Dial a new conference participant
POST/texml/Accounts/:account_sid/Conferences/:conference_sid/Participants
Dials a new conference participant
Request
Path Parameters
The id of the account the resource belongs to.
The ConferenceSid that uniquely identifies a conference.
- application/x-www-form-urlencoded
Body
required
Dial Conference Participant request object
Possible values: [true, false, onEnter, onExit]
Whether to play a notification beep to the conference when the participant enters and exits.
URL destination for Telnyx to send status callback events to for the call.
Possible values: [GET, POST]
HTTP request type used for StatusCallback.
The changes to the call's state that should generate a call to StatusCallback. Can be: initiated, ringing, answered, and completed. Separate multiple values with a space. The default value is completed.
The phone number of the called party. Phone numbers are formatted with a + and country code.
The phone number of the party that initiated the call. Phone numbers are formatted with a + and country code.
The number of seconds that we should allow the phone to ring before assuming there is no answer. Can be an integer between 5 and 120, inclusive. The default value is 30.
Whether the participant should be muted.
Whether to start the conference when the participant enters. Defaults to true.
Whether to end the conference when the participant leaves. Defaults to false.
Whether participant shall be bridged to conference before the participant answers (from early media if available). Defaults to false.
The URL the conference callbacks will be sent to.
Possible values: [GET, POST]
HTTP request type used for ConferenceStatusCallback. Defaults to POST.
The changes to the conference's state that should generate a call to ConferenceStatusCallback. Can be: start, end, join and leave. Separate multiple values with a space. By default no callbacks are sent.
The URL to call for an audio file to play while the participant is waiting for the conference to start.
The maximum number of participants in the conference. Can be a positive integer from 2 to 800. The default value is 250.
Whether the participant is coaching another call. When true, CallSidToCoach has to be given.
The SID of the participant who is being coached. The participant being coached is the only participant who can hear the participant who is coaching.
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.
The maximum duration of the call in seconds.
Possible values: [Enable, DetectMessageEnd]
Whether to detect if a human or an answering machine picked up the call. Use Enable if you would like to ne notified as soon as the called party is identified. Use DetectMessageEnd, if you would like to leave a message on an answering machine.
How long answering machine detection should go on for before sending an Unknown result. Given in milliseconds.
Default value: 3500
Maximum threshold of a human greeting. If greeting longer than this value, considered machine. Ignored when premium detection is used.
Default value: 800
Silence duration threshold after a greeting message or voice for it be considered human. Ignored when premium detection is used.
Default value: 3500
If initial silence duration is greater than this value, consider it a machine. Ignored when premium detection is used.
The URL the result of answering machine detection will be sent to.
Possible values: [GET, POST]
HTTP request type used for AmdStatusCallback. Defaults to POST.
Default value: true
Whether to cancel ongoing playback on machine detection. Defaults to true.
Default value: true
Whether to cancel ongoing playback on greeting ended detection. Defaults to true.
The list of comma-separated codecs to be offered on a call.
Whether to record the entire participant's call leg. Defaults to false.
Possible values: [mono, dual]
The number of channels in the final recording. Defaults to mono.
The URL the recording callbacks will be sent to.
Possible values: [GET, POST]
HTTP request type used for RecordingStatusCallback. Defaults to POST.
The changes to the recording's state that should generate a call to RecoridngStatusCallback. Can be: in-progress, completed and absent. Separate multiple values with a space. Defaults to completed.
Possible values: [inbound, outbound, both]
The audio track to record for the call. The default is both.
The password to use for SIP authentication.
The username to use for SIP authentication.
Possible values: [trim-silence, do-not-trim]
Whether to trim any leading and trailing silence from the recording. Defaults to trim-silence.
Possible values: [true, false, record-from-start, do-not-record]
Whether to record the conference the participant is joining. Defualts to do-not-record. The boolean values true and false are synonymous with record-from-start and do-not-record respectively.
The URL the conference recording callbacks will be sent to.
Possible values: [GET, POST]
HTTP request type used for ConferenceRecordingStatusCallback. Defaults to POST.
The changes to the conference recording's state that should generate a call to RecoridngStatusCallback. Can be: in-progress, completed and absent. Separate multiple values with a space. Defaults to completed. failed and absent are synonymous.
Default value: 0
The number of seconds that Telnyx will wait for the recording to be stopped if silence is detected. The timer only starts when the speech is detected. Please note that the transcription is used to detect silence and the related charge will be applied. The minimum value is 0. The default value is 0 (infinite)
Possible values: [trim-silence, do-not-trim]
Whether to trim any leading and trailing silence from the conference recording. Defaults to trim-silence.
Responses
200: New participant resource.
- application/json
404: Resource not found
- application/json