Transfer call

Transfer a call to a new destination. If the transfer is unsuccessful, a call.hangup webhook for the other call (Leg B) will be sent indicating that the transfer could not be completed. The original call will remain active and may be issued additional commands, potentially transfering the call to an alternate destination.

Expected Webhooks:

call.initiated
call.bridged to Leg B
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
call.machine.premium.call_screening.detected if answering_machine_detection=premium_ios_call_screening_detection was requested and an Apple Call Screening tone was detected

POST

calls

{call_control_id}

actions

transfer

JavaScript

import Telnyx from 'telnyx';

const client = new Telnyx({
  apiKey: process.env['TELNYX_API_KEY'], // This is the default and can be omitted
});

const response = await client.calls.actions.transfer('call_control_id', {
  to: '+18005550100 or sip:[email protected];secure=srtp',
});

console.log(response.data);

import os
from telnyx import Telnyx

client = Telnyx(
    api_key=os.environ.get("TELNYX_API_KEY"),  # This is the default and can be omitted
)
response = client.calls.actions.transfer(
    call_control_id="call_control_id",
    to="+18005550100 or sip:[email protected];secure=srtp",
)
print(response.data)

package main

import (
	"context"
	"fmt"

	"github.com/team-telnyx/telnyx-go"
	"github.com/team-telnyx/telnyx-go/option"
)

func main() {
	client := telnyx.NewClient(
		option.WithAPIKey("My API Key"),
	)
	response, err := client.Calls.Actions.Transfer(
		context.TODO(),
		"call_control_id",
		telnyx.CallActionTransferParams{
			To: "+18005550100 or sip:[email protected];secure=srtp",
		},
	)
	if err != nil {
		panic(err.Error())
	}
	fmt.Printf("%+v\n", response.Data)
}

package com.telnyx.sdk.example;

import com.telnyx.sdk.client.TelnyxClient;
import com.telnyx.sdk.client.okhttp.TelnyxOkHttpClient;
import com.telnyx.sdk.models.calls.actions.ActionTransferParams;
import com.telnyx.sdk.models.calls.actions.ActionTransferResponse;

public final class Main {
    private Main() {}

    public static void main(String[] args) {
        TelnyxClient client = TelnyxOkHttpClient.fromEnv();

        ActionTransferParams params = ActionTransferParams.builder()
            .callControlId("call_control_id")
            .to("+18005550100 or sip:[email protected];secure=srtp")
            .build();
        ActionTransferResponse response = client.calls().actions().transfer(params);
    }
}

require "telnyx"

telnyx = Telnyx::Client.new(api_key: "My API Key")

response = telnyx.calls.actions.transfer(
  "call_control_id",
  to: "+18005550100 or sip:[email protected];secure=srtp"
)

puts(response)

<?php

require_once dirname(__DIR__) . '/vendor/autoload.php';

use Telnyx\Client;
use Telnyx\Core\Exceptions\APIException;

$client = new Client(apiKey: getenv('TELNYX_API_KEY') ?: 'My API Key');

