pub struct CreateParticipantParams {Show 50 fields
pub account_sid: String,
pub conference_sid: String,
pub from: String,
pub to: String,
pub status_callback: Option<String>,
pub status_callback_method: Option<String>,
pub status_callback_event: Option<Vec<String>>,
pub label: Option<String>,
pub timeout: Option<i32>,
pub record: Option<bool>,
pub muted: Option<bool>,
pub beep: Option<String>,
pub start_conference_on_enter: Option<bool>,
pub end_conference_on_exit: Option<bool>,
pub wait_url: Option<String>,
pub wait_method: Option<String>,
pub early_media: Option<bool>,
pub max_participants: Option<i32>,
pub conference_record: Option<String>,
pub conference_trim: Option<String>,
pub conference_status_callback: Option<String>,
pub conference_status_callback_method: Option<String>,
pub conference_status_callback_event: Option<Vec<String>>,
pub recording_channels: Option<String>,
pub recording_status_callback: Option<String>,
pub recording_status_callback_method: Option<String>,
pub sip_auth_username: Option<String>,
pub sip_auth_password: Option<String>,
pub region: Option<String>,
pub conference_recording_status_callback: Option<String>,
pub conference_recording_status_callback_method: Option<String>,
pub recording_status_callback_event: Option<Vec<String>>,
pub conference_recording_status_callback_event: Option<Vec<String>>,
pub coaching: Option<bool>,
pub call_sid_to_coach: Option<String>,
pub jitter_buffer_size: Option<String>,
pub byoc: Option<String>,
pub caller_id: Option<String>,
pub call_reason: Option<String>,
pub recording_track: Option<String>,
pub time_limit: Option<i32>,
pub machine_detection: Option<String>,
pub machine_detection_timeout: Option<i32>,
pub machine_detection_speech_threshold: Option<i32>,
pub machine_detection_speech_end_threshold: Option<i32>,
pub machine_detection_silence_timeout: Option<i32>,
pub amd_status_callback: Option<String>,
pub amd_status_callback_method: Option<String>,
pub trim: Option<String>,
pub call_token: Option<String>,
}Expand description
struct for passing parameters to the method create_participant
Fields§
§account_sid: StringThe SID of the Account that will create the resource.
conference_sid: StringThe SID of the participant’s conference.
from: StringThe phone number, Client identifier, or username portion of SIP address that made this call. Phone numbers are in E.164 format (e.g., +16175551212). Client identifiers are formatted client:name. If using a phone number, it must be a Twilio number or a Verified outgoing caller id for your account. If the to parameter is a phone number, from must also be a phone number. If to is sip address, this value of from should be a username portion to be used to populate the P-Asserted-Identity header that is passed to the SIP endpoint.
to: StringThe phone number, SIP address, or Client identifier that received this call. Phone numbers are in E.164 format (e.g., +16175551212). SIP addresses are formatted as sip:name@company.com. Client identifiers are formatted client:name. Custom parameters may also be specified.
status_callback: Option<String>The URL we should call using the status_callback_method to send status information to your application.
status_callback_method: Option<String>The HTTP method we should use to call status_callback. Can be: GET and POST and defaults to POST.
status_callback_event: Option<Vec<String>>The conference state changes that should generate a call to status_callback. Can be: initiated, ringing, answered, and completed. Separate multiple values with a space. The default value is completed.
label: Option<String>A label for this participant. If one is supplied, it may subsequently be used to fetch, update or delete the participant.
timeout: Option<i32>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 600, inclusive. The default value is 60. We always add a 5-second timeout buffer to outgoing calls, so value of 10 would result in an actual timeout that was closer to 15 seconds.
record: Option<bool>Whether to record the participant and their conferences, including the time between conferences. Can be true or false and the default is false.
muted: Option<bool>Whether the agent is muted in the conference. Can be true or false and the default is false.
beep: Option<String>Whether to play a notification beep to the conference when the participant joins. Can be: true, false, onEnter, or onExit. The default value is true.
start_conference_on_enter: Option<bool>Whether to start the conference when the participant joins, if it has not already started. Can be: true or false and the default is true. If false and the conference has not started, the participant is muted and hears background music until another participant starts the conference.
end_conference_on_exit: Option<bool>Whether to end the conference when the participant leaves. Can be: true or false and defaults to false.
wait_url: Option<String>The URL we should call using the wait_method for the music to play while participants are waiting for the conference to start. The default value is the URL of our standard hold music. Learn more about hold music.
wait_method: Option<String>The HTTP method we should use to call wait_url. Can be GET or POST and the default is POST. When using a static audio file, this should be GET so that we can cache the file.
early_media: Option<bool>Whether to allow an agent to hear the state of the outbound call, including ringing or disconnect messages. Can be: true or false and defaults to true.
max_participants: Option<i32>The maximum number of participants in the conference. Can be a positive integer from 2 to 250. The default value is 250.
conference_record: Option<String>Whether to record the conference the participant is joining. Can be: true, false, record-from-start, and do-not-record. The default value is false.
conference_trim: Option<String>Whether to trim leading and trailing silence from the conference recording. Can be: trim-silence or do-not-trim and defaults to trim-silence.
conference_status_callback: Option<String>The URL we should call using the conference_status_callback_method when the conference events in conference_status_callback_event occur. Only the value set by the first participant to join the conference is used. Subsequent conference_status_callback values are ignored.
