telnyx 2.1.0 → 3.0.0-alpha

package/src/resources/calls/actions.ts

@@ -0,0 +1,3622 @@
+// File generated from our OpenAPI spec by Stainless. See CONTRIBUTING.md for details.
+
+import { APIResource } from '../../core/resource';
+import * as CallsAPI from './calls';
+import * as AssistantsAPI from '../ai/assistants/assistants';
+import { APIPromise } from '../../core/api-promise';
+import { RequestOptions } from '../../internal/request-options';
+import { path } from '../../internal/utils/path';
+
+export class Actions extends APIResource {
+  /**
+   * Answer an incoming call. You must issue this command before executing subsequent
+   * commands on an incoming call.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/answer-call#callbacks)
+   * below):**
+   *
+   * - `call.answered`
+   * - `streaming.started`, `streaming.stopped` or `streaming.failed` if `stream_url`
+   *   was set
+   *
+   * When the `record` parameter is set to `record-from-answer`, the response will
+   * include a `recording_id` field.
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.answer(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  answer(
+    callControlID: string,
+    body: ActionAnswerParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionAnswerResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/answer`, { body, ...options });
+  }
+
+  /**
+   * Bridge two call control calls.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/bridge-call#callbacks)
+   * below):**
+   *
+   * - `call.bridged` for Leg A
+   * - `call.bridged` for Leg B
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.bridge(
+   *   'call_control_id',
+   *   {
+   *     body_call_control_id:
+   *       'v3:MdI91X4lWFEs7IgbBEOT9M4AigoY08M0WWZFISt1Yw2axZ_IiE4pqg',
+   *   },
+   * );
+   * ```
+   */
+  bridge(
+    callControlID: string,
+    body: ActionBridgeParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionBridgeResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/bridge`, { body, ...options });
+  }
+
+  /**
+   * Put the call in a queue.
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.enqueue(
+   *   'call_control_id',
+   *   { queue_name: 'support' },
+   * );
+   * ```
+   */
+  enqueue(
+    callControlID: string,
+    body: ActionEnqueueParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionEnqueueResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/enqueue`, { body, ...options });
+  }
+
+  /**
+   * Gather DTMF signals to build interactive menus.
+   *
+   * You can pass a list of valid digits. The `Answer` command must be issued before
+   * the `gather` command.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/gather-call#callbacks)
+   * below):**
+   *
+   * - `call.dtmf.received` (you may receive many of these webhooks)
+   * - `call.gather.ended`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.gather(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  gather(
+    callControlID: string,
+    body: ActionGatherParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionGatherResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/gather`, { body, ...options });
+  }
+
+  /**
+   * Gather parameters defined in the request payload using a voice assistant.
+   *
+   * You can pass parameters described as a JSON Schema object and the voice
+   * assistant will attempt to gather these informations.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/call-gather-using-ai#callbacks)
+   * below):**
+   *
+   * - `call.ai_gather.ended`
+   * - `call.conversation.ended`
+   * - `call.ai_gather.partial_results` (if `send_partial_results` is set to `true`)
+   * - `call.ai_gather.message_history_updated` (if `send_message_history_updates` is
+   *   set to `true`)
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.gatherUsingAI(
+   *   'call_control_id',
+   *   {
+   *     parameters: {
+   *       properties: {
+   *         age: {
+   *           description: 'The age of the customer.',
+   *           type: 'integer',
+   *         },
+   *         location: {
+   *           description: 'The location of the customer.',
+   *           type: 'string',
+   *         },
+   *       },
+   *       required: ['age', 'location'],
+   *       type: 'object',
+   *     },
+   *   },
+   * );
+   * ```
+   */
+  gatherUsingAI(
+    callControlID: string,
+    body: ActionGatherUsingAIParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionGatherUsingAIResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/gather_using_ai`, { body, ...options });
+  }
+
+  /**
+   * Play an audio file on the call until the required DTMF signals are gathered to
+   * build interactive menus.
+   *
+   * You can pass a list of valid digits along with an 'invalid_audio_url', which
+   * will be played back at the beginning of each prompt. Playback will be
+   * interrupted when a DTMF signal is received. The
+   * `Answer command must be issued before the `gather_using_audio` command.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/gather-using-audio#callbacks)
+   * below):**
+   *
+   * - `call.playback.started`
+   * - `call.playback.ended`
+   * - `call.dtmf.received` (you may receive many of these webhooks)
+   * - `call.gather.ended`
+   *
+   * @example
+   * ```ts
+   * const response =
+   *   await client.calls.actions.gatherUsingAudio(
+   *     'call_control_id',
+   *   );
+   * ```
+   */
+  gatherUsingAudio(
+    callControlID: string,
+    body: ActionGatherUsingAudioParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionGatherUsingAudioResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/gather_using_audio`, { body, ...options });
+  }
+
+  /**
+   * Convert text to speech and play it on the call until the required DTMF signals
+   * are gathered to build interactive menus.
+   *
+   * You can pass a list of valid digits along with an 'invalid_payload', which will
+   * be played back at the beginning of each prompt. Speech will be interrupted when
+   * a DTMF signal is received. The `Answer` command must be issued before the
+   * `gather_using_speak` command.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/gather-using-speak#callbacks)
+   * below):**
+   *
+   * - `call.dtmf.received` (you may receive many of these webhooks)
+   * - `call.gather.ended`
+   *
+   * @example
+   * ```ts
+   * const response =
+   *   await client.calls.actions.gatherUsingSpeak(
+   *     'call_control_id',
+   *     { payload: 'say this on call', voice: 'male' },
+   *   );
+   * ```
+   */
+  gatherUsingSpeak(
+    callControlID: string,
+    body: ActionGatherUsingSpeakParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionGatherUsingSpeakResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/gather_using_speak`, { body, ...options });
+  }
+
+  /**
+   * Hang up the call.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/hangup-call#callbacks)
+   * below):**
+   *
+   * - `call.hangup`
+   * - `call.recording.saved`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.hangup(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  hangup(
+    callControlID: string,
+    body: ActionHangupParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionHangupResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/hangup`, { body, ...options });
+  }
+
+  /**
+   * Removes the call from a queue.
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.leaveQueue(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  leaveQueue(
+    callControlID: string,
+    body: ActionLeaveQueueParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionLeaveQueueResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/leave_queue`, { body, ...options });
+  }
+
+  /**
+   * Pause recording the call. Recording can be resumed via Resume recording command.
+   *
+   * **Expected Webhooks:**
+   *
+   * There are no webhooks associated with this command.
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.pauseRecording(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  pauseRecording(
+    callControlID: string,
+    body: ActionPauseRecordingParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionPauseRecordingResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/record_pause`, { body, ...options });
+  }
+
+  /**
+   * Initiate a SIP Refer on a Call Control call. You can initiate a SIP Refer at any
+   * point in the duration of a call.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/refer-call#callbacks)
+   * below):**
+   *
+   * - `call.refer.started`
+   * - `call.refer.completed`
+   * - `call.refer.failed`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.refer(
+   *   'call_control_id',
+   *   {
+   *     sip_address: 'sip:username@sip.non-telnyx-address.com',
+   *   },
+   * );
+   * ```
+   */
+  refer(
+    callControlID: string,
+    body: ActionReferParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionReferResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/refer`, { body, ...options });
+  }
+
+  /**
+   * Reject an incoming call.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/reject-call#callbacks)
+   * below):**
+   *
+   * - `call.hangup`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.reject(
+   *   'call_control_id',
+   *   { cause: 'USER_BUSY' },
+   * );
+   * ```
+   */
+  reject(
+    callControlID: string,
+    body: ActionRejectParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionRejectResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/reject`, { body, ...options });
+  }
+
+  /**
+   * Resume recording the call.
+   *
+   * **Expected Webhooks:**
+   *
+   * There are no webhooks associated with this command.
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.resumeRecording(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  resumeRecording(
+    callControlID: string,
+    body: ActionResumeRecordingParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionResumeRecordingResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/record_resume`, { body, ...options });
+  }
+
+  /**
+   * Sends DTMF tones from this leg. DTMF tones will be heard by the other end of the
+   * call.
+   *
+   * **Expected Webhooks:**
+   *
+   * There are no webhooks associated with this command.
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.sendDtmf(
+   *   'call_control_id',
+   *   { digits: '1www2WABCDw9' },
+   * );
+   * ```
+   */
+  sendDtmf(
+    callControlID: string,
+    body: ActionSendDtmfParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionSendDtmfResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/send_dtmf`, { body, ...options });
+  }
+
+  /**
+   * Sends SIP info from this leg.
+   *
+   * **Expected Webhooks:**
+   *
+   * - `call.sip_info.received` (to be received on the target call leg)
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.sendSipInfo(
+   *   'call_control_id',
+   *   {
+   *     body: '{"key": "value", "numValue": 100}',
+   *     content_type: 'application/json',
+   *   },
+   * );
+   * ```
+   */
+  sendSipInfo(
+    callControlID: string,
+    body: ActionSendSipInfoParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionSendSipInfoResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/send_sip_info`, { body, ...options });
+  }
+
+  /**
+   * Convert text to speech and play it back on the call. If multiple speak text
+   * commands are issued consecutively, the audio files will be placed in a queue
+   * awaiting playback.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/speak-call#callbacks)
+   * below):**
+   *
+   * - `call.speak.started`
+   * - `call.speak.ended`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.speak(
+   *   'call_control_id',
+   *   { payload: 'Say this on the call', voice: 'female' },
+   * );
+   * ```
+   */
+  speak(
+    callControlID: string,
+    body: ActionSpeakParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionSpeakResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/speak`, { body, ...options });
+  }
+
+  /**
+   * Start an AI assistant on the call.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/call-start-ai-assistant#callbacks)
+   * below):**
+   *
+   * - `call.conversation.ended`
+   * - `call.conversation_insights.generated`
+   *
+   * @example
+   * ```ts
+   * const response =
+   *   await client.calls.actions.startAIAssistant(
+   *     'call_control_id',
+   *   );
+   * ```
+   */
+  startAIAssistant(
+    callControlID: string,
+    body: ActionStartAIAssistantParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStartAIAssistantResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/ai_assistant_start`, { body, ...options });
+  }
+
+  /**
+   * Call forking allows you to stream the media from a call to a specific target in
+   * realtime. This stream can be used to enable realtime audio analysis to support a
+   * variety of use cases, including fraud detection, or the creation of AI-generated
+   * audio responses. Requests must specify either the `target` attribute or the `rx`
+   * and `tx` attributes.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/start-call-fork#callbacks)
+   * below):**
+   *
+   * - `call.fork.started`
+   * - `call.fork.stopped`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.startForking(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  startForking(
+    callControlID: string,
+    body: ActionStartForkingParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStartForkingResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/fork_start`, { body, ...options });
+  }
+
+  /**
+   * Noise Suppression Start (BETA)
+   *
+   * @example
+   * ```ts
+   * const response =
+   *   await client.calls.actions.startNoiseSuppression(
+   *     'call_control_id',
+   *   );
+   * ```
+   */
+  startNoiseSuppression(
+    callControlID: string,
+    body: ActionStartNoiseSuppressionParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStartNoiseSuppressionResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/suppression_start`, { body, ...options });
+  }
+
+  /**
+   * Play an audio file on the call. If multiple play audio commands are issued
+   * consecutively, the audio files will be placed in a queue awaiting playback.