try {
  $response = $client->calls->actions->transfer(
    'call_control_id',
    to: '+18005550100 or sip:[email protected];secure=srtp',
    answeringMachineDetection: 'detect',
    answeringMachineDetectionConfig: [
      'afterGreetingSilenceMillis' => 1000,
      'betweenWordsSilenceMillis' => 1000,
      'greetingDurationMillis' => 1000,
      'greetingSilenceDurationMillis' => 2000,
      'greetingTotalAnalysisTimeMillis' => 50000,
      'initialSilenceMillis' => 1000,
      'maximumNumberOfWords' => 1000,
      'maximumWordLengthMillis' => 2000,
      'silenceThreshold' => 512,
      'totalAnalysisTimeMillis' => 5000,
    ],
    audioURL: 'http://www.example.com/sounds/greeting.wav',
    clientState: 'aGF2ZSBhIG5pY2UgZGF5ID1d',
    commandID: '891510ac-f3e4-11e8-af5b-de00688a4901',
    customHeaders: [
      ['name' => 'head_1', 'value' => 'val_1'],
      ['name' => 'head_2', 'value' => 'val_2'],
    ],
    earlyMedia: true,
    from: '+18005550101',
    fromDisplayName: 'Company Name',
    mediaEncryption: 'SRTP',
    mediaName: 'my_media_uploaded_to_media_storage_api',
    muteDtmf: 'opposite',
    parkAfterUnbridge: 'self',
    preferredCodecs: 'G722,PCMU,PCMA,G729,OPUS,VP8,H264',
    privacy: 'id',
    record: 'record-from-answer',
    recordChannels: 'single',
    recordCustomFileName: 'my_recording_file_name',
    recordFormat: 'wav',
    recordMaxLength: 1000,
    recordTimeoutSecs: 100,
    recordTrack: 'outbound',
    recordTrim: 'trim-silence',
    sendDigitsOnAnswer: 'wwww200',
    sipAuthPassword: 'password',
    sipAuthUsername: 'username',
    sipHeaders: [['name' => 'User-to-User', 'value' => 'value']],
    sipRegion: 'Canada',
    sipTransportProtocol: 'TLS',
    soundModifications: [
      'octaves' => 0.1, 'pitch' => 0.8, 'semitone' => -2, 'track' => 'both'
    ],
    targetLegClientState: 'aGF2ZSBhIG5pY2UgZGF5ID1d',
    timeLimitSecs: 60,
    timeoutSecs: 60,
    webhookRetriesPolicies: [
      'call.answered' => ['retriesMs' => [1000, 2000, 5000]]
    ],
    webhookURL: 'https://www.example.com/server-b/',
    webhookURLMethod: 'POST',
    webhookURLs: [
      'call.answered' => 'https://www.example.com/webhooks/answered',
      'call.hangup' => 'https://www.example.com/webhooks/hangup',
    ],
    webhookURLsMethod: 'POST',
  );

  var_dump($response);
} catch (APIException $e) {
  echo $e->getMessage();
}

curl --request POST \
  --url https://api.telnyx.com/v2/calls/{call_control_id}/actions/transfer \
  --header 'Authorization: Bearer <token>' \
  --header 'Content-Type: application/json' \
  --data '
{
  "to": "+18005550100 or sip:[email protected];secure=srtp",
  "from": "+18005550101",
  "from_display_name": "Company Name",
  "audio_url": "http://www.example.com/sounds/greeting.wav",
  "send_digits_on_answer": "wwww200",
  "timeout_secs": 60,
  "time_limit_secs": 60,
  "webhook_url": "https://www.example.com/server-b/",
  "webhook_url_method": "POST",
  "answering_machine_detection": "detect",
  "answering_machine_detection_config": {
    "total_analysis_time_millis": 5000,
    "after_greeting_silence_millis": 1000,
    "between_words_silence_millis": 1000,
    "greeting_duration_millis": 1000,
    "initial_silence_millis": 1000,
    "maximum_number_of_words": 1000,
    "maximum_word_length_millis": 2000,
    "silence_threshold": 512,
    "greeting_total_analysis_time_millis": 50000,
    "greeting_silence_duration_millis": 2000
  },
  "custom_headers": [
    {
      "name": "head_1",
      "value": "val_1"
    },
    {
      "name": "head_2",
      "value": "val_2"
    }
  ],
  "client_state": "aGF2ZSBhIG5pY2UgZGF5ID1d",
  "command_id": "891510ac-f3e4-11e8-af5b-de00688a4901",
  "early_media": true,
  "media_encryption": "SRTP",
  "sip_auth_username": "username",
  "sip_auth_password": "password",
  "sip_headers": [
    {
      "name": "User-to-User",
      "value": "value"
    }
  ],
  "sip_transport_protocol": "TLS"
}
'

{
  "data": {
    "result": "ok"
  }
}

Authorizations

Authorization

string

header

required

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

Path Parameters

call_control_id

string

required

Unique identifier and token for controlling the call

Body

application/json

Transfer call request

string

required

The DID or SIP URI to dial out to. For SIP URI destinations, append ;secure=true or ;secure=srtp to enable SRTP media encryption for that endpoint, or ;secure=dtls to enable DTLS media encryption for that endpoint. If media_encryption is set to SRTP or DTLS, it takes precedence over any per-endpoint secure URI parameter. You may also append a comma followed by DTMF digits (e.g. +18004247767,200) to play those digits as DTMF once the transfer destination answers — equivalent to setting send_digits_on_answer separately. If both are present, the explicit send_digits_on_answer parameter takes precedence.

