telnyx 7.3.0 → 7.6.0

package/resources/calls/actions.d.mts

@@ -10,17 +10,33 @@ import { RequestOptions } from "../../internal/request-options.mjs";
  */
 export declare class Actions extends APIResource {
     /**
-     * Add messages to the conversation started by an AI assistant on the call.
+     * Start an AI assistant on the call.
+     *
+     * **Expected Webhooks:**
+     *
+     * - `call.conversation.ended`
+     * - `call.conversation_insights.generated`
      *
      * @example
      * ```ts
      * const response =
-     *   await client.calls.actions.addAIAssistantMessages(
+     *   await client.calls.actions.startAIAssistant(
      *     'call_control_id',
      *   );
      * ```
      */
-    addAIAssistantMessages(callControlID: string, body: ActionAddAIAssistantMessagesParams, options?: RequestOptions): APIPromise<ActionAddAIAssistantMessagesResponse>;
+    startAIAssistant(callControlID: string, body: ActionStartAIAssistantParams, options?: RequestOptions): APIPromise<ActionStartAIAssistantResponse>;
+    /**
+     * 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>;
     /**
      * Answer an incoming call. You must issue this command before executing subsequent
      * commands on an incoming call.
@@ -66,6 +82,19 @@ export declare class Actions extends APIResource {
      * ```
      */
     bridge(callControlIDToBridge: string, params: ActionBridgeParams, options?: RequestOptions): APIPromise<ActionBridgeResponse>;
+    /**
+     * 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>;
     /**
      * Put the call in a queue.
      *
@@ -78,6 +107,41 @@ export declare class Actions extends APIResource {
      * ```
      */
     enqueue(callControlID: string, body: ActionEnqueueParams, options?: RequestOptions): APIPromise<ActionEnqueueResponse>;
+    /**
+     * 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:**
+     *
+     * - `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>;
+    /**
+     * Stop forking a call.
+     *
+     * **Expected Webhooks:**
+     *
+     * - `call.fork.stopped`
+     *
+     * @example
+     * ```ts
+     * const response = await client.calls.actions.stopForking(
+     *   'call_control_id',
+     * );
+     * ```
+     */
+    stopForking(callControlID: string, body: ActionStopForkingParams, options?: RequestOptions): APIPromise<ActionStopForkingResponse>;
     /**
      * Gather DTMF signals to build interactive menus.
      *
@@ -97,6 +161,21 @@ export declare class Actions extends APIResource {
      * ```
      */
     gather(callControlID: string, body: ActionGatherParams, options?: RequestOptions): APIPromise<ActionGatherResponse>;
+    /**
+     * Stop current gather.
+     *
+     * **Expected Webhooks:**
+     *
+     * - `call.gather.ended`
+     *
+     * @example
+     * ```ts
+     * const response = await client.calls.actions.stopGather(
+     *   'call_control_id',
+     * );
+     * ```
+     */
+    stopGather(callControlID: string, body: ActionStopGatherParams, options?: RequestOptions): APIPromise<ActionStopGatherResponse>;
     /**
      * Gather parameters defined in the request payload using a voice assistant.
      *
@@ -192,32 +271,54 @@ export declare class Actions extends APIResource {
      */
     hangup(callControlID: string, body: ActionHangupParams, options?: RequestOptions): APIPromise<ActionHangupResponse>;
     /**
-     * Add a participant to an existing AI assistant conversation. Use this command to
-     * bring an additional call leg into a running AI conversation.
+     * Removes the call from a queue.
      *
      * @example
      * ```ts
-     * const response = await client.calls.actions.joinAIAssistant(
+     * const response = await client.calls.actions.leaveQueue(
      *   'call_control_id',
-     *   {
-     *     conversation_id: 'v3:abc123',
-     *     participant: { id: 'v3:abc123def456', role: 'user' },
-     *   },
      * );
      * ```
      */
-    joinAIAssistant(callControlID: string, body: ActionJoinAIAssistantParams, options?: RequestOptions): APIPromise<ActionJoinAIAssistantResponse>;
+    leaveQueue(callControlID: string, body: ActionLeaveQueueParams, options?: RequestOptions): APIPromise<ActionLeaveQueueResponse>;
     /**
-     * Removes the call from a queue.
+     * 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:**
+     *
+     * - `call.playback.started`
+     * - `call.playback.ended`
      *
      * @example
      * ```ts
-     * const response = await client.calls.actions.leaveQueue(
+     * const response = await client.calls.actions.startPlayback(
      *   'call_control_id',
      * );
      * ```
      */
-    leaveQueue(callControlID: string, body: ActionLeaveQueueParams, options?: RequestOptions): APIPromise<ActionLeaveQueueResponse>;
+    startPlayback(callControlID: string, body: ActionStartPlaybackParams, options?: RequestOptions): APIPromise<ActionStartPlaybackResponse>;
+    /**
+     * Stop audio being played on the call.
+     *
+     * **Expected Webhooks:**
+     *
+     * - `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>;
     /**
      * Pause recording the call. Recording can be resumed via Resume recording command.
      *
@@ -233,6 +334,55 @@ export declare class Actions extends APIResource {
      * ```
      */
     pauseRecording(callControlID: string, body: ActionPauseRecordingParams, options?: RequestOptions): APIPromise<ActionPauseRecordingResponse>;
+    /**
+     * 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>;
+    /**
+     * Start recording the call. Recording will stop on call hang-up, or can be
+     * initiated via the Stop Recording command.
+     *
+     * **Expected Webhooks:**
+     *
+     * - `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>;
+    /**
+     * Stop recording the call.
+     *
+     * **Expected Webhooks:**
+     *
+     * - `call.recording.saved`
+     *
+     * @example
+     * ```ts
+     * const response = await client.calls.actions.stopRecording(
+     *   'call_control_id',
+     * );
+     * ```
+     */
+    stopRecording(callControlID: string, body: ActionStopRecordingParams, options?: RequestOptions): APIPromise<ActionStopRecordingResponse>;
     /**
      * Initiate a SIP Refer on a Call Control call. You can initiate a SIP Refer at any
      * point in the duration of a call.
@@ -270,21 +420,6 @@ export declare class Actions extends APIResource {
      * ```
      */
     reject(callControlID: string, body: ActionRejectParams, options?: RequestOptions): APIPromise<ActionRejectResponse>;
-    /**
-     * 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>;
     /**
      * Sends DTMF tones from this leg. DTMF tones will be heard by the other end of the
      * call.
@@ -321,6 +456,38 @@ export declare class Actions extends APIResource {
      * ```
      */
     sendSipInfo(callControlID: string, body: ActionSendSipInfoParams, options?: RequestOptions): APIPromise<ActionSendSipInfoResponse>;
+    /**
+     * 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>;
+    /**
+     * 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>;
     /**
      * 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
@@ -341,64 +508,36 @@ export declare class Actions extends APIResource {
      */
     speak(callControlID: string, body: ActionSpeakParams, options?: RequestOptions): APIPromise<ActionSpeakResponse>;
     /**
-     * Start an AI assistant on the call.
