Create participant
Conference participants
Path parameters
-
The SID of the Account that will create the resource.
-
The SID of the participant's conference.
Body
-
Beep string
Whether to play a notification beep to the conference when the participant joins. Can be:
true
,false
,onEnter
, oronExit
. The default value istrue
. -
Byoc string
The SID of a BYOC (Bring Your Own Carrier) trunk to route this call with. Note that
byoc
is only meaningful whento
is a phone number; it will otherwise be ignored. (Beta)Minimum length is
34
, maximum length is34
. Format should match the following pattern:^BY[0-9a-fA-F]{32}$
. -
CallReason 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)
-
CallSidToCoach string
The SID of the participant who is being
coached
. The participant being coached is the only participant who can hear the participant who iscoaching
.Minimum length is
34
, maximum length is34
. Format should match the following pattern:^CA[0-9a-fA-F]{32}$
. -
CallerId 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 theto
parameter is a phone number,callerId
must also be a phone number. Ifto
is sip address, this value ofcallerId
should be a username portion to be used to populate the From header that is passed to the SIP endpoint. -
Coaching boolean
Whether the participant is coaching another call. Can be:
true
orfalse
. If not present, defaults tofalse
unlesscall_sid_to_coach
is defined. Iftrue
,call_sid_to_coach
must be defined. -
ConferenceRecord string
Whether to record the conference the participant is joining. Can be:
true
,false
,record-from-start
, anddo-not-record
. The default value isfalse
. -
ConferenceRecordingStatusCallback string(uri)
The URL we should call using the
conference_recording_status_callback_method
when the conference recording is available. -
ConferenceRecordingStatusCallbackEvent array[string]
The conference recording state changes that generate a call to
conference_recording_status_callback
. Can be:in-progress
,completed
,failed
, andabsent
. Separate multiple values with a space, ex:'in-progress completed failed'
-
ConferenceRecordingStatusCallbackMethod string(http-method)
The HTTP method we should use to call
conference_recording_status_callback
. Can be:GET
orPOST
and defaults toPOST
.Values are
HEAD
,GET
,POST
,PATCH
,PUT
, orDELETE
. -
ConferenceStatusCallback string(uri)
The URL we should call using the
conference_status_callback_method
when the conference events inconference_status_callback_event
occur. Only the value set by the first participant to join the conference is used. Subsequentconference_status_callback
values are ignored. -
ConferenceStatusCallbackEvent array[string]
The conference state changes that should generate a call to
conference_status_callback
. Can be:start
,end
,join
,leave
,mute
,hold
,modify
,speaker
, andannouncement
. Separate multiple values with a space. Defaults tostart end
. -
ConferenceStatusCallbackMethod string(http-method)
The HTTP method we should use to call
conference_status_callback
. Can be:GET
orPOST
and defaults toPOST
.Values are
HEAD
,GET
,POST
,PATCH
,PUT
, orDELETE
. -
ConferenceTrim string
Whether to trim leading and trailing silence from your recorded conference audio files. Can be:
trim-silence
ordo-not-trim
and defaults totrim-silence
. -
EarlyMedia boolean
Whether to allow an agent to hear the state of the outbound call, including ringing or disconnect messages. Can be:
true
orfalse
and defaults totrue
. -
EndConferenceOnExit boolean
Whether to end the conference when the participant leaves. Can be:
true
orfalse
and defaults tofalse
. -
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 theto
parameter is a phone number,from
must also be a phone number. Ifto
is sip address, this value offrom
should be a username portion to be used to populate the P-Asserted-Identity header that is passed to the SIP endpoint. -
JitterBufferSize 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
, andlarge
. Default tolarge
. -
Label string
A label for this participant. If one is supplied, it may subsequently be used to fetch, update or delete the participant.
