Call Commands

The DID or SIP URI to dial out to. Multiple DID or SIP URIs can be provided using an array of strings

The from number to be used as the caller id presented to the destination (to number). The number should be in +E164 format.

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.

The ID of the Call Control App (formerly ID of the connection) to be used when dialing the destination.

The URL of a file to be played back to the callee when the call is answered. The URL can point to either a WAV or MP3 file. media_name and audio_url cannot be used together in one request.

The media_name of a file to be played back to the callee when the call is answered. 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.

The list of comma-separated codecs in a preferred order for the forked media to be received.

The number of seconds that Telnyx will wait for the call to be answered by the destination to which it is being called. 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 120 seconds.

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.

Enables Answering Machine Detection. Telnyx offers Premium and Standard detections. With Premium detection, when a call is answered, Telnyx runs real-time detection and sends a call.machine.premium.detection.ended webhook with one of the following results: human_residence, human_business, machine, silence, fax_detected or not_sure. If we detect a beep, we also send a call.machine.premium.greeting.ended webhook with the result of beep_detected. If we detect a beep before call.machine.premium.detection.ended we only send call.machine.premium.greeting.ended, and if we detect a beep after call.machine.premium.detection.ended, we send both webhooks. With Standard 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.

Optional configuration parameters to modify 'answering_machine_detection' performance.

Custom headers to be added to the SIP INVITE.

Use this field to set the Billing Group ID for the call. Must be a valid and existing Billing Group ID.

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

Use this field to avoid duplicate commands. Telnyx will ignore others Dial commands with the same command_id.

Use another call's control id for sharing the same call session id

Defines whether media should be encrypted on the call.

SIP Authentication username used for SIP challenges.

SIP Authentication password used for SIP challenges.

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

Defines SIP transport protocol to be used on the call.

Use this field to modify sound effects, for example adjust the pitch.

The destination WebSocket address where the stream is going to be delivered.

Specifies which track should be streamed.

Generate silence RTP packets when no transmission available.

Start transcription on answer.

Transcription configuration.

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

HTTP request type used for webhook_url.

Start recording automatically after an event. Disabled by default.

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

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

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).

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. The minimum value is 0. The default value is 0 (infinite).

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

Enables Dialogflow for the current call. The default value is false.

Use this field to set the Billing Group ID for the call. Must be a valid and existing Billing Group ID.

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

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

Custom headers to be added to the SIP INVITE response.

The list of comma-separated codecs in a preferred order for the forked media to be received.

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

Use this field to modify sound effects, for example adjust the pitch.

The destination WebSocket address where the stream is going to be delivered.

Specifies which track should be streamed.

Generate silence RTP packets when no transmission available.

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

The Call Control ID of the call you want to bridge with, can't be used together with queue parameter or video_room_id parameter.

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

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

The name of the queue you want to bridge with, can't be used together with call_control_id parameter or video_room_id parameter. Bridging with a queue means bridging with the first call in the queue. The call will always be removed from the queue regardless of whether bridging succeeds. Returns an error when the queue is empty.

The ID of the video room you want to bridge with, can't be used together with call_control_id parameter or queue parameter.

The additional parameter that will be passed to the video conference. It is a text field and the user can decide how to use it. For example, you can set the participant name or pass JSON text. It can be used only with video_room_id parameter.

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

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

The maximum number of calls allowed in the queue at a given time. Can't be modified for an existing queue.

The number of seconds after which the call will be removed from the queue.

The network target, udp:ip_address:port, where the call's RTP media packets should be forwarded. Both incoming and outgoing media packets will be delivered to the specified target, and information about the stream will be included in the encapsulation protocol header, including the direction (0 = inbound; 1 = outbound), leg (0 = A-leg; 1 = B-leg), and call_leg_id.

The network target, udp:ip_address:port, where the call's incoming RTP media packets should be forwarded.

Optionally specify a media type to stream. If decrypted selected, Telnyx will decrypt incoming SIP media before forking to the target. rx and tx are required fields if decrypted selected.

The network target, udp:ip_address:port, where the call's outgoing RTP media packets should be forwarded.

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

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

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

The minimum number of digits to fetch. This parameter has a minimum value of 1.

The maximum number of digits to fetch. This parameter has a maximum value of 128.

The number of milliseconds to wait to complete the request.

The number of milliseconds to wait for input between digits.

The number of milliseconds to wait for the first DTMF.

The digit used to terminate input if fewer than maximum_digits digits have been gathered.

A list of all digits accepted as valid.

An id that will be sent back in the corresponding call.gather.ended webhook. Will be randomly generated if not specified.

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

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

The URL of a file to be played back at the beginning of each prompt. The URL can point to either a WAV or MP3 file. media_name and audio_url cannot be used together in one request.

The media_name of a file to be played back at the beginning of each prompt. 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.

The URL of a file to play when digits don't match the valid_digits parameter or the number of digits is not between min and max. The URL can point to either a WAV or MP3 file. invalid_media_name and invalid_audio_url cannot be used together in one request.

The media_name of a file to be played back when digits don't match the valid_digits parameter or the number of digits is not between min and max. 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.

The minimum number of digits to fetch. This parameter has a minimum value of 1.

The maximum number of digits to fetch. This parameter has a maximum value of 128.

The maximum number of times the file should be played if there is no input from the user on the call.

The number of milliseconds to wait for a DTMF response after file playback ends before a replaying the sound file.

The digit used to terminate input if fewer than maximum_digits digits have been gathered.

A list of all digits accepted as valid.

The number of milliseconds to wait for input between digits.

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

The text or SSML to be converted into speech. There is a 3,000 character limit.

The text or SSML to be converted into speech when digits don't match the valid_digits parameter or the number of digits is not between min and max. There is a 3,000 character limit.

The type of the provided payload. The payload can either be plain text, or Speech Synthesis Markup Language (SSML).

This parameter impacts speech quality, language options and payload types. When using basic, only the en-US language and payload type text are allowed.

The gender of the voice used to speak back the text.

The language you want spoken.

The minimum number of digits to fetch. This parameter has a minimum value of 1.

The maximum number of digits to fetch. This parameter has a maximum value of 128.

The maximum number of times that a file should be played back if there is no input from the user on the call.

The number of milliseconds to wait for a DTMF response after speak ends before a replaying the sound file.