conference_status_callback_method: Option<String>The HTTP method we should use to call conference_status_callback. Can be: GET or POST and defaults to POST.
conference_status_callback_event: Option<Vec<String>>The conference state changes that should generate a call to conference_status_callback. Can be: start, end, join, leave, mute, hold, modify, speaker, and announcement. Separate multiple values with a space. Defaults to start end.
recording_channels: Option<String>The recording channels for the final recording. Can be: mono or dual and the default is mono.
recording_status_callback: Option<String>The URL that we should call using the recording_status_callback_method when the recording status changes.
recording_status_callback_method: Option<String>The HTTP method we should use when we call recording_status_callback. Can be: GET or POST and defaults to POST.
sip_auth_username: Option<String>The SIP username used for authentication.
sip_auth_password: Option<String>The SIP password for authentication.
region: Option<String>The region where we should mix the recorded audio. Can be:us1, ie1, de1, sg1, br1, au1, or jp1.
conference_recording_status_callback: Option<String>The URL we should call using the conference_recording_status_callback_method when the conference recording is available.
conference_recording_status_callback_method: Option<String>The HTTP method we should use to call conference_recording_status_callback. Can be: GET or POST and defaults to POST.
recording_status_callback_event: Option<Vec<String>>The recording state changes that should generate a call to recording_status_callback. Can be: started, in-progress, paused, resumed, stopped, completed, failed, and absent. Separate multiple values with a space, ex: 'in-progress completed failed'.
conference_recording_status_callback_event: Option<Vec<String>>The conference recording state changes that generate a call to conference_recording_status_callback. Can be: in-progress, completed, failed, and absent. Separate multiple values with a space, ex: 'in-progress completed failed'
coaching: Option<bool>Whether the participant is coaching another call. Can be: true or false. If not present, defaults to false unless call_sid_to_coach is defined. If true, call_sid_to_coach must be defined.
call_sid_to_coach: Option<String>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.
jitter_buffer_size: Option<String>Jitter buffer size for the connecting participant. Twilio will use this setting to apply Jitter Buffer before participant’s audio is mixed into the conference. Can be: off, small, medium, and large. Default to large.
byoc: Option<String>The SID of a BYOC (Bring Your Own Carrier) trunk to route this call with. Note that byoc is only meaningful when to is a phone number; it will otherwise be ignored. (Beta)
caller_id: Option<String>The phone number, Client identifier, or username portion of SIP address that made this call. Phone numbers are in E.164 format (e.g., +16175551212). Client identifiers are formatted client:name. If using a phone number, it must be a Twilio number or a Verified outgoing caller id for your account. If the to parameter is a phone number, callerId must also be a phone number. If to is sip address, this value of callerId should be a username portion to be used to populate the From header that is passed to the SIP endpoint.
call_reason: Option<String>The Reason for the outgoing call. Use it to specify the purpose of the call that is presented on the called party’s phone. (Branded Calls Beta)
recording_track: Option<String>The audio track to record for the call. Can be: inbound, outbound or both. The default is both. inbound records the audio that is received by Twilio. outbound records the audio that is sent from Twilio. both records the audio that is received and sent by Twilio.
time_limit: Option<i32>The maximum duration of the call in seconds. Constraints depend on account and configuration.
machine_detection: Option<String>Whether to detect if a human, answering machine, or fax has picked up the call. Can be: Enable or DetectMessageEnd. Use Enable if you would like us to return AnsweredBy as soon as the called party is identified. Use DetectMessageEnd, if you would like to leave a message on an answering machine. For more information, see Answering Machine Detection.
machine_detection_timeout: Option<i32>The number of seconds that we should attempt to detect an answering machine before timing out and sending a voice request with AnsweredBy of unknown. The default timeout is 30 seconds.
machine_detection_speech_threshold: Option<i32>The number of milliseconds that is used as the measuring stick for the length of the speech activity, where durations lower than this value will be interpreted as a human and longer than this value as a machine. Possible Values: 1000-6000. Default: 2400.
machine_detection_speech_end_threshold: Option<i32>The number of milliseconds of silence after speech activity at which point the speech activity is considered complete. Possible Values: 500-5000. Default: 1200.
machine_detection_silence_timeout: Option<i32>The number of milliseconds of initial silence after which an unknown AnsweredBy result will be returned. Possible Values: 2000-10000. Default: 5000.
amd_status_callback: Option<String>The URL that we should call using the amd_status_callback_method to notify customer application whether the call was answered by human, machine or fax.
amd_status_callback_method: Option<String>The HTTP method we should use when calling the amd_status_callback URL. Can be: GET or POST and the default is POST.
trim: Option<String>Whether to trim any leading and trailing silence from the participant recording. Can be: trim-silence or do-not-trim and the default is trim-silence.
call_token: Option<String>A token string needed to invoke a forwarded call. A call_token is generated when an incoming call is received on a Twilio number. Pass an incoming call’s call_token value to a forwarded call via the call_token parameter when creating a new call. A forwarded call should bear the same CallerID of the original incoming call.
Trait Implementations§
Source§impl Clone for CreateParticipantParams
impl Clone for CreateParticipantParams
Source§fn clone(&self) -> CreateParticipantParams
fn clone(&self) -> CreateParticipantParams
1.0.0 · Source§fn clone_from(&mut self, source: &Self)
fn clone_from(&mut self, source: &Self)
source. Read more