+   *
+   * _Notes:_
+   *
+   * - When `overlay` is enabled, `target_legs` is limited to `self`.
+   * - A customer cannot Play Audio with `overlay=true` unless there is a Play Audio
+   *   with `overlay=false` actively playing.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/start-call-playback#callbacks)
+   * below):**
+   *
+   * - `call.playback.started`
+   * - `call.playback.ended`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.startPlayback(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  startPlayback(
+    callControlID: string,
+    body: ActionStartPlaybackParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStartPlaybackResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/playback_start`, { body, ...options });
+  }
+
+  /**
+   * Start recording the call. Recording will stop on call hang-up, or can be
+   * initiated via the Stop Recording command.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/start-call-record#callbacks)
+   * below):**
+   *
+   * - `call.recording.saved`
+   * - `call.recording.transcription.saved`
+   * - `call.recording.error`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.startRecording(
+   *   'call_control_id',
+   *   { channels: 'single', format: 'wav' },
+   * );
+   * ```
+   */
+  startRecording(
+    callControlID: string,
+    body: ActionStartRecordingParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStartRecordingResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/record_start`, { body, ...options });
+  }
+
+  /**
+   * Start siprec session to configured in SIPREC connector SRS.
+   *
+   * **Expected Webhooks:**
+   *
+   * - `siprec.started`
+   * - `siprec.stopped`
+   * - `siprec.failed`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.startSiprec(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  startSiprec(
+    callControlID: string,
+    body: ActionStartSiprecParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStartSiprecResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/siprec_start`, { body, ...options });
+  }
+
+  /**
+   * Start streaming the media from a call to a specific WebSocket address or
+   * Dialogflow connection in near-realtime. Audio will be delivered as
+   * base64-encoded RTP payload (raw audio), wrapped in JSON payloads.
+   *
+   * Please find more details about media streaming messages specification under the
+   * [link](https://developers.telnyx.com/docs/voice/programmable-voice/media-streaming).
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.startStreaming(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  startStreaming(
+    callControlID: string,
+    body: ActionStartStreamingParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStartStreamingResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/streaming_start`, { body, ...options });
+  }
+
+  /**
+   * Start real-time transcription. Transcription will stop on call hang-up, or can
+   * be initiated via the Transcription stop command.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/start-call-transcription#callbacks)
+   * below):**
+   *
+   * - `call.transcription`
+   *
+   * @example
+   * ```ts
+   * const response =
+   *   await client.calls.actions.startTranscription(
+   *     'call_control_id',
+   *   );
+   * ```
+   */
+  startTranscription(
+    callControlID: string,
+    body: ActionStartTranscriptionParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStartTranscriptionResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/transcription_start`, { body, ...options });
+  }
+
+  /**
+   * Stop an AI assistant on the call.
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.stopAIAssistant(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  stopAIAssistant(
+    callControlID: string,
+    body: ActionStopAIAssistantParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStopAIAssistantResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/ai_assistant_stop`, { body, ...options });
+  }
+
+  /**
+   * Stop forking a call.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/stop-call-fork#callbacks)
+   * below):**
+   *
+   * - `call.fork.stopped`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.stopForking(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  stopForking(
+    callControlID: string,
+    body: ActionStopForkingParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStopForkingResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/fork_stop`, { body, ...options });
+  }
+
+  /**
+   * Stop current gather.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/stop-call-gather#callbacks)
+   * below):**
+   *
+   * - `call.gather.ended`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.stopGather(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  stopGather(
+    callControlID: string,
+    body: ActionStopGatherParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStopGatherResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/gather_stop`, { body, ...options });
+  }
+
+  /**
+   * Noise Suppression Stop (BETA)
+   *
+   * @example
+   * ```ts
+   * const response =
+   *   await client.calls.actions.stopNoiseSuppression(
+   *     'call_control_id',
+   *   );
+   * ```
+   */
+  stopNoiseSuppression(
+    callControlID: string,
+    body: ActionStopNoiseSuppressionParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStopNoiseSuppressionResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/suppression_stop`, { body, ...options });
+  }
+
+  /**
+   * Stop audio being played on the call.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/stop-call-playback#callbacks)
+   * below):**
+   *
+   * - `call.playback.ended` or `call.speak.ended`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.stopPlayback(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  stopPlayback(
+    callControlID: string,
+    body: ActionStopPlaybackParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStopPlaybackResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/playback_stop`, { body, ...options });
+  }
+
+  /**
+   * Stop recording the call.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/stop-call-recording#callbacks)
+   * below):**
+   *
+   * - `call.recording.saved`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.stopRecording(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  stopRecording(
+    callControlID: string,
+    body: ActionStopRecordingParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStopRecordingResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/record_stop`, { body, ...options });
+  }
+
+  /**
+   * Stop SIPREC session.
+   *
+   * **Expected Webhooks:**
+   *
+   * - `siprec.stopped`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.stopSiprec(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  stopSiprec(
+    callControlID: string,
+    body: ActionStopSiprecParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStopSiprecResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/siprec_stop`, { body, ...options });
+  }
+
+  /**
+   * Stop streaming a call to a WebSocket.
+   *
+   * **Expected Webhooks (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/stop-call-streaming#callbacks)
+   * below):**
+   *
+   * - `streaming.stopped`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.stopStreaming(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  stopStreaming(
+    callControlID: string,
+    body: ActionStopStreamingParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStopStreamingResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/streaming_stop`, { body, ...options });
+  }
+
+  /**
+   * Stop real-time transcription.
+   *
+   * @example
+   * ```ts
+   * const response =
+   *   await client.calls.actions.stopTranscription(
+   *     'call_control_id',
+   *   );
+   * ```
+   */
+  stopTranscription(
+    callControlID: string,
+    body: ActionStopTranscriptionParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStopTranscriptionResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/transcription_stop`, { body, ...options });
+  }
+
+  /**
+   * Switch the supervisor role for a bridged call. This allows switching between
+   * different supervisor modes during an active call
+   *
+   * @example
+   * ```ts
+   * const response =
+   *   await client.calls.actions.switchSupervisorRole(
+   *     'call_control_id',
+   *     { role: 'barge' },
+   *   );
+   * ```
+   */
+  switchSupervisorRole(
+    callControlID: string,
+    body: ActionSwitchSupervisorRoleParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionSwitchSupervisorRoleResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/switch_supervisor_role`, {
+      body,
+      ...options,
+    });
+  }
+
+  /**
+   * 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 (see
+   * [callback schema](https://developers.telnyx.com/api/call-control/transfer-call#callbacks)
+   * below):**
+   *
+   * - `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
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.transfer(
+   *   'call_control_id',
+   *   { to: '+18005550100 or sip:username@sip.telnyx.com' },
+   * );
+   * ```
+   */
+  transfer(
+    callControlID: string,
+    body: ActionTransferParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionTransferResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/transfer`, { body, ...options });
+  }
+
+  /**
+   * Updates client state
+   *
+   * @example
+   * ```ts
+   * const response =
+   *   await client.calls.actions.updateClientState(
+   *     'call_control_id',
+   *     { client_state: 'aGF2ZSBhIG5pY2UgZGF5ID1d' },
+   *   );
+   * ```
+   */
+  updateClientState(
+    callControlID: string,
+    body: ActionUpdateClientStateParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionUpdateClientStateResponse> {
+    return this._client.put(path`/calls/${callControlID}/actions/client_state_update`, { body, ...options });
+  }
+}
+
+export type AwsVoiceSettings = unknown;
+
+export interface CallControlCommandResult {
+  result?: string;
+}
+
+export interface ElevenLabsVoiceSettings {
+  /**
+   * The `identifier` for an integration secret
+   * [/v2/integration_secrets](https://developers.telnyx.com/api/secrets-manager/integration-secrets/create-integration-secret)
+   * that refers to your ElevenLabs API key. Warning: Free plans are unlikely to work
+   * with this integration.
