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Call Commands

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
post/calls/{call_control_id}/actions/transfer

Path parameters

call_control_idstring required

Unique identifier and token for controlling the call

Request body

tostring 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.

fromstring

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.

diversionstring

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.

from_display_namestring

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.

privacy'id' | 'none'

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.

audio_urlstring

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.

send_digits_on_answerstring

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.

early_mediaboolean

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

media_namestring

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.

timeout_secsinteger

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.

time_limit_secsinteger

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.

park_after_unbridgestring

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.

answering_machine_detection'premium' | 'detect' | 'detect_beep' | 'detect_words' | 'greeting_end' | '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.

client_statestring

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

target_leg_client_statestring

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

command_idstring

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

media_encryption'disabled' | 'SRTP' | 'DTLS'

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.

sip_auth_usernamestring

SIP Authentication username used for SIP challenges.

sip_auth_passwordstring

SIP Authentication password used for SIP challenges.

sip_transport_protocol'UDP' | 'TCP' | 'TLS'

Defines SIP transport protocol to be used on the call.

webhook_urlstring

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

webhook_url_method'POST' | 'GET'

HTTP request type used for webhook_url.

mute_dtmf'none' | 'both' | 'self' | 'opposite'

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

record'record-from-answer'

Start recording automatically after an event. Disabled by default.

record_channels'single' | 'dual'

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

record_format'wav' | 'mp3'

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

record_max_lengthinteger

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).

record_timeout_secsinteger

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).

record_track'both' | 'inbound' | 'outbound'

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).

record_trim'trim-silence'

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

record_custom_file_namestring

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

sip_region'US' | 'Europe' | 'Canada' | 'Australia' | 'Middle East'

Defines the SIP region to be used for the call.

preferred_codecsstring

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.

webhook_urlsobject

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.

webhook_urls_method'POST' | 'GET'

HTTP request method to invoke webhook_urls.

webhook_retries_policiesobject

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.

route_to_mobileboolean

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 request

{
  "to": "+18005550100 or sip:username@sip.telnyx.com;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"
}

Response

Successful response upon making a call control command.

Example response

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

Changes

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