-     *
-     * **Expected Webhooks:**
+     * 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.
      *
-     * - `call.conversation.ended`
-     * - `call.conversation_insights.generated`
+     * 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.startAIAssistant(
-     *     'call_control_id',
-     *   );
+     * const response = await client.calls.actions.startStreaming(
+     *   'call_control_id',
+     * );
      * ```
      */
-    startAIAssistant(callControlID: string, body: ActionStartAIAssistantParams, options?: RequestOptions): APIPromise<ActionStartAIAssistantResponse>;
+    startStreaming(callControlID: string, body: ActionStartStreamingParams, options?: RequestOptions): APIPromise<ActionStartStreamingResponse>;
     /**
-     * Start a Conversation Relay session on an active call. Conversation Relay
-     * connects the call audio to your WebSocket so your application can exchange
-     * realtime messages with the caller while Telnyx handles speech recognition and
-     * text-to-speech. Only one AI Assistant or Conversation Relay session can be
-     * active on a call at a time.
+     * Stop streaming a call to a WebSocket.
      *
      * **Expected Webhooks:**
      *
-     * - `call.conversation.ended` - Sent when the Conversation Relay session ends. If
-     *   the customer WebSocket disconnects, the webhook payload `reason` is
-     *   `customer_disconnect`.
+     * - `streaming.stopped`
      *
      * @example
      * ```ts
-     * const response =
-     *   await client.calls.actions.startConversationRelay(
-     *     'call_control_id',
-     *   );
-     * ```
-     */
-    startConversationRelay(callControlID: string, body: ActionStartConversationRelayParams, options?: RequestOptions): APIPromise<ActionStartConversationRelayResponse>;
-    /**
-     * 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:**
-     *
-     * - `call.fork.started`
-     * - `call.fork.stopped`
-     *
-     * @example
-     * ```ts
-     * const response = await client.calls.actions.startForking(
+     * const response = await client.calls.actions.stopStreaming(
      *   'call_control_id',
      * );
      * ```
      */
-    startForking(callControlID: string, body: ActionStartForkingParams, options?: RequestOptions): APIPromise<ActionStartForkingResponse>;
+    stopStreaming(callControlID: string, body: ActionStopStreamingParams, options?: RequestOptions): APIPromise<ActionStopStreamingResponse>;
     /**
      * Noise Suppression Start (BETA)
      *
@@ -412,80 +551,31 @@ export declare class Actions extends APIResource {
      */
     startNoiseSuppression(callControlID: string, body: ActionStartNoiseSuppressionParams, options?: RequestOptions): APIPromise<ActionStartNoiseSuppressionResponse>;
     /**
-     * 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:**
-     *
-     * - `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>;
-    /**
-     * Start recording the call. Recording will stop on call hang-up, or can be
-     * initiated via the Stop Recording command.
-     *
-     * **Expected Webhooks:**
-     *
-     * - `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>;
-    /**
-     * Start siprec session to configured in SIPREC connector SRS.
-     *
-     * **Expected Webhooks:**
-     *
-     * - `siprec.started`
-     * - `siprec.stopped`
-     * - `siprec.failed`
+     * Noise Suppression Stop (BETA)
      *
      * @example
      * ```ts
-     * const response = await client.calls.actions.startSiprec(
-     *   'call_control_id',
-     * );
+     * const response =
+     *   await client.calls.actions.stopNoiseSuppression(
+     *     'call_control_id',
+     *   );
      * ```
      */
-    startSiprec(callControlID: string, body: ActionStartSiprecParams, options?: RequestOptions): APIPromise<ActionStartSiprecResponse>;
+    stopNoiseSuppression(callControlID: string, body: ActionStopNoiseSuppressionParams, options?: RequestOptions): APIPromise<ActionStopNoiseSuppressionResponse>;
     /**
-     * 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).
+     * 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.startStreaming(
-     *   'call_control_id',
-     * );
+     * const response =
+     *   await client.calls.actions.switchSupervisorRole(
+     *     'call_control_id',
+     *     { role: 'barge' },
+     *   );
      * ```
      */
-    startStreaming(callControlID: string, body: ActionStartStreamingParams, options?: RequestOptions): APIPromise<ActionStartStreamingResponse>;
+    switchSupervisorRole(callControlID: string, body: ActionSwitchSupervisorRoleParams, options?: RequestOptions): APIPromise<ActionSwitchSupervisorRoleResponse>;
     /**
      * Start real-time transcription. Transcription will stop on call hang-up, or can
      * be initiated via the Transcription stop command.
@@ -504,202 +594,112 @@ export declare class Actions extends APIResource {
      */
     startTranscription(callControlID: string, body: ActionStartTranscriptionParams, options?: RequestOptions): APIPromise<ActionStartTranscriptionResponse>;
     /**
-     * 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>;
-    /**
-     * Stop the active Conversation Relay session on a call.
+     * Stop real-time transcription.
      *
      * @example
      * ```ts
      * const response =
-     *   await client.calls.actions.stopConversationRelay(
+     *   await client.calls.actions.stopTranscription(
      *     'call_control_id',
      *   );
      * ```
      */
-    stopConversationRelay(callControlID: string, body: ActionStopConversationRelayParams, options?: RequestOptions): APIPromise<ActionStopConversationRelayResponse>;
-    /**
-     * Stop forking a call.
-     *
-     * **Expected Webhooks:**
-     *
-     * - `call.fork.stopped`
-     *
-     * @example
-     * ```ts
-     * const response = await client.calls.actions.stopForking(
-     *   'call_control_id',
-     * );
-     * ```
-     */
-    stopForking(callControlID: string, body: ActionStopForkingParams, options?: RequestOptions): APIPromise<ActionStopForkingResponse>;
+    stopTranscription(callControlID: string, body: ActionStopTranscriptionParams, options?: RequestOptions): APIPromise<ActionStopTranscriptionResponse>;
     /**
-     * Stop current gather.
+     * Transfer a call to a new destination. If the transfer is unsuccessful, a
+     * `call.hangup` webhook for the other call (Leg B) will be sent indicating that
+     * the transfer could not be completed. The original call will remain active and
+     * may be issued additional commands, potentially transfering the call to an
+     * alternate destination.
      *
      * **Expected Webhooks:**
      *
-     * - `call.gather.ended`
+     * - `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.stopGather(
+     * const response = await client.calls.actions.transfer(
      *   'call_control_id',
+     *   {
+     *     to: '+18005550100 or sip:username@sip.telnyx.com;secure=srtp',
+     *   },
      * );
      * ```
      */
-    stopGather(callControlID: string, body: ActionStopGatherParams, options?: RequestOptions): APIPromise<ActionStopGatherResponse>;
+    transfer(callControlID: string, body: ActionTransferParams, options?: RequestOptions): APIPromise<ActionTransferResponse>;
     /**
-     * Noise Suppression Stop (BETA)
+     * Add messages to the conversation started by an AI assistant on the call.
