIP region to be used for the call.

Available options:

US,

Europe,

Canada,

Australia,

Middle East

Example:

"Canada"

Response

Successful response with details about a call status that includes recording_id.

data

Call With Recording ID · object

Show child attributes

data.record_type

enum<string>

required

Available options:

call

Example:

"call"

data.call_session_id

string

required

ID that is unique to the call session and can be used to correlate webhook events. Call session is a group of related call legs that logically belong to the same phone call, e.g. an inbound and outbound leg of a transferred call

Example:

"428c31b6-7af4-4bcb-b68e-5013ef9657c1"

data.call_leg_id

string

required

ID that is unique to the call and can be used to correlate webhook events

Example:

"428c31b6-7af4-4bcb-b7f5-5013ef9657c1"

data.call_control_id

string

required

Unique identifier and token for controlling the call.

Example:

"v3:MdI91X4lWFEs7IgbBEOT9M4AigoY08M0WWZFISt1Yw2axZ_IiE4pqg"

data.is_alive

boolean

required

Indicates whether the call is alive or not. For Dial command it will always be false (dialing is asynchronous).

Example:

true

data.client_state

string

State received from a command.

Example:

"aGF2ZSBhIG5pY2UgZGF5ID1d"

data.call_duration

integer

Indicates the duration of the call in seconds

Example:

50

data.recording_id

string<uuid>

The ID of the recording. Only present when the record parameter is set to record-from-answer.

Example:

"d7e9c1d4-8b2a-4b8f-b3a7-9a671c9e9b0a"

data.start_time

string

ISO 8601 formatted date indicating when the call started

Example:

"2019-01-23T18:10:02.574Z"

data.end_time

string

ISO 8601 formatted date indicating when the call ended. Only present when the call is not alive

Example:

"2019-01-23T18:11:52.574Z"

Example:

{
  "call_control_id": "v3:MdI91X4lWFEs7IgbBEOT9M4AigoY08M0WWZFISt1Yw2axZ_IiE4pqg",
  "call_leg_id": "2dc6fc34-f9e0-11ea-b68e-02420a0f7768",
  "call_session_id": "2dc1b3c8-f9e0-11ea-bc5a-02420a0f7768",
  "client_state": "aGF2ZSBhIG5pY2UgZGF5ID1d",
  "is_alive": false,
  "call_duration": 50,
  "record_type": "call",
  "recording_id": "d7e9c1d4-8b2a-4b8f-b3a7-9a671c9e9b0a",
  "start_time": "2019-01-23T18:10:02.574Z",
  "end_time": "2019-01-23T18:11:52.574Z"
}

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