+   */
+  api_key_ref?: string;
+}
+
+/**
+ * Language to use for speech recognition
+ */
+export type GoogleTranscriptionLanguage =
+  | 'af'
+  | 'sq'
+  | 'am'
+  | 'ar'
+  | 'hy'
+  | 'az'
+  | 'eu'
+  | 'bn'
+  | 'bs'
+  | 'bg'
+  | 'my'
+  | 'ca'
+  | 'yue'
+  | 'zh'
+  | 'hr'
+  | 'cs'
+  | 'da'
+  | 'nl'
+  | 'en'
+  | 'et'
+  | 'fil'
+  | 'fi'
+  | 'fr'
+  | 'gl'
+  | 'ka'
+  | 'de'
+  | 'el'
+  | 'gu'
+  | 'iw'
+  | 'hi'
+  | 'hu'
+  | 'is'
+  | 'id'
+  | 'it'
+  | 'ja'
+  | 'jv'
+  | 'kn'
+  | 'kk'
+  | 'km'
+  | 'ko'
+  | 'lo'
+  | 'lv'
+  | 'lt'
+  | 'mk'
+  | 'ms'
+  | 'ml'
+  | 'mr'
+  | 'mn'
+  | 'ne'
+  | 'no'
+  | 'fa'
+  | 'pl'
+  | 'pt'
+  | 'pa'
+  | 'ro'
+  | 'ru'
+  | 'rw'
+  | 'sr'
+  | 'si'
+  | 'sk'
+  | 'sl'
+  | 'ss'
+  | 'st'
+  | 'es'
+  | 'su'
+  | 'sw'
+  | 'sv'
+  | 'ta'
+  | 'te'
+  | 'th'
+  | 'tn'
+  | 'tr'
+  | 'ts'
+  | 'uk'
+  | 'ur'
+  | 'uz'
+  | 've'
+  | 'vi'
+  | 'xh'
+  | 'zu';
+
+/**
+ * Settings for handling user interruptions during assistant speech
+ */
+export interface InterruptionSettings {
+  /**
+   * When true, allows users to interrupt the assistant while speaking
+   */
+  enable?: boolean;
+}
+
+export type Loopcount = string | number;
+
+export interface StopRecordingRequest {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * Uniquely identifies the resource.
+   */
+  recording_id?: string;
+}
+
+export interface TelnyxVoiceSettings {
+  /**
+   * The voice speed to be used for the voice. The voice speed must be between 0.1
+   * and 2.0. Default value is 1.0.
+   */
+  voice_speed?: number;
+}
+
+/**
+ * The settings associated with speech to text for the voice assistant. This is
+ * only relevant if the assistant uses a text-to-text language model. Any assistant
+ * using a model with native audio support (e.g. `fixie-ai/ultravox-v0_4`) will
+ * ignore this field.
+ */
+export interface TranscriptionConfig {
+  /**
+   * The speech to text model to be used by the voice assistant.
+   *
+   * - `distil-whisper/distil-large-v2` is lower latency but English-only.
+   * - `openai/whisper-large-v3-turbo` is multi-lingual with automatic language
+   *   detection but slightly higher latency.
+   * - `google` is a multi-lingual option, please describe the language in the
+   *   `language` field.
+   */
+  model?: string;
+}
+
+export interface TranscriptionEngineAConfig {
+  /**
+   * Enables speaker diarization.
+   */
+  enable_speaker_diarization?: boolean;
+
+  /**
+   * Hints to improve transcription accuracy.
+   */
+  hints?: Array<string>;
+
+  /**
+   * Whether to send also interim results. If set to false, only final results will
+   * be sent.
+   */
+  interim_results?: boolean;
+
+  /**
+   * Language to use for speech recognition
+   */
+  language?: GoogleTranscriptionLanguage;
+
+  /**
+   * Defines maximum number of speakers in the conversation.
+   */
+  max_speaker_count?: number;
+
+  /**
+   * Defines minimum number of speakers in the conversation.
+   */
+  min_speaker_count?: number;
+
+  /**
+   * The model to use for transcription.
+   */
+  model?:
+    | 'latest_long'
+    | 'latest_short'
+    | 'command_and_search'
+    | 'phone_call'
+    | 'video'
+    | 'default'
+    | 'medical_conversation'
+    | 'medical_dictation';
+
+  /**
+   * Enables profanity_filter.
+   */
+  profanity_filter?: boolean;
+
+  /**
+   * Speech context to improve transcription accuracy.
+   */
+  speech_context?: Array<TranscriptionEngineAConfig.SpeechContext>;
+
+  /**
+   * Engine identifier for Google transcription service
+   */
+  transcription_engine?: 'A';
+
+  /**
+   * Enables enhanced transcription, this works for models `phone_call` and `video`.
+   */
+  use_enhanced?: boolean;
+}
+
+export namespace TranscriptionEngineAConfig {
+  export interface SpeechContext {
+    /**
+     * Boost factor for the speech context.
+     */
+    boost?: number;
+
+    phrases?: Array<string>;
+  }
+}
+
+export interface TranscriptionEngineBConfig {
+  /**
+   * Language to use for speech recognition
+   */
+  language?:
+    | 'en'
+    | 'zh'
+    | 'de'
+    | 'es'
+    | 'ru'
+    | 'ko'
+    | 'fr'
+    | 'ja'
+    | 'pt'
+    | 'tr'
+    | 'pl'
+    | 'ca'
+    | 'nl'
+    | 'ar'
+    | 'sv'
+    | 'it'
+    | 'id'
+    | 'hi'
+    | 'fi'
+    | 'vi'
+    | 'he'
+    | 'uk'
+    | 'el'
+    | 'ms'
+    | 'cs'
+    | 'ro'
+    | 'da'
+    | 'hu'
+    | 'ta'
+    | 'no'
+    | 'th'
+    | 'ur'
+    | 'hr'
+    | 'bg'
+    | 'lt'
+    | 'la'
+    | 'mi'
+    | 'ml'
+    | 'cy'
+    | 'sk'
+    | 'te'
+    | 'fa'
+    | 'lv'
+    | 'bn'
+    | 'sr'
+    | 'az'
+    | 'sl'
+    | 'kn'
+    | 'et'
+    | 'mk'
+    | 'br'
+    | 'eu'
+    | 'is'
+    | 'hy'
+    | 'ne'
+    | 'mn'
+    | 'bs'
+    | 'kk'
+    | 'sq'
+    | 'sw'
+    | 'gl'
+    | 'mr'
+    | 'pa'
+    | 'si'
+    | 'km'
+    | 'sn'
+    | 'yo'
+    | 'so'
+    | 'af'
+    | 'oc'
+    | 'ka'
+    | 'be'
+    | 'tg'
+    | 'sd'
+    | 'gu'
+    | 'am'
+    | 'yi'
+    | 'lo'
+    | 'uz'
+    | 'fo'
+    | 'ht'
+    | 'ps'
+    | 'tk'
+    | 'nn'
+    | 'mt'
+    | 'sa'
+    | 'lb'
+    | 'my'
+    | 'bo'
+    | 'tl'
+    | 'mg'
+    | 'as'
+    | 'tt'
+    | 'haw'
+    | 'ln'
+    | 'ha'
+    | 'ba'
+    | 'jw'
+    | 'su'
+    | 'auto_detect';
+
+  /**
+   * Engine identifier for Telnyx transcription service
+   */
+  transcription_engine?: 'B';
+
+  /**
+   * The model to use for transcription.
+   */
+  transcription_model?: 'openai/whisper-tiny' | 'openai/whisper-large-v3-turbo';
+}
+
+export interface TranscriptionStartRequest {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * Engine to use for speech recognition. `A` - `Google`, `B` - `Telnyx`.
+   */
+  transcription_engine?: 'A' | 'B';
+
+  transcription_engine_config?: TranscriptionEngineAConfig | TranscriptionEngineBConfig;
+
+  /**
+   * Indicates which leg of the call will be transcribed. Use `inbound` for the leg
+   * that requested the transcription, `outbound` for the other leg, and `both` for
+   * both legs of the call. Will default to `inbound`.
+   */
+  transcription_tracks?: string;
+}
+
+export interface ActionAnswerResponse {
+  data?: ActionAnswerResponse.Data;
+}
+
+export namespace ActionAnswerResponse {
+  export interface Data {
+    /**
+     * The ID of the recording. Only present when the record parameter is set to
+     * record-from-answer.
+     */
+    recording_id?: string;
+
+    result?: string;
+  }
+}
+
+export interface ActionBridgeResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionEnqueueResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionGatherResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionGatherUsingAIResponse {
+  data?: ActionGatherUsingAIResponse.Data;
+}
+
+export namespace ActionGatherUsingAIResponse {
+  export interface Data {
+    /**
+     * The ID of the conversation created by the command.
+     */
+    conversation_id?: string;
+
+    result?: string;
+  }
+}
+
+export interface ActionGatherUsingAudioResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionGatherUsingSpeakResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionHangupResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionLeaveQueueResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionPauseRecordingResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionReferResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionRejectResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionResumeRecordingResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionSendDtmfResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionSendSipInfoResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionSpeakResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionStartAIAssistantResponse {
+  data?: ActionStartAIAssistantResponse.Data;
+}
+
+export namespace ActionStartAIAssistantResponse {
+  export interface Data {
+    /**
+     * The ID of the conversation created by the command.
+     */
+    conversation_id?: string;
+
+    result?: string;
+  }
+}
+
+export interface ActionStartForkingResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionStartNoiseSuppressionResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionStartPlaybackResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionStartRecordingResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionStartSiprecResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionStartStreamingResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionStartTranscriptionResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionStopAIAssistantResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionStopForkingResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionStopGatherResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionStopNoiseSuppressionResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionStopPlaybackResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionStopRecordingResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionStopSiprecResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionStopStreamingResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionStopTranscriptionResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionSwitchSupervisorRoleResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionTransferResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionUpdateClientStateResponse {
+  data?: CallControlCommandResult;
+}
+
+export interface ActionAnswerParams {
+  /**
+   * Use this field to set the Billing Group ID for the call. Must be a valid and
+   * existing Billing Group ID.
+   */
+  billing_group_id?: string;
+
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * Custom headers to be added to the SIP INVITE response.
+   */
+  custom_headers?: Array<CallsAPI.CustomSipHeader>;
+
+  /**
+   * The list of comma-separated codecs in a preferred order for the forked media to
+   * be received.
+   */
+  preferred_codecs?: 'G722,PCMU,PCMA,G729,OPUS,VP8,H264';
+
+  /**
+   * Start recording automatically after an event. Disabled by default.
+   */
+  record?: 'record-from-answer';
+
+  /**
+   * Defines which channel should be recorded ('single' or 'dual') when `record` is
+   * specified.
+   */
+  record_channels?: 'single' | 'dual';
+
+  /**
+   * The custom recording file name to be used instead of the default `call_leg_id`.
+   * Telnyx will still add a Unix timestamp suffix.
+   */
+  record_custom_file_name?: string;
+
+  /**
+   * Defines the format of the recording ('wav' or 'mp3') when `record` is specified.