      *
      * @example
      * ```ts
      * const response =
-     *   await client.calls.actions.stopNoiseSuppression(
+     *   await client.calls.actions.addAIAssistantMessages(
      *     'call_control_id',
      *   );
      * ```
      */
-    stopNoiseSuppression(callControlID: string, body: ActionStopNoiseSuppressionParams, options?: RequestOptions): APIPromise<ActionStopNoiseSuppressionResponse>;
-    /**
-     * Stop audio being played on the call.
-     *
-     * **Expected Webhooks:**
-     *
-     * - `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>;
-    /**
-     * Stop recording the call.
-     *
-     * **Expected Webhooks:**
-     *
-     * - `call.recording.saved`
-     *
-     * @example
-     * ```ts
-     * const response = await client.calls.actions.stopRecording(
-     *   'call_control_id',
-     * );
-     * ```
-     */
-    stopRecording(callControlID: string, body: ActionStopRecordingParams, options?: RequestOptions): APIPromise<ActionStopRecordingResponse>;
+    addAIAssistantMessages(callControlID: string, body: ActionAddAIAssistantMessagesParams, options?: RequestOptions): APIPromise<ActionAddAIAssistantMessagesResponse>;
     /**
-     * Stop SIPREC session.
-     *
-     * **Expected Webhooks:**
-     *
-     * - `siprec.stopped`
+     * Add a participant to an existing AI assistant conversation. Use this command to
+     * bring an additional call leg into a running AI conversation.
      *
      * @example
      * ```ts
-     * const response = await client.calls.actions.stopSiprec(
+     * const response = await client.calls.actions.joinAIAssistant(
      *   'call_control_id',
+     *   {
+     *     conversation_id: 'v3:abc123',
+     *     participant: { id: 'v3:abc123def456', role: 'user' },
+     *   },
      * );
      * ```
      */
-    stopSiprec(callControlID: string, body: ActionStopSiprecParams, options?: RequestOptions): APIPromise<ActionStopSiprecResponse>;
+    joinAIAssistant(callControlID: string, body: ActionJoinAIAssistantParams, options?: RequestOptions): APIPromise<ActionJoinAIAssistantResponse>;
     /**
-     * Stop streaming a call to a WebSocket.
+     * Start a Conversation Relay session on an active call. Conversation Relay
+     * connects the call audio to your WebSocket so your application can exchange
+     * realtime messages with the caller while Telnyx handles speech recognition and
+     * text-to-speech. Only one AI Assistant or Conversation Relay session can be
+     * active on a call at a time.
      *
      * **Expected Webhooks:**
      *
-     * - `streaming.stopped`
-     *
-     * @example
-     * ```ts
-     * const response = await client.calls.actions.stopStreaming(
-     *   'call_control_id',
-     * );
-     * ```
-     */
-    stopStreaming(callControlID: string, body: ActionStopStreamingParams, options?: RequestOptions): APIPromise<ActionStopStreamingResponse>;
-    /**
-     * Stop real-time transcription.
+     * - `call.conversation.ended` - Sent when the Conversation Relay session ends. If
+     *   the customer WebSocket disconnects, the webhook payload `reason` is
+     *   `customer_disconnect`.
      *
      * @example
      * ```ts
      * const response =
-     *   await client.calls.actions.stopTranscription(
+     *   await client.calls.actions.startConversationRelay(
      *     'call_control_id',
      *   );
      * ```
      */
-    stopTranscription(callControlID: string, body: ActionStopTranscriptionParams, options?: RequestOptions): APIPromise<ActionStopTranscriptionResponse>;
+    startConversationRelay(callControlID: string, body: ActionStartConversationRelayParams, options?: RequestOptions): APIPromise<ActionStartConversationRelayResponse>;
     /**
-     * Switch the supervisor role for a bridged call. This allows switching between
-     * different supervisor modes during an active call
+     * Stop the active Conversation Relay session on a call.
      *
      * @example
      * ```ts
      * const response =
-     *   await client.calls.actions.switchSupervisorRole(
+     *   await client.calls.actions.stopConversationRelay(
      *     'call_control_id',
-     *     { role: 'barge' },
      *   );
      * ```
      */
-    switchSupervisorRole(callControlID: string, body: ActionSwitchSupervisorRoleParams, options?: RequestOptions): APIPromise<ActionSwitchSupervisorRoleResponse>;
-    /**
-     * Transfer a call to a new destination. If the transfer is unsuccessful, a
-     * `call.hangup` webhook for the other call (Leg B) will be sent indicating that
-     * the transfer could not be completed. The original call will remain active and
-     * may be issued additional commands, potentially transfering the call to an
-     * alternate destination.
-     *
-     * **Expected Webhooks:**
-     *
-     * - `call.initiated`
-     * - `call.bridged` to Leg B
-     * - `call.answered` or `call.hangup`
-     * - `call.machine.detection.ended` if `answering_machine_detection` was requested
-     * - `call.machine.greeting.ended` if `answering_machine_detection` was requested
-     *   to detect the end of machine greeting
-     * - `call.machine.premium.detection.ended` if
-     *   `answering_machine_detection=premium` was requested
-     * - `call.machine.premium.greeting.ended` if `answering_machine_detection=premium`
-     *   was requested and a beep was detected
-     *
-     * @example
-     * ```ts
-     * const response = await client.calls.actions.transfer(
-     *   'call_control_id',
-     *   {
-     *     to: '+18005550100 or sip:username@sip.telnyx.com;secure=srtp',
-     *   },
-     * );
-     * ```
-     */
-    transfer(callControlID: string, body: ActionTransferParams, options?: RequestOptions): APIPromise<ActionTransferResponse>;
-    /**
-     * 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>;
-}
-export interface AIAssistantJoinParticipant {
+    stopConversationRelay(callControlID: string, body: ActionStopConversationRelayParams, options?: RequestOptions): APIPromise<ActionStopConversationRelayResponse>;
+}
+export interface AIAssistantJoinParticipant {
     /**
      * The call_control_id of the participant to add to the conversation.
      */
@@ -1472,7 +1472,13 @@ export interface ActionTransferResponse {
 export interface ActionUpdateClientStateResponse {
     data?: CallControlCommandResult;
 }
-export interface ActionAddAIAssistantMessagesParams {
+export interface ActionStartAIAssistantParams {
+    /**
+     * AI Assistant configuration. All fields except `id` are optional — the
+     * assistant's stored configuration will be used as fallback for any omitted
+     * fields.