-
MaxParticipants integer
The maximum number of participants in the conference. Can be a positive integer from
2
to250
. The default value is250
. -
Muted boolean
Whether the agent is muted in the conference. Can be
true
orfalse
and the default isfalse
. -
Record boolean
Whether to record the participant and their conferences, including the time between conferences. Can be
true
orfalse
and the default isfalse
. -
RecordingChannels string
The recording channels for the final recording. Can be:
mono
ordual
and the default ismono
. -
RecordingStatusCallback string(uri)
The URL that we should call using the
recording_status_callback_method
when the recording status changes. -
RecordingStatusCallbackEvent array[string]
The recording state changes that should generate a call to
recording_status_callback
. Can be:started
,in-progress
,paused
,resumed
,stopped
,completed
,failed
, andabsent
. Separate multiple values with a space, ex:'in-progress completed failed'
. -
RecordingStatusCallbackMethod string(http-method)
The HTTP method we should use when we call
recording_status_callback
. Can be:GET
orPOST
and defaults toPOST
.Values are
HEAD
,GET
,POST
,PATCH
,PUT
, orDELETE
. -
RecordingTrack string
The audio track to record for the call. Can be:
inbound
,outbound
orboth
. The default isboth
.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. -
Region string
The region where we should mix the recorded audio. Can be:
us1
,ie1
,de1
,sg1
,br1
,au1
, orjp1
. -
SipAuthPassword string
The SIP password for authentication.
-
SipAuthUsername string
The SIP username used for authentication.
-
StartConferenceOnEnter boolean
Whether to start the conference when the participant joins, if it has not already started. Can be:
true
orfalse
and the default istrue
. Iffalse
and the conference has not started, the participant is muted and hears background music until another participant starts the conference. -
StatusCallback string(uri)
The URL we should call using the
status_callback_method
to send status information to your application. -
StatusCallbackEvent array[string]
The conference state changes that should generate a call to
status_callback
. Can be:initiated
,ringing
,answered
, andcompleted
. Separate multiple values with a space. The default value iscompleted
. -
StatusCallbackMethod string(http-method)
The HTTP method we should use to call
status_callback
. Can be:GET
andPOST
and defaults toPOST
.Values are
HEAD
,GET
,POST
,PATCH
,PUT
, orDELETE
. -
TimeLimit integer
The maximum duration of the call in seconds. Constraints depend on account and configuration.
-
Timeout integer
The number of seconds that we should allow the phone to ring before assuming there is no answer. Can be an integer between
5
and600
, inclusive. The default value is60
. 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. -
The 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 formattedclient:name
. Custom parameters may also be specified. -
WaitMethod string(http-method)
The HTTP method we should use to call
wait_url
. Can beGET
orPOST
and the default isPOST
. When using a static audio file, this should beGET
so that we can cache the file.Values are
HEAD
,GET
,POST
,PATCH
,PUT
, orDELETE
. -
WaitUrl string(uri)
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.
curl \
-X POST https://api.twilio.com/2010-04-01/Accounts/{AccountSid}/Conferences/{ConferenceSid}/Participants.json \
--user "username:password" \
-H "Content-Type: application/x-www-form-urlencoded" \