+   */
+  record_format?: 'wav' | 'mp3';
+
+  /**
+   * 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_max_length?: number;
+
+  /**
+   * 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_timeout_secs?: number;
+
+  /**
+   * 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_track?: 'both' | 'inbound' | 'outbound';
+
+  /**
+   * When set to `trim-silence`, silence will be removed from the beginning and end
+   * of the recording.
+   */
+  record_trim?: 'trim-silence';
+
+  /**
+   * Generate silence RTP packets when no transmission available.
+   */
+  send_silence_when_idle?: boolean;
+
+  /**
+   * SIP headers to be added to the SIP INVITE response. Currently only User-to-User
+   * header is supported.
+   */
+  sip_headers?: Array<CallsAPI.SipHeader>;
+
+  /**
+   * Use this field to modify sound effects, for example adjust the pitch.
+   */
+  sound_modifications?: CallsAPI.SoundModifications;
+
+  /**
+   * Indicates codec for bidirectional streaming RTP payloads. Used only with
+   * stream_bidirectional_mode=rtp. Case sensitive.
+   */
+  stream_bidirectional_codec?: CallsAPI.StreamBidirectionalCodec;
+
+  /**
+   * Configures method of bidirectional streaming (mp3, rtp).
+   */
+  stream_bidirectional_mode?: CallsAPI.StreamBidirectionalMode;
+
+  /**
+   * Specifies which call legs should receive the bidirectional stream audio.
+   */
+  stream_bidirectional_target_legs?: CallsAPI.StreamBidirectionalTargetLegs;
+
+  /**
+   * Specifies the codec to be used for the streamed audio. When set to 'default' or
+   * when transcoding is not possible, the codec from the call will be used.
+   * Currently, transcoding is only supported between PCMU and PCMA codecs.
+   */
+  stream_codec?: CallsAPI.StreamCodec;
+
+  /**
+   * Specifies which track should be streamed.
+   */
+  stream_track?: 'inbound_track' | 'outbound_track' | 'both_tracks';
+
+  /**
+   * The destination WebSocket address where the stream is going to be delivered.
+   */
+  stream_url?: string;
+
+  /**
+   * Enable transcription upon call answer. The default value is false.
+   */
+  transcription?: boolean;
+
+  transcription_config?: TranscriptionStartRequest;
+
+  /**
+   * Use this field to override the URL for which Telnyx will send subsequent
+   * webhooks to for this call.
+   */
+  webhook_url?: string;
+
+  /**
+   * HTTP request type used for `webhook_url`.
+   */
+  webhook_url_method?: 'POST' | 'GET';
+}
+
+export interface ActionBridgeParams {
+  /**
+   * 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.
+   */
+  body_call_control_id: string;
+
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * When enabled, DTMF tones are not passed to the call participant. The webhooks
+   * containing the DTMF information will be sent.
+   */
+  mute_dtmf?: 'none' | 'both' | 'self' | 'opposite';
+
+  /**
+   * 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.
+   */
+  park_after_unbridge?: string;
+
+  /**
+   * Specifies whether to play a ringtone if the call you want to bridge with has not
+   * yet been answered.
+   */
+  play_ringtone?: boolean;
+
+  /**
+   * 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.
+   */
+  queue?: string;
+
+  /**
+   * Start recording automatically after an event. Disabled by default.
+   */
+  record?: 'record-from-answer';
+
+  /**
+   * Defines which channel should be recorded ('single' or 'dual') when `record` is
+   * specified.
+   */
+  record_channels?: 'single' | 'dual';
+
+  /**
+   * The custom recording file name to be used instead of the default `call_leg_id`.
+   * Telnyx will still add a Unix timestamp suffix.
+   */
+  record_custom_file_name?: string;
+
+  /**
+   * Defines the format of the recording ('wav' or 'mp3') when `record` is specified.
+   */
+  record_format?: 'wav' | 'mp3';
+
+  /**
+   * 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_max_length?: number;
+
+  /**
+   * 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_timeout_secs?: number;
+
+  /**
+   * 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_track?: 'both' | 'inbound' | 'outbound';
+
+  /**
+   * When set to `trim-silence`, silence will be removed from the beginning and end
+   * of the recording.
+   */
+  record_trim?: 'trim-silence';
+
+  /**
+   * Specifies which country ringtone to play when `play_ringtone` is set to `true`.
+   * If not set, the US ringtone will be played.
+   */
+  ringtone?:
+    | 'at'
+    | 'au'
+    | 'be'
+    | 'bg'
+    | 'br'
+    | 'ch'
+    | 'cl'
+    | 'cn'
+    | 'cz'
+    | 'de'
+    | 'dk'
+    | 'ee'
+    | 'es'
+    | 'fi'
+    | 'fr'
+    | 'gr'
+    | 'hu'
+    | 'il'
+    | 'in'
+    | 'it'
+    | 'jp'
+    | 'lt'
+    | 'mx'
+    | 'my'
+    | 'nl'
+    | 'no'
+    | 'nz'
+    | 'ph'
+    | 'pl'
+    | 'pt'
+    | 'ru'
+    | 'se'
+    | 'sg'
+    | 'th'
+    | 'tw'
+    | 'uk'
+    | 'us-old'
+    | 'us'
+    | 've'
+    | 'za';
+
+  /**
+   * 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.
+   */
+  video_room_context?: string;
+
+  /**
+   * The ID of the video room you want to bridge with, can't be used together with
+   * call_control_id parameter or queue parameter.
+   */
+  video_room_id?: string;
+}
+
+export interface ActionEnqueueParams {
+  /**
+   * The name of the queue the call should be put in. If a queue with a given name
+   * doesn't exist yet it will be created.
+   */
+  queue_name: string;
+
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * The maximum number of calls allowed in the queue at a given time. Can't be
+   * modified for an existing queue.
+   */
+  max_size?: number;
+
+  /**
+   * The number of seconds after which the call will be removed from the queue.
+   */
+  max_wait_time_secs?: number;
+}
+
+export interface ActionGatherParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * An id that will be sent back in the corresponding `call.gather.ended` webhook.
+   * Will be randomly generated if not specified.
+   */
+  gather_id?: string;
+
+  /**
+   * The number of milliseconds to wait for the first DTMF.
+   */
+  initial_timeout_millis?: number;
+
+  /**
+   * The number of milliseconds to wait for input between digits.
+   */
+  inter_digit_timeout_millis?: number;
+
+  /**
+   * The maximum number of digits to fetch. This parameter has a maximum value
+   * of 128.
+   */
+  maximum_digits?: number;
+
+  /**
+   * The minimum number of digits to fetch. This parameter has a minimum value of 1.
+   */
+  minimum_digits?: number;
+
+  /**
+   * The digit used to terminate input if fewer than `maximum_digits` digits have
+   * been gathered.
+   */
+  terminating_digit?: string;
+
+  /**
+   * The number of milliseconds to wait to complete the request.
+   */
+  timeout_millis?: number;
+
+  /**
+   * A list of all digits accepted as valid.
+   */
+  valid_digits?: string;
+}
+
+export interface ActionGatherUsingAIParams {
+  /**
+   * The parameters described as a JSON Schema object that needs to be gathered by
+   * the voice assistant. See the
+   * [JSON Schema reference](https://json-schema.org/understanding-json-schema) for
+   * documentation about the format
+   */
+  parameters: unknown;
+
+  /**
+   * Assistant configuration including choice of LLM, custom instructions, and tools.
+   */
+  assistant?: AssistantsAPI.Assistant;
+
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * Text that will be played when the gathering starts, if none then nothing will be
+   * played when the gathering starts. The greeting can be text for any voice or SSML
+   * for `AWS.Polly.<voice_id>` voices. There is a 3,000 character limit.
+   */
+  greeting?: string;
+
+  /**
+   * Settings for handling user interruptions during assistant speech
+   */
+  interruption_settings?: InterruptionSettings;
+
+  /**
+   * Language to use for speech recognition
+   */
+  language?: GoogleTranscriptionLanguage;
+
+  /**
+   * The message history you want the voice assistant to be aware of, this can be
+   * useful to keep the context of the conversation, or to pass additional
+   * information to the voice assistant.
+   */
+  message_history?: Array<ActionGatherUsingAIParams.MessageHistory>;
+
+  /**
+   * Default is `false`. If set to `true`, the voice assistant will send updates to
+   * the message history via the `call.ai_gather.message_history_updated`
+   * [callback](https://developers.telnyx.com/api/call-control/call-gather-using-ai#callbacks)
+   * in real time as the message history is updated.
+   */
+  send_message_history_updates?: boolean;
+
+  /**
+   * Default is `false`. If set to `true`, the voice assistant will send partial
+   * results via the `call.ai_gather.partial_results`
+   * [callback](https://developers.telnyx.com/api/call-control/call-gather-using-ai#callbacks)
+   * in real time as individual fields are gathered. If set to `false`, the voice
+   * assistant will only send the final result via the `call.ai_gather.ended`
+   * callback.
+   */
+  send_partial_results?: boolean;
+
+  /**
+   * The settings associated with speech to text for the voice assistant. This is
+   * only relevant if the assistant uses a text-to-text language model. Any assistant
+   * using a model with native audio support (e.g. `fixie-ai/ultravox-v0_4`) will
+   * ignore this field.
+   */
+  transcription?: TranscriptionConfig;
+
+  /**
+   * The number of milliseconds to wait for a user response before the voice
+   * assistant times out and check if the user is still there.
+   */
+  user_response_timeout_ms?: number;
+
+  /**
+   * The voice to be used by the voice assistant. Currently we support ElevenLabs,
+   * Telnyx and AWS voices.
+   *
+   * **Supported Providers:**
+   *
+   * - **AWS:** Use `AWS.Polly.<VoiceId>` (e.g., `AWS.Polly.Joanna`). For neural
+   *   voices, which provide more realistic, human-like speech, append `-Neural` to
+   *   the `VoiceId` (e.g., `AWS.Polly.Joanna-Neural`). Check the
+   *   [available voices](https://docs.aws.amazon.com/polly/latest/dg/available-voices.html)
+   *   for compatibility.