Example:

"+18005550100 or sip:[email protected];secure=srtp"

from

string

The from number to be used as the caller id presented to the destination (to number). The number should be in +E164 format. This attribute will default to the to number of the original call if omitted.

Example:

"+18005550101"

diversion

string

The to number of an active inbound call, in +E164 format. Telnyx checks whether there is currently an active inbound call where to matches this diversion value and from matches the from number supplied for this request. If such a call exists, the from number is treated as verified (since it is already on an active inbound call to you) and can be used as the caller id for this outbound call.

Example:

"+18005550101"

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"

privacy

enum<string>

Indicates the privacy level to be used for the call. When set to id, caller ID information (name and number) will be hidden from the called party. When set to none or omitted, caller ID will be shown normally.

Available options:

id,

none

Example:

"id"

audio_url

string

The URL of a file to be played back when the transfer destination answers before bridging the call. 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"

send_digits_on_answer

string

DTMF digits to send automatically after the transfer destination answers. Useful for reaching an extension behind an IVR (e.g. "200" to dial extension 200 once the called party picks up). Allowed characters: 0-9, A-D, w (0.5s pause), W (1s pause), *, #. Maximum 64 characters. When omitted, no automatic DTMF is sent. May also be supplied inline by appending ,<digits> to to (e.g. to=+18004247767,200); if both forms are present, this explicit field takes precedence.

Example:

"wwww200"

early_media

boolean

default:true

If set to false, early media will not be passed to the originating leg.

Example:

false

media_name

string

The media_name of a file to be played back when the transfer destination answers before bridging the call. 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"

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 transferred. 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

park_after_unbridge

string

Specifies behavior after the bridge ends (i.e. the opposite leg either hangs up or is transferred). 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.

Example:

"self"

answering_machine_detection

enum<string>

default:disabled

Enables Answering Machine 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. If answering_machine_detection is set to premium_ios_call_screening_detection, Premium AMD runs with iOS Call Screening support: after an initial machine result, Telnyx listens for the iOS call-screening prompt to end or for an Apple Call Screening tone, sends call.machine.premium.greeting.ended with result=prompt_ended or call.machine.premium.call_screening.detected with result=screening respectively. When the Apple Call Screening tone is detected, Premium AMD is restarted on the screened call and a call.machine.premium.detection.ended webhook with the post-screening classification follows.

Available options:

premium,

premium_ios_call_screening_detection,

detect,

detect_beep,

detect_words,

greeting_end,

disabled

answering_machine_detection_config

object

Optional configuration parameters to modify 'answering_machine_detection' performance. Only total_analysis_time_millis and greeting_duration_millis parameters are applicable when premium is selected as answering_machine_detection. prompt_end_timeout_millis is additionally applicable when premium_ios_call_screening_detection is selected.

Show child attributes

custom_headers

Custom SIP Header · object[]

Custom headers to be added to the SIP INVITE.

Show child attributes

Example:

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

client_state

string

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

Example:

"aGF2ZSBhIG5pY2UgZGF5ID1d"

target_leg_client_state

string

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

Example:

"aGF2ZSBhIG5pY2UgZGF5ID1d"

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"

media_encryption

enum<string>

default:disabled

Defines whether media should be encrypted on the new call leg. For SIP URI destinations, media encryption can also be requested per endpoint with the secure URI parameter: ;secure=true or ;secure=srtp enables SRTP, and ;secure=dtls enables DTLS. This parameter, when set to SRTP or DTLS, takes precedence over the per-endpoint secure value.

Available options:

disabled,

SRTP,

DTLS

sip_auth_username

string

SIP Authentication username used for SIP challenges.

Example:

"username"

sip_auth_password

string

SIP Authentication password used for SIP challenges.

Example:

"password"

sip_headers

SIP Header · object[]

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

Show child attributes

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

Example:

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

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"

mute_dtmf

enum<string>

default:none

When enabled, DTMF tones are not passed to the call participant. The webhooks containing the DTMF information will be sent.

Available options:

none,

both,

self,

opposite

Example:

"opposite"

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"

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"

preferred_codecs

string

The list of comma-separated codecs in order of preference to be used during the call. The codecs supported are G722, PCMU, PCMA, G729, OPUS, VP8, H264, AMR-WB.

Example:

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

webhook_urls

object

A map of event types to arrays of webhook URLs. When an event of the specified type occurs, the webhook URLs associated with that event type will be called instead of webhook_url. Events not mapped here will use the default webhook_url.