+     */
+    assistant?: CallsAPI.CallAssistantRequest;
     /**
      * Use this field to add state to every subsequent webhook. It must be a valid
      * Base-64 encoded string.
@@ -1484,9 +1490,82 @@ export interface ActionAddAIAssistantMessagesParams {
      */
     command_id?: string;
     /**
-     * The messages to add to the conversation.
+     * 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.
      */
-    messages?: Array<UserMessage | AssistantMessage | ToolMessage | SystemMessage | DeveloperMessage>;
+    greeting?: string;
+    /**
+     * Settings for handling user interruptions during assistant speech
+     */
+    interruption_settings?: InterruptionSettings;
+    /**
+     * A list of messages to seed the conversation history before the assistant starts.
+     * Follows the same message format as the `ai_assistant_add_messages` command.
+     */
+    message_history?: Array<UserMessage | AssistantMessage | ToolMessage | SystemMessage | DeveloperMessage>;
+    /**
+     * A list of participants to add to the conversation when it starts.
+     */
+    participants?: Array<AIAssistantJoinParticipant>;
+    /**
+     * When `true`, a webhook is sent each time the conversation message history is
+     * updated.
+     */
+    send_message_history_updates?: 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 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>`
+     * - **Inworld:** Use `Inworld.<ModelId>.<VoiceId>` (e.g., `Inworld.Mini.Loretta`,
+     *   `Inworld.Max.Oliver`, `Inworld.TTS2.Loretta`). Supported models: `Mini`,
+     *   `Max`, `TTS2`.
+     * - **xAI:** Use `xAI.<VoiceId>` (e.g., `xAI.eve`). Available voices: `eve`,
+     *   `ara`, `rex`, `sal`, `leo`.
+     */
+    voice?: string;
+    /**
+     * The settings associated with the voice selected
+     */
+    voice_settings?: ElevenLabsVoiceSettings | TelnyxVoiceSettings | AwsVoiceSettings | Shared.AzureVoiceSettings | Shared.RimeVoiceSettings | Shared.ResembleVoiceSettings | Shared.XaiVoiceSettings;
+}
+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 ActionAnswerParams {
     /**
@@ -1793,6 +1872,13 @@ export interface ActionBridgeParams {
      */
     video_room_id?: string;
 }
+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 interface ActionEnqueueParams {
     /**
      * The name of the queue the call should be put in. If a queue with a given name
@@ -1825,6 +1911,51 @@ export interface ActionEnqueueParams {
      */
     max_wait_time_secs?: number;
 }
+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 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 ActionGatherParams {
     /**
      * Use this field to add state to every subsequent webhook. It must be a valid
@@ -1872,6 +2003,18 @@ export interface ActionGatherParams {
      */
     valid_digits?: string;
 }
+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 ActionGatherUsingAIParams {
     /**
      * The parameters described as a JSON Schema object that needs to be gathered by
@@ -2175,667 +2318,43 @@ export interface ActionGatherUsingSpeakParams {
      * 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 | Shared.MinimaxVoiceSettings | Shared.AzureVoiceSettings | Shared.RimeVoiceSettings | Shared.ResembleVoiceSettings | Shared.InworldVoiceSettings | Shared.XaiVoiceSettings;
-}
-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;
-    /**
-     * Custom headers to be added to the SIP BYE message.
-     */
-    custom_headers?: Array<CallsAPI.CustomSipHeader>;
-}
-export interface ActionJoinAIAssistantParams {
-    /**
-     * The ID of the AI assistant conversation to join.
-     */
-    conversation_id: string;
-    participant: AIAssistantJoinParticipant;
-    /**
-     * 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-US-BrianMultilingualNeural`,
-     *   `Azure.en-US-Ava:DragonHDLatestNeural`). For a complete list of voices, go to
-     *   [Azure Voice Gallery](https://speech.microsoft.com/portal/voicegallery). Use
-     *   `voice_settings` to configure custom deployments, regions, or API keys.
-     * - **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>` (e.g., `Telnyx.KokoroTTS.af`).
-     *   Use `voice_settings` to configure voice_speed and other synthesis parameters.
-     * - **Minimax:** Use `Minimax.<ModelId>.<VoiceId>` (e.g.,
-     *   `Minimax.speech-02-hd.Wise_Woman`). Supported models: `speech-02-turbo`,
-     *   `speech-02-hd`, `speech-2.6-turbo`, `speech-2.8-turbo`. Use `voice_settings`
-     *   to configure speed, volume, pitch, and language_boost.
-     * - **Rime:** Use `Rime.<model_id>.<voice_id>` (e.g., `Rime.Arcana.cove`).
-     *   Supported model_ids: `Arcana`, `Mist`, `ArcanaV3`, `Coda`. Use
-     *   `voice_settings` to configure voice_speed.
-     * - **Resemble:** Use `Resemble.Turbo.<voice_id>` (e.g.,
-     *   `Resemble.Turbo.my_voice`). Only `Turbo` model is supported. Use
-     *   `voice_settings` to configure precision, sample_rate, and format.
-     * - **Inworld:** Use `Inworld.<ModelId>.<VoiceId>` (e.g., `Inworld.Mini.Loretta`,
-     *   `Inworld.Max.Oliver`, `Inworld.TTS2.Loretta`). Supported models: `Mini`,
-     *   `Max`, `TTS2`. Use `voice_settings` to configure `delivery_mode` (`STABLE`,
-     *   `BALANCED`, `CREATIVE`), supported by `TTS2` only.
-     * - **xAI:** Use `xAI.<VoiceId>` (e.g., `xAI.eve`). Available voices: `eve`,
-     *   `ara`, `rex`, `sal`, `leo`.
-     *
-     * 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 number of times to play the audio file. Use `infinity` to loop indefinitely.
-     * Defaults to 1.
-     */
-    loop?: Loopcount;
-    /**
-     * 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;
-    /**
-     * Specifies which legs of the call should receive the spoken audio.
-     */
-    target_legs?: 'self' | 'opposite' | 'both';
-    /**
-     * The settings associated with the voice selected
-     */
-    voice_settings?: ElevenLabsVoiceSettings | TelnyxVoiceSettings | AwsVoiceSettings | Shared.MinimaxVoiceSettings | Shared.AzureVoiceSettings | Shared.RimeVoiceSettings | Shared.ResembleVoiceSettings | Shared.InworldVoiceSettings | Shared.XaiVoiceSettings;
-}
-export interface ActionStartAIAssistantParams {
-    /**
-     * AI Assistant configuration. All fields except `id` are optional — the
-     * assistant's stored configuration will be used as fallback for any omitted
-     * fields.
-     */
-    assistant?: CallsAPI.CallAssistantRequest;
-    /**
-     * 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;
-    /**
-     * A list of messages to seed the conversation history before the assistant starts.
-     * Follows the same message format as the `ai_assistant_add_messages` command.