+   * - **Azure:** Use `Azure.<VoiceId>. (e.g. Azure.en-CA-ClaraNeural,
+   *   Azure.en-CA-LiamNeural, Azure.en-US-BrianMultilingualNeural,
+   *   Azure.en-US-Ava:DragonHDLatestNeural. For a complete list of voices, go to
+   *   [Azure Voice Gallery](https://speech.microsoft.com/portal/voicegallery).)
+   * - **ElevenLabs:** Use `ElevenLabs.<ModelId>.<VoiceId>` (e.g.,
+   *   `ElevenLabs.BaseModel.John`). The `ModelId` part is optional. To use
+   *   ElevenLabs, you must provide your ElevenLabs API key as an integration secret
+   *   under `"voice_settings": {"api_key_ref": "<secret_id>"}`. See
+   *   [integration secrets documentation](https://developers.telnyx.com/api/secrets-manager/integration-secrets/create-integration-secret)
+   *   for details. Check
+   *   [available voices](https://elevenlabs.io/docs/api-reference/get-voices).
+   * - **Telnyx:** Use `Telnyx.<model_id>.<voice_id>`
+   */
+  voice?: string;
+
+  /**
+   * The settings associated with the voice selected
+   */
+  voice_settings?: ElevenLabsVoiceSettings | TelnyxVoiceSettings | AwsVoiceSettings;
+}
+
+export namespace ActionGatherUsingAIParams {
+  export interface MessageHistory {
+    /**
+     * The content of the message
+     */
+    content?: string;
+
+    /**
+     * The role of the message sender
+     */
+    role?: 'assistant' | 'user';
+  }
+}
+
+export interface ActionGatherUsingAudioParams {
+  /**
+   * 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.
+   */
+  audio_url?: string;
+
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * The number of milliseconds to wait for input between digits.
+   */
+  inter_digit_timeout_millis?: number;
+
+  /**
+   * 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.
+   */
+  invalid_audio_url?: string;
+
+  /**
+   * 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.
+   */
+  invalid_media_name?: string;
+
+  /**
+   * The maximum number of digits to fetch. This parameter has a maximum value
+   * of 128.
+   */
+  maximum_digits?: number;
+
+  /**
+   * The maximum number of times the file should be played if there is no input from
+   * the user on the call.
+   */
+  maximum_tries?: number;
+
+  /**
+   * 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.
+   */
+  media_name?: string;
+
+  /**
+   * The minimum number of digits to fetch. This parameter has a minimum value of 1.
+   */
+  minimum_digits?: number;
+
+  /**
+   * The digit used to terminate input if fewer than `maximum_digits` digits have
+   * been gathered.
+   */
+  terminating_digit?: string;
+
+  /**
+   * The number of milliseconds to wait for a DTMF response after file playback ends
+   * before a replaying the sound file.
+   */
+  timeout_millis?: number;
+
+  /**
+   * A list of all digits accepted as valid.
+   */
+  valid_digits?: string;
+}
+
+export interface ActionGatherUsingSpeakParams {
+  /**
+   * The text or SSML to be converted into speech. There is a 3,000 character limit.
+   */
+  payload: string;
+
+  /**
+   * Specifies the voice used in speech synthesis.
+   *
+   * - Define voices using the format `<Provider>.<Model>.<VoiceId>`. Specifying only
+   *   the provider will give default values for voice_id and model_id.
+   *
+   *   **Supported Providers:**
+   *
+   * - **AWS:** Use `AWS.Polly.<VoiceId>` (e.g., `AWS.Polly.Joanna`). For neural
+   *   voices, which provide more realistic, human-like speech, append `-Neural` to
+   *   the `VoiceId` (e.g., `AWS.Polly.Joanna-Neural`). Check the
+   *   [available voices](https://docs.aws.amazon.com/polly/latest/dg/available-voices.html)
+   *   for compatibility.
+   * - **Azure:** Use `Azure.<VoiceId>. (e.g. Azure.en-CA-ClaraNeural,
+   *   Azure.en-CA-LiamNeural, Azure.en-US-BrianMultilingualNeural,
+   *   Azure.en-US-Ava:DragonHDLatestNeural. For a complete list of voices, go to
+   *   [Azure Voice Gallery](https://speech.microsoft.com/portal/voicegallery).)
+   * - **ElevenLabs:** Use `ElevenLabs.<ModelId>.<VoiceId>` (e.g.,
+   *   `ElevenLabs.eleven_multilingual_v2.21m00Tcm4TlvDq8ikWAM`). The `ModelId` part
+   *   is optional. To use ElevenLabs, you must provide your ElevenLabs API key as an
+   *   integration identifier secret in
+   *   `"voice_settings": {"api_key_ref": "<secret_identifier>"}`. See
+   *   [integration secrets documentation](https://developers.telnyx.com/api/secrets-manager/integration-secrets/create-integration-secret)
+   *   for details. Check
+   *   [available voices](https://elevenlabs.io/docs/api-reference/get-voices).
+   * - **Telnyx:** Use `Telnyx.<model_id>.<voice_id>`
+   *
+   * For service_level basic, you may define the gender of the speaker (male or
+   * female).
+   */
+  voice: string;
+
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * The number of milliseconds to wait for input between digits.
+   */
+  inter_digit_timeout_millis?: number;
+
+  /**
+   * 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.
+   */
+  invalid_payload?: string;
+
+  /**
+   * The language you want spoken. This parameter is ignored when a `Polly.*` voice
+   * is specified.
+   */
+  language?:
+    | 'arb'
+    | 'cmn-CN'
+    | 'cy-GB'
+    | 'da-DK'
+    | 'de-DE'
+    | 'en-AU'
+    | 'en-GB'
+    | 'en-GB-WLS'
+    | 'en-IN'
+    | 'en-US'
+    | 'es-ES'
+    | 'es-MX'
+    | 'es-US'
+    | 'fr-CA'
+    | 'fr-FR'
+    | 'hi-IN'
+    | 'is-IS'
+    | 'it-IT'
+    | 'ja-JP'
+    | 'ko-KR'
+    | 'nb-NO'
+    | 'nl-NL'
+    | 'pl-PL'
+    | 'pt-BR'
+    | 'pt-PT'
+    | 'ro-RO'
+    | 'ru-RU'
+    | 'sv-SE'
+    | 'tr-TR';
+
+  /**
+   * The maximum number of digits to fetch. This parameter has a maximum value
+   * of 128.
+   */
+  maximum_digits?: number;
+
+  /**
+   * The maximum number of times that a file should be played back if there is no
+   * input from the user on the call.
+   */
+  maximum_tries?: number;
+
+  /**
+   * The minimum number of digits to fetch. This parameter has a minimum value of 1.
+   */
+  minimum_digits?: number;
+
+  /**
+   * The type of the provided payload. The payload can either be plain text, or
+   * Speech Synthesis Markup Language (SSML).
+   */
+  payload_type?: 'text' | '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.
+   */
+  service_level?: 'basic' | 'premium';
+
+  /**
+   * The digit used to terminate input if fewer than `maximum_digits` digits have
+   * been gathered.
+   */
+  terminating_digit?: string;
+
+  /**
+   * The number of milliseconds to wait for a DTMF response after speak ends before a
+   * replaying the sound file.
+   */
+  timeout_millis?: number;
+
+  /**
+   * A list of all digits accepted as valid.
+   */
+  valid_digits?: string;
+
+  /**
+   * The settings associated with the voice selected
+   */
+  voice_settings?: ElevenLabsVoiceSettings | TelnyxVoiceSettings | AwsVoiceSettings;
+}
+
+export interface ActionHangupParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+}
+
+export interface ActionLeaveQueueParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+}
+
+export interface ActionPauseRecordingParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * Uniquely identifies the resource.
+   */
+  recording_id?: string;
+}
+
+export interface ActionReferParams {
+  /**
+   * The SIP URI to which the call will be referred to.
+   */
+  sip_address: string;
+
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid execution of duplicate commands. Telnyx will ignore
+   * subsequent commands with the same `command_id` as one that has already been
+   * executed.
+   */
+  command_id?: string;
+
+  /**
+   * Custom headers to be added to the SIP INVITE.
+   */
+  custom_headers?: Array<CallsAPI.CustomSipHeader>;
+
+  /**
+   * SIP Authentication password used for SIP challenges.
+   */
+  sip_auth_password?: string;
+
+  /**
+   * SIP Authentication username used for SIP challenges.
+   */
+  sip_auth_username?: string;
+
+  /**
+   * SIP headers to be added to the request. Currently only User-to-User header is
+   * supported.
+   */
+  sip_headers?: Array<CallsAPI.SipHeader>;
+}
+
+export interface ActionRejectParams {
+  /**
+   * Cause for call rejection.
+   */
+  cause: 'CALL_REJECTED' | 'USER_BUSY';
+
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+}
+
+export interface ActionResumeRecordingParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * Uniquely identifies the resource.
+   */
+  recording_id?: string;
+}
+
+export interface ActionSendDtmfParams {
+  /**
+   * DTMF digits to send. Valid digits are 0-9, A-D, \*, and #. Pauses can be added
+   * using w (0.5s) and W (1s).
+   */
+  digits: string;
+
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * Specifies for how many milliseconds each digit will be played in the audio
+   * stream. Ranges from 100 to 500ms
+   */
+  duration_millis?: number;
+}
+
+export interface ActionSendSipInfoParams {
+  /**
+   * Content of the SIP INFO
+   */
+  body: string;
+
+  /**
+   * Content type of the INFO body. Must be MIME type compliant. There is a 1,400
+   * bytes limit
+   */
+  content_type: string;
+
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+}
+
+export interface ActionSpeakParams {
+  /**
+   * The text or SSML to be converted into speech. There is a 3,000 character limit.
+   */
+  payload: string;
+
+  /**
+   * Specifies the voice used in speech synthesis.
+   *
+   * - Define voices using the format `<Provider>.<Model>.<VoiceId>`. Specifying only
+   *   the provider will give default values for voice_id and model_id.
+   *
+   *   **Supported Providers:**
+   *
+   * - **AWS:** Use `AWS.Polly.<VoiceId>` (e.g., `AWS.Polly.Joanna`). For neural
+   *   voices, which provide more realistic, human-like speech, append `-Neural` to
+   *   the `VoiceId` (e.g., `AWS.Polly.Joanna-Neural`). Check the
+   *   [available voices](https://docs.aws.amazon.com/polly/latest/dg/available-voices.html)
+   *   for compatibility.