Show child attributes

Example:

{
  "call.answered": ["https://www.example.com/webhooks/answered"],
  "call.hangup": ["https://www.example.com/webhooks/hangup"]
}

webhook_urls_method

enum<string>

default:POST

HTTP request method to invoke webhook_urls.

Available options:

POST,

GET

Example:

"POST"

webhook_retries_policies

object

A map of event types to retry policies. Each retry policy contains an array of retries_ms specifying the delays between retry attempts in milliseconds. Maximum 5 retries, total delay cannot exceed 60 seconds.

Show child attributes

Example:

{
  "call.answered": { "retries_ms": [1000, 2000, 5000] }
}

route_to_mobile

boolean

default:false

When set to true, routes the call directly to the mobile device associated with the destination Telnyx Mobile number, bypassing Inbound Calls Interception configured in the Telnyx Portal under Mobile Numbers → select the number → Voice → Call Interception. Use this when transferring an intercepted call to the mobile device to prevent the call from being intercepted again. Defaults to false.

Example:

true

Response

Successful response upon making a call control command.

data

Call Control Command Result · object

Show child attributes

Example:

{ "result": "ok" }

JavaScript

import Telnyx from 'telnyx';

const client = new Telnyx({
  apiKey: process.env['TELNYX_API_KEY'], // This is the default and can be omitted
});

const response = await client.calls.actions.transfer('call_control_id', {
  to: '+18005550100 or sip:[email protected];secure=srtp',
});

console.log(response.data);

import os
from telnyx import Telnyx

client = Telnyx(
    api_key=os.environ.get("TELNYX_API_KEY"),  # This is the default and can be omitted
)
response = client.calls.actions.transfer(
    call_control_id="call_control_id",
    to="+18005550100 or sip:[email protected];secure=srtp",
)
print(response.data)

package main

import (
	"context"
	"fmt"

	"github.com/team-telnyx/telnyx-go"
	"github.com/team-telnyx/telnyx-go/option"
)

func main() {
	client := telnyx.NewClient(
		option.WithAPIKey("My API Key"),
	)
	response, err := client.Calls.Actions.Transfer(
		context.TODO(),
		"call_control_id",
		telnyx.CallActionTransferParams{
			To: "+18005550100 or sip:[email protected];secure=srtp",
		},
	)
	if err != nil {
		panic(err.Error())
	}
	fmt.Printf("%+v\n", response.Data)
}

package com.telnyx.sdk.example;

import com.telnyx.sdk.client.TelnyxClient;
import com.telnyx.sdk.client.okhttp.TelnyxOkHttpClient;
import com.telnyx.sdk.models.calls.actions.ActionTransferParams;
import com.telnyx.sdk.models.calls.actions.ActionTransferResponse;

public final class Main {
    private Main() {}

    public static void main(String[] args) {
        TelnyxClient client = TelnyxOkHttpClient.fromEnv();

        ActionTransferParams params = ActionTransferParams.builder()
            .callControlId("call_control_id")
            .to("+18005550100 or sip:[email protected];secure=srtp")
            .build();
        ActionTransferResponse response = client.calls().actions().transfer(params);
    }
}

require "telnyx"

telnyx = Telnyx::Client.new(api_key: "My API Key")

response = telnyx.calls.actions.transfer(
  "call_control_id",
  to: "+18005550100 or sip:[email protected];secure=srtp"
)

puts(response)

<?php

require_once dirname(__DIR__) . '/vendor/autoload.php';

use Telnyx\Client;
use Telnyx\Core\Exceptions\APIException;

$client = new Client(apiKey: getenv('TELNYX_API_KEY') ?: 'My API Key');