-     */
-    message_history?: Array<UserMessage | AssistantMessage | ToolMessage | SystemMessage | DeveloperMessage>;
-    /**
-     * A list of participants to add to the conversation when it starts.
-     */
-    participants?: Array<AIAssistantJoinParticipant>;
-    /**
-     * When `true`, a webhook is sent each time the conversation message history is
-     * updated.
-     */
-    send_message_history_updates?: 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 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>`
-     * - **Inworld:** Use `Inworld.<ModelId>.<VoiceId>` (e.g., `Inworld.Mini.Loretta`,
-     *   `Inworld.Max.Oliver`, `Inworld.TTS2.Loretta`). Supported models: `Mini`,
-     *   `Max`, `TTS2`.
-     * - **xAI:** Use `xAI.<VoiceId>` (e.g., `xAI.eve`). Available voices: `eve`,
-     *   `ara`, `rex`, `sal`, `leo`.
-     */
-    voice?: string;
-    /**
-     * The settings associated with the voice selected
-     */
-    voice_settings?: ElevenLabsVoiceSettings | TelnyxVoiceSettings | AwsVoiceSettings | Shared.AzureVoiceSettings | Shared.RimeVoiceSettings | Shared.ResembleVoiceSettings | Shared.XaiVoiceSettings;
-}
-export interface ActionStartConversationRelayParams {
-    /**
-     * Custom parameters for the Conversation Relay session. Pass key-value data as
-     * `assistant.dynamic_variables` to make it available to the relay session.
-     */
-    assistant?: ActionStartConversationRelayParams.Assistant;
-    /**
-     * Use this field to add state to subsequent webhooks. 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;
-    /**
-     * Enable DTMF detection for the relay session.
-     */
-    conversation_relay_dtmf_detection?: boolean;
-    /**
-     * Conversation Relay connection settings. This object can provide `url`,
-     * `dtmf_detection`, `interruptible`, `interruptible_greeting`, and `languages`.
-     * Top-level aliases override nested values when both are present.
-     */
-    conversation_relay_settings?: ActionStartConversationRelayParams.ConversationRelaySettings;
-    /**
-     * WebSocket URL for your Conversation Relay server. Must start with `ws://` or
-     * `wss://`.
-     */
-    conversation_relay_url?: string;
-    /**
-     * Custom key-value parameters forwarded to the relay session as
-     * `assistant.dynamic_variables`. If `assistant.dynamic_variables` is also present,
-     * these values are merged in.
-     */
-    custom_parameters?: {
-        [key: string]: unknown;
-    };
-    /**
-     * Public alias for `conversation_relay_dtmf_detection`. If both are present, this
-     * value wins.
-     */
-    dtmf_detection?: boolean;
-    /**
-     * Text played when the relay session starts.
-     */
-    greeting?: string;
-    /**
-     * Controls when caller input can interrupt assistant speech. `any` allows speech
-     * or DTMF interruptions; `none` disables interruptions; `speech` allows speech
-     * only; `dtmf` allows DTMF only.
-     */
-    interruptible?: ConversationRelayInterruptible;
-    /**
-     * Controls when caller input can interrupt assistant speech. `any` allows speech
-     * or DTMF interruptions; `none` disables interruptions; `speech` allows speech
-     * only; `dtmf` allows DTMF only.
-     */
-    interruptible_greeting?: ConversationRelayInterruptible;
-    /**
-     * Settings for handling caller interruptions during Conversation Relay speech.
-     */
-    interruption_settings?: CallsAPI.ConversationRelayInterruptionSettings;
-    /**
-     * Default language for the relay session. This value is used for both
-     * text-to-speech and speech recognition.
-     */
-    language?: string;
-    /**
-     * Per-language TTS and transcription settings.
-     */
-    languages?: Array<CallsAPI.ConversationRelayLanguage>;
-    /**
-     * Structured voice provider. Must be supplied together with `structured_provider`.
-     */
-    provider?: string;
-    /**
-     * Provider-specific structured voice settings. Must be supplied together with
-     * `provider`; Telnyx sends the value as the nested provider configuration for
-     * Conversation Relay.
-     */
-    structured_provider?: {
-        [key: string]: unknown;
-    };
-    /**
-     * @deprecated Not supported for Conversation Relay start requests. Use
-     * `transcription_engine` and `transcription_engine_config` instead.
-     */
-    transcription?: {
-        [key: string]: unknown;
-    };
-    /**
-     * Engine to use for speech recognition. Legacy values `A` - `Google`, `B` -
-     * `Telnyx` are supported for backward compatibility. For Conversation Relay, use
-     * this field with `transcription_engine_config`; the `transcription` object is not
-     * supported.
-     */
-    transcription_engine?: 'Google' | 'Telnyx' | 'Deepgram' | 'Azure' | 'xAI' | 'AssemblyAI' | 'Speechmatics' | 'Soniox' | 'A' | 'B';
-    /**
-     * Engine-specific transcription settings for Conversation Relay. This accepts the
-     * same provider-specific options used by the Call Transcription Start command,
-     * such as `transcription_model`, without requiring the engine discriminator to be
-     * repeated inside this object.
-     */
-    transcription_engine_config?: {
-        [key: string]: unknown;
-    };
-    /**
-     * Text-to-speech provider. If omitted, Telnyx derives it from `voice` or
-     * `provider`.
-     */
-    tts_provider?: string;
-    /**
-     * Public alias for `conversation_relay_url`. Must start with `ws://` or `wss://`.
-     * If both are present, this value wins.
-     */
-    url?: string;
-    /**
-     * 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>`
-     * - **Inworld:** Use `Inworld.<ModelId>.<VoiceId>` (e.g., `Inworld.Mini.Loretta`,
-     *   `Inworld.Max.Oliver`, `Inworld.TTS2.Loretta`). Supported models: `Mini`,
-     *   `Max`, `TTS2`.
-     * - **xAI:** Use `xAI.<VoiceId>` (e.g., `xAI.eve`). Available voices: `eve`,
-     *   `ara`, `rex`, `sal`, `leo`.
-     */
-    voice?: string;
-    /**
-     * The settings associated with the voice selected
-     */
-    voice_settings?: ElevenLabsVoiceSettings | TelnyxVoiceSettings | AwsVoiceSettings | Shared.MinimaxVoiceSettings | Shared.AzureVoiceSettings | Shared.RimeVoiceSettings | Shared.ResembleVoiceSettings | Shared.InworldVoiceSettings | Shared.XaiVoiceSettings;
-}
-export declare namespace ActionStartConversationRelayParams {
-    /**
-     * Custom parameters for the Conversation Relay session. Pass key-value data as
-     * `assistant.dynamic_variables` to make it available to the relay session.
-     */
-    interface Assistant {
-        /**
-         * Custom key-value parameters forwarded to the Conversation Relay session.