+   * - **Azure:** Use `Azure.<VoiceId>. (e.g. Azure.en-CA-ClaraNeural,
+   *   Azure.en-CA-LiamNeural, Azure.en-US-BrianMultilingualNeural,
+   *   Azure.en-US-Ava:DragonHDLatestNeural. For a complete list of voices, go to
+   *   [Azure Voice Gallery](https://speech.microsoft.com/portal/voicegallery).)
+   * - **ElevenLabs:** Use `ElevenLabs.<ModelId>.<VoiceId>` (e.g.,
+   *   `ElevenLabs.eleven_multilingual_v2.21m00Tcm4TlvDq8ikWAM`). The `ModelId` part
+   *   is optional. To use ElevenLabs, you must provide your ElevenLabs API key as an
+   *   integration identifier secret in
+   *   `"voice_settings": {"api_key_ref": "<secret_identifier>"}`. See
+   *   [integration secrets documentation](https://developers.telnyx.com/api/secrets-manager/integration-secrets/create-integration-secret)
+   *   for details. Check
+   *   [available voices](https://elevenlabs.io/docs/api-reference/get-voices).
+   * - **Telnyx:** Use `Telnyx.<model_id>.<voice_id>`
+   *
+   * For service_level basic, you may define the gender of the speaker (male or
+   * female).
+   */
+  voice: string;
+
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * The language you want spoken. This parameter is ignored when a `Polly.*` voice
+   * is specified.
+   */
+  language?:
+    | 'arb'
+    | 'cmn-CN'
+    | 'cy-GB'
+    | 'da-DK'
+    | 'de-DE'
+    | 'en-AU'
+    | 'en-GB'
+    | 'en-GB-WLS'
+    | 'en-IN'
+    | 'en-US'
+    | 'es-ES'
+    | 'es-MX'
+    | 'es-US'
+    | 'fr-CA'
+    | 'fr-FR'
+    | 'hi-IN'
+    | 'is-IS'
+    | 'it-IT'
+    | 'ja-JP'
+    | 'ko-KR'
+    | 'nb-NO'
+    | 'nl-NL'
+    | 'pl-PL'
+    | 'pt-BR'
+    | 'pt-PT'
+    | 'ro-RO'
+    | 'ru-RU'
+    | 'sv-SE'
+    | 'tr-TR';
+
+  /**
+   * The type of the provided payload. The payload can either be plain text, or
+   * Speech Synthesis Markup Language (SSML).
+   */
+  payload_type?: 'text' | '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.
+   */
+  service_level?: 'basic' | 'premium';
+
+  /**
+   * When specified, it stops the current audio being played. Specify `current` to
+   * stop the current audio being played, and to play the next file in the queue.
+   * Specify `all` to stop the current audio file being played and to also clear all
+   * audio files from the queue.
+   */
+  stop?: string;
+
+  /**
+   * The settings associated with the voice selected
+   */
+  voice_settings?: ElevenLabsVoiceSettings | TelnyxVoiceSettings | AwsVoiceSettings;
+}
+
+export interface ActionStartAIAssistantParams {
+  /**
+   * AI Assistant configuration
+   */
+  assistant?: ActionStartAIAssistantParams.Assistant;
+
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * Text that will be played when the assistant starts, if none then nothing will be
+   * played when the assistant starts. The greeting can be text for any voice or SSML
+   * for `AWS.Polly.<voice_id>` voices. There is a 3,000 character limit.
+   */
+  greeting?: string;
+
+  /**
+   * Settings for handling user interruptions during assistant speech
+   */
+  interruption_settings?: InterruptionSettings;
+
+  /**
+   * The settings associated with speech to text for the voice assistant. This is
+   * only relevant if the assistant uses a text-to-text language model. Any assistant
+   * using a model with native audio support (e.g. `fixie-ai/ultravox-v0_4`) will
+   * ignore this field.
+   */
+  transcription?: TranscriptionConfig;
+
+  /**
+   * The voice to be used by the voice assistant. Currently we support ElevenLabs,
+   * Telnyx and AWS voices.
+   *
+   * **Supported Providers:**
+   *
+   * - **AWS:** Use `AWS.Polly.<VoiceId>` (e.g., `AWS.Polly.Joanna`). For neural
+   *   voices, which provide more realistic, human-like speech, append `-Neural` to
+   *   the `VoiceId` (e.g., `AWS.Polly.Joanna-Neural`). Check the
+   *   [available voices](https://docs.aws.amazon.com/polly/latest/dg/available-voices.html)
+   *   for compatibility.
+   * - **Azure:** Use `Azure.<VoiceId>. (e.g. Azure.en-CA-ClaraNeural,
+   *   Azure.en-CA-LiamNeural, Azure.en-US-BrianMultilingualNeural,
+   *   Azure.en-US-Ava:DragonHDLatestNeural. For a complete list of voices, go to
+   *   [Azure Voice Gallery](https://speech.microsoft.com/portal/voicegallery).)
+   * - **ElevenLabs:** Use `ElevenLabs.<ModelId>.<VoiceId>` (e.g.,
+   *   `ElevenLabs.BaseModel.John`). The `ModelId` part is optional. To use
+   *   ElevenLabs, you must provide your ElevenLabs API key as an integration secret
+   *   under `"voice_settings": {"api_key_ref": "<secret_id>"}`. See
+   *   [integration secrets documentation](https://developers.telnyx.com/api/secrets-manager/integration-secrets/create-integration-secret)
+   *   for details. Check
+   *   [available voices](https://elevenlabs.io/docs/api-reference/get-voices).
+   * - **Telnyx:** Use `Telnyx.<model_id>.<voice_id>`
+   */
+  voice?: string;
+
+  /**
+   * The settings associated with the voice selected
+   */
+  voice_settings?: ElevenLabsVoiceSettings | TelnyxVoiceSettings | AwsVoiceSettings;
+}
+
+export namespace ActionStartAIAssistantParams {
+  /**
+   * AI Assistant configuration
+   */
+  export interface Assistant {
+    /**
+     * The identifier of the AI assistant to use
+     */
+    id?: string;
+
+    /**
+     * The system instructions that the voice assistant uses during the start assistant
+     * command. This will overwrite the instructions set in the assistant
+     * configuration.
+     */
+    instructions?: string;
+
+    /**
+     * Reference to the OpenAI API key. Required only when using OpenAI models
+     */
+    openai_api_key_ref?: string;
+  }
+}
+
+export interface ActionStartForkingParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * The network target, <udp:ip_address:port>, where the call's incoming RTP media
+   * packets should be forwarded.
+   */
+  rx?: string;
+
+  /**
+   * 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.
+   */
+  stream_type?: 'decrypted';
+
+  /**
+   * The network target, <udp:ip_address:port>, where the call's outgoing RTP media
+   * packets should be forwarded.
+   */
+  tx?: string;
+}
+
+export interface ActionStartNoiseSuppressionParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * The direction of the audio stream to be noise suppressed.
+   */
+  direction?: 'inbound' | 'outbound' | 'both';
+
+  /**
+   * The engine to use for noise suppression. A - rnnoise engine B - deepfilter
+   * engine.
+   */
+  noise_suppression_engine?: 'A' | 'B';
+}
+
+export interface ActionStartPlaybackParams {
+  /**
+   * Specifies the type of audio provided in `audio_url` or `playback_content`.
+   */
+  audio_type?: 'mp3' | 'wav';
+
+  /**
+   * The URL of a file to be played back on 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.
+   */
+  audio_url?: string;
+
+  /**
+   * Caches the audio file. Useful when playing the same audio file multiple times
+   * during the call.
+   */
+  cache_audio?: boolean;
+
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * The number of times the audio file should be played. If supplied, the value must
+   * be an integer between 1 and 100, or the special string `infinity` for an endless
+   * loop.
+   */
+  loop?: Loopcount;
+
+  /**
+   * The media_name of a file to be played back on 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.
+   */
+  media_name?: string;
+
+  /**
+   * When enabled, audio will be mixed on top of any other audio that is actively
+   * being played back. Note that `overlay: true` will only work if there is another
+   * audio file already being played on the call.
+   */
+  overlay?: boolean;
+
+  /**
+   * Allows a user to provide base64 encoded mp3 or wav. Note: when using this
+   * parameter, `media_url` and `media_name` in the `playback_started` and
+   * `playback_ended` webhooks will be empty
+   */
+  playback_content?: string;
+
+  /**
+   * When specified, it stops the current audio being played. Specify `current` to
+   * stop the current audio being played, and to play the next file in the queue.
+   * Specify `all` to stop the current audio file being played and to also clear all
+   * audio files from the queue.
+   */
+  stop?: string;
+
+  /**
+   * Specifies the leg or legs on which audio will be played. If supplied, the value
+   * must be either `self`, `opposite` or `both`.
+   */
+  target_legs?: string;
+}
+
+export interface ActionStartRecordingParams {
+  /**
+   * When `dual`, final audio file will be stereo recorded with the first leg on
+   * channel A, and the rest on channel B.
+   */
+  channels: 'single' | 'dual';
+
+  /**
+   * The audio file format used when storing the call recording. Can be either `mp3`
+   * or `wav`.
+   */
+  format: 'wav' | 'mp3';
+
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * The custom recording file name to be used instead of the default `call_leg_id`.
+   * Telnyx will still add a Unix timestamp suffix.
+   */
+  custom_file_name?: string;
+
+  /**
+   * Defines the maximum length for the recording in seconds. The minimum value is 0.
+   * The maximum value is 14400. The default value is 0 (infinite)
+   */
+  max_length?: number;
+
+  /**
+   * If enabled, a beep sound will be played at the start of a recording.
+   */
+  play_beep?: boolean;
+
+  /**
+   * 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).
+   */
+  recording_track?: 'both' | 'inbound' | 'outbound';
+
+  /**
+   * The number of seconds that Telnyx will wait for the recording to be stopped if
+   * silence is detected. 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)
+   */
+  timeout_secs?: number;
+
+  /**
+   * Enable post recording transcription. The default value is false.