try {
  $response = $client->calls->actions->transfer(
    'call_control_id',
    to: '+18005550100 or sip:[email protected];secure=srtp',
    answeringMachineDetection: 'detect',
    answeringMachineDetectionConfig: [
      'afterGreetingSilenceMillis' => 1000,
      'betweenWordsSilenceMillis' => 1000,
      'greetingDurationMillis' => 1000,
      'greetingSilenceDurationMillis' => 2000,
      'greetingTotalAnalysisTimeMillis' => 50000,
      'initialSilenceMillis' => 1000,
      'maximumNumberOfWords' => 1000,
      'maximumWordLengthMillis' => 2000,
      'silenceThreshold' => 512,
      'totalAnalysisTimeMillis' => 5000,
    ],
    audioURL: 'http://www.example.com/sounds/greeting.wav',
    clientState: 'aGF2ZSBhIG5pY2UgZGF5ID1d',
    commandID: '891510ac-f3e4-11e8-af5b-de00688a4901',
    customHeaders: [
      ['name' => 'head_1', 'value' => 'val_1'],
      ['name' => 'head_2', 'value' => 'val_2'],
    ],
    earlyMedia: true,
    from: '+18005550101',
    fromDisplayName: 'Company Name',
    mediaEncryption: 'SRTP',
    mediaName: 'my_media_uploaded_to_media_storage_api',
    muteDtmf: 'opposite',
    parkAfterUnbridge: 'self',
    preferredCodecs: 'G722,PCMU,PCMA,G729,OPUS,VP8,H264',
    privacy: 'id',
    record: 'record-from-answer',
    recordChannels: 'single',
    recordCustomFileName: 'my_recording_file_name',
    recordFormat: 'wav',
    recordMaxLength: 1000,
    recordTimeoutSecs: 100,
    recordTrack: 'outbound',
    recordTrim: 'trim-silence',
    sendDigitsOnAnswer: 'wwww200',
    sipAuthPassword: 'password',
    sipAuthUsername: 'username',
    sipHeaders: [['name' => 'User-to-User', 'value' => 'value']],
    sipRegion: 'Canada',
    sipTransportProtocol: 'TLS',
    soundModifications: [
      'octaves' => 0.1, 'pitch' => 0.8, 'semitone' => -2, 'track' => 'both'
    ],
    targetLegClientState: 'aGF2ZSBhIG5pY2UgZGF5ID1d',
    timeLimitSecs: 60,
    timeoutSecs: 60,
    webhookRetriesPolicies: [
      'call.answered' => ['retriesMs' => [1000, 2000, 5000]]
    ],
    webhookURL: 'https://www.example.com/server-b/',
    webhookURLMethod: 'POST',
    webhookURLs: [
      'call.answered' => 'https://www.example.com/webhooks/answered',
      'call.hangup' => 'https://www.example.com/webhooks/hangup',
    ],
    webhookURLsMethod: 'POST',
  );

  var_dump($response);
} catch (APIException $e) {
  echo $e->getMessage();
}

curl --request POST \
  --url https://api.telnyx.com/v2/calls/{call_control_id}/actions/transfer \
  --header 'Authorization: Bearer <token>' \
  --header 'Content-Type: application/json' \
  --data '
{
  "to": "+18005550100 or sip:[email protected];secure=srtp",
  "from": "+18005550101",
  "from_display_name": "Company Name",
  "audio_url": "http://www.example.com/sounds/greeting.wav",
  "send_digits_on_answer": "wwww200",
  "timeout_secs": 60,
  "time_limit_secs": 60,
  "webhook_url": "https://www.example.com/server-b/",
  "webhook_url_method": "POST",
  "answering_machine_detection": "detect",
  "answering_machine_detection_config": {
    "total_analysis_time_millis": 5000,
    "after_greeting_silence_millis": 1000,
    "between_words_silence_millis": 1000,
    "greeting_duration_millis": 1000,
    "initial_silence_millis": 1000,
    "maximum_number_of_words": 1000,
    "maximum_word_length_millis": 2000,
    "silence_threshold": 512,
    "greeting_total_analysis_time_millis": 50000,
    "greeting_silence_duration_millis": 2000
  },
  "custom_headers": [
    {
      "name": "head_1",
      "value": "val_1"
    },
    {
      "name": "head_2",
      "value": "val_2"
    }
  ],
  "client_state": "aGF2ZSBhIG5pY2UgZGF5ID1d",
  "command_id": "891510ac-f3e4-11e8-af5b-de00688a4901",
  "early_media": true,
  "media_encryption": "SRTP",
  "sip_auth_username": "username",
  "sip_auth_password": "password",
  "sip_headers": [
    {
      "name": "User-to-User",
      "value": "value"
    }
  ],
  "sip_transport_protocol": "TLS"
}
'

{
  "data": {
    "result": "ok"
  }
}

Voice API Fundamentals

Voice Features

AI Capabilities

Tutorials

Regional & Restrictions

API Reference

For AI Agents

Transfer call

Authorizations

Path Parameters

Body

Response