-         */
-        dynamic_variables?: {
-            [key: string]: string;
-        };
-        [k: string]: unknown;
-    }
-    /**
-     * Conversation Relay connection settings. This object can provide `url`,
-     * `dtmf_detection`, `interruptible`, `interruptible_greeting`, and `languages`.
-     * Top-level aliases override nested values when both are present.
-     */
-    interface ConversationRelaySettings {
-        /**
-         * WebSocket URL for your Conversation Relay server. Must start with `ws://` or
-         * `wss://`.
-         */
-        url: string;
-        /**
-         * Whether to enable DTMF detection during the relay session.
-         */
-        dtmf_detection?: boolean;
-        /**
-         * Controls when caller input can interrupt assistant speech. `any` allows speech
-         * or DTMF interruptions; `none` disables interruptions; `speech` allows speech
-         * only; `dtmf` allows DTMF only.
-         */
-        interruptible?: ActionsAPI.ConversationRelayInterruptible;
-        /**
-         * Controls when caller input can interrupt assistant speech. `any` allows speech
-         * or DTMF interruptions; `none` disables interruptions; `speech` allows speech
-         * only; `dtmf` allows DTMF only.
-         */
-        interruptible_greeting?: ActionsAPI.ConversationRelayInterruptible;
-        /**
-         * Language-specific TTS and transcription settings.
-         */
-        languages?: Array<CallsAPI.ConversationRelayLanguage>;
-    }
-}
-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. For backward compatibility, engines A,
-     * B, C, and D are also supported, but are deprecated: A - Denoiser B -
-     * DeepFilterNet C - Krisp D - AiCoustics
-     */
-    noise_suppression_engine?: 'Denoiser' | 'DeepFilterNet' | 'Krisp' | 'AiCoustics' | 'aic_l_quail' | 'aic_l_rook' | 'aic_s_quail' | 'aic_s_rook' | 'quail_voice_focus_s' | 'quail_voice_focus_xs';
-    /**
-     * Configuration parameters for noise suppression engines. Different engines
-     * support different parameters.
-     */
-    noise_suppression_engine_config?: ActionStartNoiseSuppressionParams.NoiseSuppressionEngineConfig;
-}
-export declare namespace ActionStartNoiseSuppressionParams {
-    /**
-     * Configuration parameters for noise suppression engines. Different engines
-     * support different parameters.
-     */
-    interface NoiseSuppressionEngineConfig {
-        /**
-         * The attenuation limit for noise suppression (0-100). Only applicable for
-         * DeepFilterNet.
-         */
-        attenuation_limit?: number;
-        /**
-         * Enhancement intensity (0.0-1.0). Only applicable for AiCoustics.
-         */
-        enhancement_level?: number;
-        /**
-         * AiCoustics model family. 'sparrow' optimized for human-to-human calls, 'quail'
-         * optimized for Voice AI/STT. Only applicable for AiCoustics.
-         */
-        family?: 'sparrow' | 'quail';
-        /**
-         * Processing mode. Only applicable for DeepFilterNet.
-         */
-        mode?: 'standard' | 'advanced';
-        /**
-         * The Krisp model to use. Only applicable for Krisp.
-         */
-        model?: 'krisp-viva-tel-v2.kef' | 'krisp-viva-tel-lite-v1.kef' | 'krisp-viva-pro-v1.kef' | 'krisp-viva-ss-v1.kef';
-        /**
-         * AiCoustics model size. 's' and 'l' work with both families. 'xs' and 'xxs' are
-         * sparrow-only. 'vf_l' and 'vf_1_1_l' are quail-only. Only applicable for
-         * AiCoustics.
-         */
-        size?: 's' | 'l' | 'xs' | 'xxs' | 'vf_l' | 'vf_1_1_l';
-        /**
-         * Suppression level (0.0-100.0). Only applicable for Krisp.
-         */
-        suppression_level?: number;
-        /**
-         * Voice gain multiplier (0.1-4.0). Only applicable for AiCoustics.
-         */
-        voice_gain?: number;
-    }
+    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 | Shared.MinimaxVoiceSettings | Shared.AzureVoiceSettings | Shared.RimeVoiceSettings | Shared.ResembleVoiceSettings | Shared.InworldVoiceSettings | Shared.XaiVoiceSettings;
+}
+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;
+    /**
+     * Custom headers to be added to the SIP BYE message.
+     */
+    custom_headers?: Array<CallsAPI.CustomSipHeader>;
+}
+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 ActionStartPlaybackParams {
     /**
@@ -2900,6 +2419,59 @@ export interface ActionStartPlaybackParams {
      */
     target_legs?: 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 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 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 ActionStartRecordingParams {
     /**
      * When `dual`, final audio file will be stereo recorded with the first leg on
@@ -2979,50 +2551,271 @@ export interface ActionStartRecordingParams {
      */
     transcription_profanity_filter?: boolean;
     /**
-     * Enables speaker diarization. Applies to `google` engine only.
+     * 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 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 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 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 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 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 ActionSpeakParams {
+    /**
+     * The text or SSML to be converted into speech. There is a 3,000 character limit.
      */
-    transcription_speaker_diarization?: boolean;
+    payload: string;
     /**
-     * When set to `trim-silence`, silence will be removed from the beginning and end
-     * of the recording.
+     * 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-US-BrianMultilingualNeural`,
+     *   `Azure.en-US-Ava:DragonHDLatestNeural`). For a complete list of voices, go to
+     *   [Azure Voice Gallery](https://speech.microsoft.com/portal/voicegallery). Use
+     *   `voice_settings` to configure custom deployments, regions, or API keys.
+     * - **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>` (e.g., `Telnyx.KokoroTTS.af`).
+     *   Use `voice_settings` to configure voice_speed and other synthesis parameters.
+     * - **Minimax:** Use `Minimax.<ModelId>.<VoiceId>` (e.g.,
+     *   `Minimax.speech-02-hd.Wise_Woman`). Supported models: `speech-02-turbo`,
+     *   `speech-02-hd`, `speech-2.6-turbo`, `speech-2.8-turbo`. Use `voice_settings`
+     *   to configure speed, volume, pitch, and language_boost.
+     * - **Rime:** Use `Rime.<model_id>.<voice_id>` (e.g., `Rime.Arcana.cove`).
+     *   Supported model_ids: `Arcana`, `Mist`, `ArcanaV3`, `Coda`. Use
+     *   `voice_settings` to configure voice_speed.
+     * - **Resemble:** Use `Resemble.Turbo.<voice_id>` (e.g.,
+     *   `Resemble.Turbo.my_voice`). Only `Turbo` model is supported. Use
+     *   `voice_settings` to configure precision, sample_rate, and format.