+   */
+  transcription?: boolean;
+
+  /**
+   * Engine to use for speech recognition. `A` - `Google`
+   */
+  transcription_engine?: string;
+
+  /**
+   * Language to use for speech recognition
+   */
+  transcription_language?:
+    | 'af-ZA'
+    | 'am-ET'
+    | 'ar-AE'
+    | 'ar-BH'
+    | 'ar-DZ'
+    | 'ar-EG'
+    | 'ar-IL'
+    | 'ar-IQ'
+    | 'ar-JO'
+    | 'ar-KW'
+    | 'ar-LB'
+    | 'ar-MA'
+    | 'ar-MR'
+    | 'ar-OM'
+    | 'ar-PS'
+    | 'ar-QA'
+    | 'ar-SA'
+    | 'ar-TN'
+    | 'ar-YE'
+    | 'az-AZ'
+    | 'bg-BG'
+    | 'bn-BD'
+    | 'bn-IN'
+    | 'bs-BA'
+    | 'ca-ES'
+    | 'cs-CZ'
+    | 'da-DK'
+    | 'de-AT'
+    | 'de-CH'
+    | 'de-DE'
+    | 'el-GR'
+    | 'en-AU'
+    | 'en-CA'
+    | 'en-GB'
+    | 'en-GH'
+    | 'en-HK'
+    | 'en-IE'
+    | 'en-IN'
+    | 'en-KE'
+    | 'en-NG'
+    | 'en-NZ'
+    | 'en-PH'
+    | 'en-PK'
+    | 'en-SG'
+    | 'en-TZ'
+    | 'en-US'
+    | 'en-ZA'
+    | 'es-AR'
+    | 'es-BO'
+    | 'es-CL'
+    | 'es-CO'
+    | 'es-CR'
+    | 'es-DO'
+    | 'es-EC'
+    | 'es-ES'
+    | 'es-GT'
+    | 'es-HN'
+    | 'es-MX'
+    | 'es-NI'
+    | 'es-PA'
+    | 'es-PE'
+    | 'es-PR'
+    | 'es-PY'
+    | 'es-SV'
+    | 'es-US'
+    | 'es-UY'
+    | 'es-VE'
+    | 'et-EE'
+    | 'eu-ES'
+    | 'fa-IR'
+    | 'fi-FI'
+    | 'fil-PH'
+    | 'fr-BE'
+    | 'fr-CA'
+    | 'fr-CH'
+    | 'fr-FR'
+    | 'gl-ES'
+    | 'gu-IN'
+    | 'hi-IN'
+    | 'hr-HR'
+    | 'hu-HU'
+    | 'hy-AM'
+    | 'id-ID'
+    | 'is-IS'
+    | 'it-CH'
+    | 'it-IT'
+    | 'iw-IL'
+    | 'ja-JP'
+    | 'jv-ID'
+    | 'ka-GE'
+    | 'kk-KZ'
+    | 'km-KH'
+    | 'kn-IN'
+    | 'ko-KR'
+    | 'lo-LA'
+    | 'lt-LT'
+    | 'lv-LV'
+    | 'mk-MK'
+    | 'ml-IN'
+    | 'mn-MN'
+    | 'mr-IN'
+    | 'ms-MY'
+    | 'my-MM'
+    | 'ne-NP'
+    | 'nl-BE'
+    | 'nl-NL'
+    | 'no-NO'
+    | 'pa-Guru-IN'
+    | 'pl-PL'
+    | 'pt-BR'
+    | 'pt-PT'
+    | 'ro-RO'
+    | 'ru-RU'
+    | 'rw-RW'
+    | 'si-LK'
+    | 'sk-SK'
+    | 'sl-SI'
+    | 'sq-AL'
+    | 'sr-RS'
+    | 'ss-latn-za'
+    | 'st-ZA'
+    | 'su-ID'
+    | 'sv-SE'
+    | 'sw-KE'
+    | 'sw-TZ'
+    | 'ta-IN'
+    | 'ta-LK'
+    | 'ta-MY'
+    | 'ta-SG'
+    | 'te-IN'
+    | 'th-TH'
+    | 'tn-latn-za'
+    | 'tr-TR'
+    | 'ts-ZA'
+    | 'uk-UA'
+    | 'ur-IN'
+    | 'ur-PK'
+    | 'uz-UZ'
+    | 've-ZA'
+    | 'vi-VN'
+    | 'xh-ZA'
+    | 'yue-Hant-HK'
+    | 'zh'
+    | 'zh-TW'
+    | 'zu-ZA';
+
+  /**
+   * Defines maximum number of speakers in the conversation. Applies to `google`
+   * engine only.
+   */
+  transcription_max_speaker_count?: number;
+
+  /**
+   * Defines minimum number of speakers in the conversation. Applies to `google`
+   * engine only.
+   */
+  transcription_min_speaker_count?: number;
+
+  /**
+   * Enables profanity_filter. Applies to `google` engine only.
+   */
+  transcription_profanity_filter?: boolean;
+
+  /**
+   * Enables speaker diarization. Applies to `google` engine only.
+   */
+  transcription_speaker_diarization?: boolean;
+
+  /**
+   * When set to `trim-silence`, silence will be removed from the beginning and end
+   * of the recording.
+   */
+  trim?: 'trim-silence';
+}
+
+export interface ActionStartSiprecParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Name of configured SIPREC connector to be used.
+   */
+  connector_name?: string;
+
+  /**
+   * When set, custom parameters will be added as metadata
+   * (recording.session.ExtensionParameters). Otherwise, they’ll be added to sip
+   * headers.
+   */
+  include_metadata_custom_headers?: true | false;
+
+  /**
+   * Controls whether to encrypt media sent to your SRS using SRTP and TLS. When set
+   * you need to configure SRS port in your connector to 5061.
+   */
+  secure?: true | false;
+
+  /**
+   * Sets `Session-Expires` header to the INVITE. A reinvite is sent every half the
+   * value set. Usefull for session keep alive. Minimum value is 90, set to 0 to
+   * disable.
+   */
+  session_timeout_secs?: number;
+
+  /**
+   * Specifies SIP transport protocol.
+   */
+  sip_transport?: 'udp' | 'tcp' | 'tls';
+
+  /**
+   * Specifies which track should be sent on siprec session.
+   */
+  siprec_track?: 'inbound_track' | 'outbound_track' | 'both_tracks';
+}
+
+export interface ActionStartStreamingParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  dialogflow_config?: CallsAPI.DialogflowConfig;
+
+  /**
+   * Enables Dialogflow for the current call. The default value is false.
+   */
+  enable_dialogflow?: boolean;
+
+  /**
+   * Indicates codec for bidirectional streaming RTP payloads. Used only with
+   * stream_bidirectional_mode=rtp. Case sensitive.
+   */
+  stream_bidirectional_codec?: CallsAPI.StreamBidirectionalCodec;
+
+  /**
+   * Configures method of bidirectional streaming (mp3, rtp).
+   */
+  stream_bidirectional_mode?: CallsAPI.StreamBidirectionalMode;
+
+  /**
+   * Specifies which call legs should receive the bidirectional stream audio.
+   */
+  stream_bidirectional_target_legs?: CallsAPI.StreamBidirectionalTargetLegs;
+
+  /**
+   * Specifies the codec to be used for the streamed audio. When set to 'default' or
+   * when transcoding is not possible, the codec from the call will be used.
+   * Currently, transcoding is only supported between PCMU and PCMA codecs.
+   */
+  stream_codec?: CallsAPI.StreamCodec;
+
+  /**
+   * Specifies which track should be streamed.
+   */
+  stream_track?: 'inbound_track' | 'outbound_track' | 'both_tracks';
+
+  /**
+   * The destination WebSocket address where the stream is going to be delivered.
+   */
+  stream_url?: string;
+}
+
+export interface ActionStartTranscriptionParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * Engine to use for speech recognition. `A` - `Google`, `B` - `Telnyx`.
+   */
+  transcription_engine?: 'A' | 'B';
+
+  transcription_engine_config?: TranscriptionEngineAConfig | TranscriptionEngineBConfig;
+
+  /**
+   * Indicates which leg of the call will be transcribed. Use `inbound` for the leg
+   * that requested the transcription, `outbound` for the other leg, and `both` for
+   * both legs of the call. Will default to `inbound`.
+   */
+  transcription_tracks?: string;
+}
+
+export interface ActionStopAIAssistantParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+}
+
+export interface ActionStopForkingParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * Optionally specify a `stream_type`. This should match the `stream_type` that was
+   * used in `fork_start` command to properly stop the fork.
+   */
+  stream_type?: 'raw' | 'decrypted';
+}
+
+export interface ActionStopGatherParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+}
+
+export interface ActionStopNoiseSuppressionParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+}
+
+export interface ActionStopPlaybackParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * When enabled, it stops the audio being played in the overlay queue.
+   */
+  overlay?: boolean;
+
+  /**
+   * Use `current` to stop the current audio being played. Use `all` to stop the
+   * current audio file being played and clear all audio files from the queue.
+   */
+  stop?: string;
+}
+
+export interface ActionStopRecordingParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * Uniquely identifies the resource.
+   */
+  recording_id?: string;
+}
+
+export interface ActionStopSiprecParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+}
+
+export interface ActionStopStreamingParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * Identifies the stream. If the `stream_id` is not provided the command stops all
+   * streams associated with a given `call_control_id`.
+   */
+  stream_id?: string;
+}
+
+export interface ActionStopTranscriptionParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+}
+
+export interface ActionSwitchSupervisorRoleParams {
+  /**
+   * The supervisor role to switch to. 'barge' allows speaking to both parties,
+   * 'whisper' allows speaking to caller only, 'monitor' allows listening only.
+   */
+  role: 'barge' | 'whisper' | 'monitor';
+}
+
+export interface ActionTransferParams {
+  /**
+   * The DID or SIP URI to dial out to.
+   */
+  to: string;
+
+  /**
+   * 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.
+   */
+  answering_machine_detection?:
+    | 'premium'
+    | 'detect'
+    | 'detect_beep'
+    | 'detect_words'
+    | 'greeting_end'
+    | 'disabled';
+
+  /**
+   * Optional configuration parameters to modify 'answering_machine_detection'
+   * performance.
+   */
+  answering_machine_detection_config?: ActionTransferParams.AnsweringMachineDetectionConfig;
+
+  /**
+   * 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.