-d 'Beep=string&Byoc=string&CallReason=string&CallSidToCoach=string&CallerId=string&Coaching=true&ConferenceRecord=string&ConferenceRecordingStatusCallback=https%3A%2F%2Fexample.com&ConferenceRecordingStatusCallbackEvent=string&ConferenceRecordingStatusCallbackMethod=HEAD&ConferenceStatusCallback=https%3A%2F%2Fexample.com&ConferenceStatusCallbackEvent=string&ConferenceStatusCallbackMethod=HEAD&ConferenceTrim=string&EarlyMedia=true&EndConferenceOnExit=true&From=string&JitterBufferSize=string&Label=string&MaxParticipants=42&Muted=true&Record=true&RecordingChannels=string&RecordingStatusCallback=https%3A%2F%2Fexample.com&RecordingStatusCallbackEvent=string&RecordingStatusCallbackMethod=HEAD&RecordingTrack=string&Region=string&SipAuthPassword=string&SipAuthUsername=string&StartConferenceOnEnter=true&StatusCallback=https%3A%2F%2Fexample.com&StatusCallbackEvent=string&StatusCallbackMethod=HEAD&TimeLimit=42&Timeout=42&To=string&WaitMethod=HEAD&WaitUrl=https%3A%2F%2Fexample.com'
{
"Beep": "string",
"Byoc": "string",
"CallReason": "string",
"CallSidToCoach": "string",
"CallerId": "string",
"Coaching": true,
"ConferenceRecord": "string",
"ConferenceRecordingStatusCallback": "https://example.com",
"ConferenceRecordingStatusCallbackEvent": [
"string"
],
"ConferenceRecordingStatusCallbackMethod": "HEAD",
"ConferenceStatusCallback": "https://example.com",
"ConferenceStatusCallbackEvent": [
"string"
],
"ConferenceStatusCallbackMethod": "HEAD",
"ConferenceTrim": "string",
"EarlyMedia": true,
"EndConferenceOnExit": true,
"From": "string",
"JitterBufferSize": "string",
"Label": "string",
"MaxParticipants": 42,
"Muted": true,
"Record": true,
"RecordingChannels": "string",
"RecordingStatusCallback": "https://example.com",
"RecordingStatusCallbackEvent": [
"string"
],
"RecordingStatusCallbackMethod": "HEAD",
"RecordingTrack": "string",
"Region": "string",
"SipAuthPassword": "string",
"SipAuthUsername": "string",
"StartConferenceOnEnter": true,
"StatusCallback": "https://example.com",
"StatusCallbackEvent": [
"string"
],
"StatusCallbackMethod": "HEAD",
"TimeLimit": 42,
"Timeout": 42,
"To": "string",
"WaitMethod": "HEAD",
"WaitUrl": "https://example.com"
}
{
"Beep": "string",
"Byoc": "string",
"CallReason": "string",
"CallSidToCoach": "string",
"CallerId": "string",
"Coaching": true,
"ConferenceRecord": "string",
"ConferenceRecordingStatusCallback": "https://example.com",
"ConferenceRecordingStatusCallbackEvent": [
"string"
],
"ConferenceRecordingStatusCallbackMethod": "HEAD",
"ConferenceStatusCallback": "https://example.com",
"ConferenceStatusCallbackEvent": [
"string"
],
"ConferenceStatusCallbackMethod": "HEAD",
"ConferenceTrim": "string",
"EarlyMedia": true,
"EndConferenceOnExit": true,
"From": "string",
"JitterBufferSize": "string",
"Label": "string",
"MaxParticipants": 42,
"Muted": true,
"Record": true,
"RecordingChannels": "string",
"RecordingStatusCallback": "https://example.com",
"RecordingStatusCallbackEvent": [
"string"
],
"RecordingStatusCallbackMethod": "HEAD",
"RecordingTrack": "string",
"Region": "string",
"SipAuthPassword": "string",
"SipAuthUsername": "string",
"StartConferenceOnEnter": true,
"StatusCallback": "https://example.com",
"StatusCallbackEvent": [
"string"
],
"StatusCallbackMethod": "HEAD",
"TimeLimit": 42,
"Timeout": 42,
"To": "string",
"WaitMethod": "HEAD",
"WaitUrl": "https://example.com"
}
{
"account_sid": "string",
"call_sid": "string",
"call_sid_to_coach": "string",
"coaching": true,
"conference_sid": "string",
"date_created": "string",
"date_updated": "string",
"end_conference_on_exit": true,
"hold": true,
"label": "string",
"muted": true,
"start_conference_on_enter": true,
"status": "queued",
"uri": "string"
}
{
"account_sid": "string",
"call_sid": "string",
"call_sid_to_coach": "string",
"coaching": true,
"conference_sid": "string",
"date_created": "string",
"date_updated": "string",
"end_conference_on_exit": true,
"hold": true,
"label": "string",
"muted": true,
"start_conference_on_enter": true,
"status": "queued",
"uri": "string"
}