+     * - **Inworld:** Use `Inworld.<ModelId>.<VoiceId>` (e.g., `Inworld.Mini.Loretta`,
+     *   `Inworld.Max.Oliver`, `Inworld.TTS2.Loretta`). Supported models: `Mini`,
+     *   `Max`, `TTS2`. Use `voice_settings` to configure `delivery_mode` (`STABLE`,
+     *   `BALANCED`, `CREATIVE`), supported by `TTS2` only.
+     * - **xAI:** Use `xAI.<VoiceId>` (e.g., `xAI.eve`). Available voices: `eve`,
+     *   `ara`, `rex`, `sal`, `leo`.
+     *
+     * For service_level basic, you may define the gender of the speaker (male or
+     * female).
      */
-    trim?: 'trim-silence';
-}
-export interface ActionStartSiprecParams {
+    voice: string;
     /**
      * 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.
+     * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+     * the same `command_id` for the same `call_control_id`.
      */
-    connector_name?: string;
+    command_id?: string;
     /**
-     * When set, custom parameters will be added as metadata
-     * (recording.session.ExtensionParameters). Otherwise, they’ll be added to sip
-     * headers.
+     * The language you want spoken. This parameter is ignored when a `Polly.*` voice
+     * is specified.
      */
-    include_metadata_custom_headers?: true | false;
+    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';
     /**
-     * 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.
+     * The number of times to play the audio file. Use `infinity` to loop indefinitely.
+     * Defaults to 1.
      */
-    secure?: true | false;
+    loop?: Loopcount;
     /**
-     * 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.
+     * The type of the provided payload. The payload can either be plain text, or
+     * Speech Synthesis Markup Language (SSML).
      */
-    session_timeout_secs?: number;
+    payload_type?: 'text' | 'ssml';
     /**
-     * Specifies SIP transport protocol.
+     * This parameter impacts speech quality, language options and payload types. When
+     * using `basic`, only the `en-US` language and payload type `text` are allowed.
      */
-    sip_transport?: 'udp' | 'tcp' | 'tls';
+    service_level?: 'basic' | 'premium';
     /**
-     * Specifies which track should be sent on siprec session.
+     * 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.
      */
-    siprec_track?: 'inbound_track' | 'outbound_track' | 'both_tracks';
+    stop?: string;
+    /**
+     * Specifies which legs of the call should receive the spoken audio.
+     */
+    target_legs?: 'self' | 'opposite' | 'both';
+    /**
+     * The settings associated with the voice selected
+     */
+    voice_settings?: ElevenLabsVoiceSettings | TelnyxVoiceSettings | AwsVoiceSettings | Shared.MinimaxVoiceSettings | Shared.AzureVoiceSettings | Shared.RimeVoiceSettings | Shared.ResembleVoiceSettings | Shared.InworldVoiceSettings | Shared.XaiVoiceSettings;
 }
 export interface ActionStartStreamingParams {
     /**
@@ -3092,7 +2885,7 @@ export declare namespace ActionStartStreamingParams {
         value?: string;
     }
 }
-export interface ActionStartTranscriptionParams {
+export interface ActionStopStreamingParams {
     /**
      * Use this field to add state to every subsequent webhook. It must be a valid
      * Base-64 encoded string.
@@ -3104,19 +2897,12 @@ export interface ActionStartTranscriptionParams {
      */
     command_id?: string;
     /**
-     * Engine to use for speech recognition. Legacy values `A` - `Google`, `B` -
-     * `Telnyx` are supported for backward compatibility.
-     */
-    transcription_engine?: 'Google' | 'Telnyx' | 'Deepgram' | 'Azure' | 'xAI' | 'AssemblyAI' | 'Speechmatics' | 'Soniox' | 'Parakeet' | 'A' | 'B';
-    transcription_engine_config?: TranscriptionEngineGoogleConfig | TranscriptionEngineTelnyxConfig | TranscriptionEngineAzureConfig | TranscriptionEngineXaiConfig | TranscriptionEngineAssemblyaiConfig | TranscriptionEngineSpeechmaticsConfig | TranscriptionEngineSonioxConfig | TranscriptionEngineParakeetConfig | TranscriptionEngineAConfig | TranscriptionEngineBConfig | DeepgramNova2Config | DeepgramNova3Config;
-    /**
-     * 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`.
+     * Identifies the stream. If the `stream_id` is not provided the command stops all
+     * streams associated with a given `call_control_id`.
      */
-    transcription_tracks?: string;
+    stream_id?: string;
 }
-export interface ActionStopAIAssistantParams {
+export interface ActionStartNoiseSuppressionParams {
     /**
      * Use this field to add state to every subsequent webhook. It must be a valid
      * Base-64 encoded string.
@@ -3127,47 +2913,65 @@ export interface ActionStopAIAssistantParams {
      * the same `command_id` for the same `call_control_id`.
      */
     command_id?: string;
-}
-export interface ActionStopConversationRelayParams {
-    /**
-     * Use this field to add state to subsequent webhooks. 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.
+     * The direction of the audio stream to be noise suppressed.
      */
-    client_state?: string;
+    direction?: 'inbound' | 'outbound' | 'both';
     /**
-     * Use this field to avoid duplicate commands. Telnyx will ignore any command with
-     * the same `command_id` for the same `call_control_id`.
+     * The engine to use for noise suppression. For backward compatibility, engines A,
+     * B, C, and D are also supported, but are deprecated: A - Denoiser B -
+     * DeepFilterNet C - Krisp D - AiCoustics
      */
-    command_id?: string;
+    noise_suppression_engine?: 'Denoiser' | 'DeepFilterNet' | 'Krisp' | 'AiCoustics' | 'aic_l_quail' | 'aic_l_rook' | 'aic_s_quail' | 'aic_s_rook' | 'quail_voice_focus_s' | 'quail_voice_focus_xs';
     /**
-     * Optionally specify a `stream_type`. This should match the `stream_type` that was
-     * used in `fork_start` command to properly stop the fork.
+     * Configuration parameters for noise suppression engines. Different engines
+     * support different parameters.
      */
-    stream_type?: 'raw' | 'decrypted';
+    noise_suppression_engine_config?: ActionStartNoiseSuppressionParams.NoiseSuppressionEngineConfig;
 }
-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;
+export declare namespace ActionStartNoiseSuppressionParams {
     /**
-     * Use this field to avoid duplicate commands. Telnyx will ignore any command with
-     * the same `command_id` for the same `call_control_id`.
+     * Configuration parameters for noise suppression engines. Different engines
+     * support different parameters.
      */
-    command_id?: string;
+    interface NoiseSuppressionEngineConfig {
+        /**
+         * The attenuation limit for noise suppression (0-100). Only applicable for
+         * DeepFilterNet.
+         */
+        attenuation_limit?: number;
+        /**
+         * Enhancement intensity (0.0-1.0). Only applicable for AiCoustics.