+   */
+  audio_url?: string;
+
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state?: string;
+
+  /**
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
+   */
+  command_id?: string;
+
+  /**
+   * Custom headers to be added to the SIP INVITE.
+   */
+  custom_headers?: Array<CallsAPI.CustomSipHeader>;
+
+  /**
+   * If set to false, early media will not be passed to the originating leg.
+   */
+  early_media?: boolean;
+
+  /**
+   * 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.
+   */
+  from?: 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.
+   */
+  from_display_name?: string;
+
+  /**
+   * Defines whether media should be encrypted on the new call leg.
+   */
+  media_encryption?: 'disabled' | 'SRTP' | 'DTLS';
+
+  /**
+   * 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.
+   */
+  media_name?: string;
+
+  /**
+   * When enabled, DTMF tones are not passed to the call participant. The webhooks
+   * containing the DTMF information will be sent.
+   */
+  mute_dtmf?: 'none' | 'both' | 'self' | 'opposite';
+
+  /**
+   * 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.
+   */
+  park_after_unbridge?: string;
+
+  /**
+   * SIP Authentication password used for SIP challenges.
+   */
+  sip_auth_password?: string;
+
+  /**
+   * SIP Authentication username used for SIP challenges.
+   */
+  sip_auth_username?: string;
+
+  /**
+   * SIP headers to be added to the SIP INVITE. Currently only User-to-User header is
+   * supported.
+   */
+  sip_headers?: Array<CallsAPI.SipHeader>;
+
+  /**
+   * Defines SIP transport protocol to be used on the call.
+   */
+  sip_transport_protocol?: 'UDP' | 'TCP' | 'TLS';
+
+  /**
+   * Use this field to modify sound effects, for example adjust the pitch.
+   */
+  sound_modifications?: CallsAPI.SoundModifications;
+
+  /**
+   * Use this field to add state to every subsequent webhook for the new leg. It must
+   * be a valid Base-64 encoded string.
+   */
+  target_leg_client_state?: string;
+
+  /**
+   * 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.
+   */
+  time_limit_secs?: number;
+
+  /**
+   * 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.
+   */
+  timeout_secs?: number;
+
+  /**
+   * Use this field to override the URL for which Telnyx will send subsequent
+   * webhooks to for this call.
+   */
+  webhook_url?: string;
+
+  /**
+   * HTTP request type used for `webhook_url`.
+   */
+  webhook_url_method?: 'POST' | 'GET';
+}
+
+export namespace ActionTransferParams {
+  /**
+   * Optional configuration parameters to modify 'answering_machine_detection'
+   * performance.
+   */
+  export interface AnsweringMachineDetectionConfig {
+    /**
+     * Silence duration threshold after a greeting message or voice for it be
+     * considered human.
+     */
+    after_greeting_silence_millis?: number;
+
+    /**
+     * Maximum threshold for silence between words.
+     */
+    between_words_silence_millis?: number;
+
+    /**
+     * Maximum threshold of a human greeting. If greeting longer than this value,
+     * considered machine.
+     */
+    greeting_duration_millis?: number;
+
+    /**
+     * If machine already detected, maximum threshold for silence between words. If
+     * exceeded, the greeting is considered ended.
+     */
+    greeting_silence_duration_millis?: number;
+
+    /**
+     * If machine already detected, maximum timeout threshold to determine the end of
+     * the machine greeting.
+     */
+    greeting_total_analysis_time_millis?: number;
+
+    /**
+     * If initial silence duration is greater than this value, consider it a machine.
+     */
+    initial_silence_millis?: number;
+
+    /**
+     * If number of detected words is greater than this value, consder it a machine.
+     */
+    maximum_number_of_words?: number;
+
+    /**
+     * If a single word lasts longer than this threshold, consider it a machine.
+     */
+    maximum_word_length_millis?: number;
+
+    /**
+     * Minimum noise threshold for any analysis.
+     */
+    silence_threshold?: number;
+
+    /**
+     * Maximum timeout threshold for overall detection.
+     */
+    total_analysis_time_millis?: number;
+  }
+}
+
+export interface ActionUpdateClientStateParams {
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
+   */
+  client_state: string;
+}
+
+export declare namespace Actions {
+  export {
+    type AwsVoiceSettings as AwsVoiceSettings,
+    type CallControlCommandResult as CallControlCommandResult,
+    type ElevenLabsVoiceSettings as ElevenLabsVoiceSettings,
+    type GoogleTranscriptionLanguage as GoogleTranscriptionLanguage,
+    type InterruptionSettings as InterruptionSettings,
+    type Loopcount as Loopcount,
+    type StopRecordingRequest as StopRecordingRequest,
+    type TelnyxVoiceSettings as TelnyxVoiceSettings,
+    type TranscriptionConfig as TranscriptionConfig,
+    type TranscriptionEngineAConfig as TranscriptionEngineAConfig,
+    type TranscriptionEngineBConfig as TranscriptionEngineBConfig,
+    type TranscriptionStartRequest as TranscriptionStartRequest,
+    type ActionAnswerResponse as ActionAnswerResponse,
+    type ActionBridgeResponse as ActionBridgeResponse,
+    type ActionEnqueueResponse as ActionEnqueueResponse,
+    type ActionGatherResponse as ActionGatherResponse,
+    type ActionGatherUsingAIResponse as ActionGatherUsingAIResponse,
+    type ActionGatherUsingAudioResponse as ActionGatherUsingAudioResponse,
+    type ActionGatherUsingSpeakResponse as ActionGatherUsingSpeakResponse,
+    type ActionHangupResponse as ActionHangupResponse,
+    type ActionLeaveQueueResponse as ActionLeaveQueueResponse,
+    type ActionPauseRecordingResponse as ActionPauseRecordingResponse,
+    type ActionReferResponse as ActionReferResponse,
+    type ActionRejectResponse as ActionRejectResponse,
+    type ActionResumeRecordingResponse as ActionResumeRecordingResponse,
+    type ActionSendDtmfResponse as ActionSendDtmfResponse,
+    type ActionSendSipInfoResponse as ActionSendSipInfoResponse,
+    type ActionSpeakResponse as ActionSpeakResponse,
+    type ActionStartAIAssistantResponse as ActionStartAIAssistantResponse,
+    type ActionStartForkingResponse as ActionStartForkingResponse,
+    type ActionStartNoiseSuppressionResponse as ActionStartNoiseSuppressionResponse,
+    type ActionStartPlaybackResponse as ActionStartPlaybackResponse,
+    type ActionStartRecordingResponse as ActionStartRecordingResponse,
+    type ActionStartSiprecResponse as ActionStartSiprecResponse,
+    type ActionStartStreamingResponse as ActionStartStreamingResponse,
+    type ActionStartTranscriptionResponse as ActionStartTranscriptionResponse,
+    type ActionStopAIAssistantResponse as ActionStopAIAssistantResponse,
+    type ActionStopForkingResponse as ActionStopForkingResponse,
+    type ActionStopGatherResponse as ActionStopGatherResponse,
+    type ActionStopNoiseSuppressionResponse as ActionStopNoiseSuppressionResponse,
+    type ActionStopPlaybackResponse as ActionStopPlaybackResponse,
+    type ActionStopRecordingResponse as ActionStopRecordingResponse,
+    type ActionStopSiprecResponse as ActionStopSiprecResponse,
+    type ActionStopStreamingResponse as ActionStopStreamingResponse,
+    type ActionStopTranscriptionResponse as ActionStopTranscriptionResponse,
+    type ActionSwitchSupervisorRoleResponse as ActionSwitchSupervisorRoleResponse,
+    type ActionTransferResponse as ActionTransferResponse,
+    type ActionUpdateClientStateResponse as ActionUpdateClientStateResponse,
+    type ActionAnswerParams as ActionAnswerParams,
+    type ActionBridgeParams as ActionBridgeParams,
+    type ActionEnqueueParams as ActionEnqueueParams,
+    type ActionGatherParams as ActionGatherParams,
+    type ActionGatherUsingAIParams as ActionGatherUsingAIParams,
+    type ActionGatherUsingAudioParams as ActionGatherUsingAudioParams,
+    type ActionGatherUsingSpeakParams as ActionGatherUsingSpeakParams,
+    type ActionHangupParams as ActionHangupParams,
+    type ActionLeaveQueueParams as ActionLeaveQueueParams,
+    type ActionPauseRecordingParams as ActionPauseRecordingParams,
+    type ActionReferParams as ActionReferParams,
+    type ActionRejectParams as ActionRejectParams,
+    type ActionResumeRecordingParams as ActionResumeRecordingParams,
+    type ActionSendDtmfParams as ActionSendDtmfParams,
+    type ActionSendSipInfoParams as ActionSendSipInfoParams,
+    type ActionSpeakParams as ActionSpeakParams,
+    type ActionStartAIAssistantParams as ActionStartAIAssistantParams,
+    type ActionStartForkingParams as ActionStartForkingParams,
+    type ActionStartNoiseSuppressionParams as ActionStartNoiseSuppressionParams,
+    type ActionStartPlaybackParams as ActionStartPlaybackParams,
+    type ActionStartRecordingParams as ActionStartRecordingParams,
+    type ActionStartSiprecParams as ActionStartSiprecParams,
+    type ActionStartStreamingParams as ActionStartStreamingParams,
+    type ActionStartTranscriptionParams as ActionStartTranscriptionParams,
+    type ActionStopAIAssistantParams as ActionStopAIAssistantParams,
+    type ActionStopForkingParams as ActionStopForkingParams,
+    type ActionStopGatherParams as ActionStopGatherParams,
+    type ActionStopNoiseSuppressionParams as ActionStopNoiseSuppressionParams,
+    type ActionStopPlaybackParams as ActionStopPlaybackParams,
+    type ActionStopRecordingParams as ActionStopRecordingParams,
+    type ActionStopSiprecParams as ActionStopSiprecParams,
+    type ActionStopStreamingParams as ActionStopStreamingParams,
+    type ActionStopTranscriptionParams as ActionStopTranscriptionParams,
+    type ActionSwitchSupervisorRoleParams as ActionSwitchSupervisorRoleParams,
+    type ActionTransferParams as ActionTransferParams,
+    type ActionUpdateClientStateParams as ActionUpdateClientStateParams,
+  };
+}