+         */
+        enhancement_level?: number;
+        /**
+         * AiCoustics model family. 'sparrow' optimized for human-to-human calls, 'quail'
+         * optimized for Voice AI/STT. Only applicable for AiCoustics.
+         */
+        family?: 'sparrow' | 'quail';
+        /**
+         * Processing mode. Only applicable for DeepFilterNet.
+         */
+        mode?: 'standard' | 'advanced';
+        /**
+         * The Krisp model to use. Only applicable for Krisp.
+         */
+        model?: 'krisp-viva-tel-v2.kef' | 'krisp-viva-tel-lite-v1.kef' | 'krisp-viva-pro-v1.kef' | 'krisp-viva-ss-v1.kef';
+        /**
+         * AiCoustics model size. 's' and 'l' work with both families. 'xs' and 'xxs' are
+         * sparrow-only. 'vf_l' and 'vf_1_1_l' are quail-only. Only applicable for
+         * AiCoustics.
+         */
+        size?: 's' | 'l' | 'xs' | 'xxs' | 'vf_l' | 'vf_1_1_l';
+        /**
+         * Suppression level (0.0-100.0). Only applicable for Krisp.
+         */
+        suppression_level?: number;
+        /**
+         * Voice gain multiplier (0.1-4.0). Only applicable for AiCoustics.
+         */
+        voice_gain?: number;
+    }
 }
 export interface ActionStopNoiseSuppressionParams {
     /**
@@ -3181,44 +2985,14 @@ export interface ActionStopNoiseSuppressionParams {
      */
     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;
+export interface ActionSwitchSupervisorRoleParams {
     /**
-     * Uniquely identifies the resource.
+     * The supervisor role to switch to. 'barge' allows speaking to both parties,
+     * 'whisper' allows speaking to caller only, 'monitor' allows listening only.
      */
-    recording_id?: string;
+    role: 'barge' | 'whisper' | 'monitor';
 }
-export interface ActionStopSiprecParams {
+export interface ActionStartTranscriptionParams {
     /**
      * Use this field to add state to every subsequent webhook. It must be a valid
      * Base-64 encoded string.
@@ -3229,23 +3003,18 @@ export interface ActionStopSiprecParams {
      * 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`.
+     * Engine to use for speech recognition. Legacy values `A` - `Google`, `B` -
+     * `Telnyx` are supported for backward compatibility.
      */
-    command_id?: string;
+    transcription_engine?: 'Google' | 'Telnyx' | 'Deepgram' | 'Azure' | 'xAI' | 'AssemblyAI' | 'Speechmatics' | 'Soniox' | 'Parakeet' | 'A' | 'B';
+    transcription_engine_config?: TranscriptionEngineGoogleConfig | TranscriptionEngineTelnyxConfig | TranscriptionEngineAzureConfig | TranscriptionEngineXaiConfig | TranscriptionEngineAssemblyaiConfig | TranscriptionEngineSpeechmaticsConfig | TranscriptionEngineSonioxConfig | TranscriptionEngineParakeetConfig | TranscriptionEngineAConfig | TranscriptionEngineBConfig | DeepgramNova2Config | DeepgramNova3Config;
     /**
-     * Identifies the stream. If the `stream_id` is not provided the command stops all
-     * streams associated with a given `call_control_id`.
+     * 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`.
      */
-    stream_id?: string;
+    transcription_tracks?: string;
 }
 export interface ActionStopTranscriptionParams {
     /**
@@ -3259,13 +3028,6 @@ export interface ActionStopTranscriptionParams {
      */
     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. For SIP URI destinations, append
@@ -3563,14 +3325,252 @@ export declare namespace ActionTransferParams {
         retries_ms?: Array<number>;
     }
 }
-export interface ActionUpdateClientStateParams {
+export interface ActionAddAIAssistantMessagesParams {
     /**
      * Use this field to add state to every subsequent webhook. It must be a valid
      * Base-64 encoded string.
      */
-    client_state: 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 messages to add to the conversation.
+     */
+    messages?: Array<UserMessage | AssistantMessage | ToolMessage | SystemMessage | DeveloperMessage>;
+}
+export interface ActionJoinAIAssistantParams {
+    /**
+     * The ID of the AI assistant conversation to join.
+     */
+    conversation_id: string;
+    participant: AIAssistantJoinParticipant;
+    /**
+     * 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 ActionStartConversationRelayParams {
+    /**
+     * Custom parameters for the Conversation Relay session. Pass key-value data as
+     * `assistant.dynamic_variables` to make it available to the relay session.
+     */
+    assistant?: ActionStartConversationRelayParams.Assistant;
+    /**
+     * Use this field to add state to subsequent webhooks. 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;
+    /**
+     * Enable DTMF detection for the relay session.
+     */
+    conversation_relay_dtmf_detection?: boolean;
+    /**
+     * Conversation Relay connection settings. This object can provide `url`,
+     * `dtmf_detection`, `interruptible`, `interruptible_greeting`, and `languages`.
+     * Top-level aliases override nested values when both are present.
+     */
+    conversation_relay_settings?: ActionStartConversationRelayParams.ConversationRelaySettings;
+    /**
+     * WebSocket URL for your Conversation Relay server. Must start with `ws://` or
+     * `wss://`.
+     */
+    conversation_relay_url?: string;
+    /**
+     * Custom key-value parameters forwarded to the relay session as
+     * `assistant.dynamic_variables`. If `assistant.dynamic_variables` is also present,
+     * these values are merged in.
+     */
+    custom_parameters?: {
+        [key: string]: unknown;
+    };
+    /**
+     * Public alias for `conversation_relay_dtmf_detection`. If both are present, this
+     * value wins.
+     */
+    dtmf_detection?: boolean;
+    /**
+     * Text played when the relay session starts.
+     */
+    greeting?: string;
+    /**
+     * Controls when caller input can interrupt assistant speech. `any` allows speech
+     * or DTMF interruptions; `none` disables interruptions; `speech` allows speech
+     * only; `dtmf` allows DTMF only.
+     */
+    interruptible?: ConversationRelayInterruptible;
+    /**
+     * Controls when caller input can interrupt assistant speech. `any` allows speech
+     * or DTMF interruptions; `none` disables interruptions; `speech` allows speech
+     * only; `dtmf` allows DTMF only.
+     */
+    interruptible_greeting?: ConversationRelayInterruptible;
+    /**
+     * Settings for handling caller interruptions during Conversation Relay speech.
+     */
+    interruption_settings?: CallsAPI.ConversationRelayInterruptionSettings;
+    /**
+     * Default language for the relay session. This value is used for both
+     * text-to-speech and speech recognition.
+     */
+    language?: string;
+    /**
+     * Per-language TTS and transcription settings.
+     */
+    languages?: Array<CallsAPI.ConversationRelayLanguage>;
+    /**
+     * Structured voice provider. Must be supplied together with `structured_provider`.
+     */
+    provider?: string;
+    /**
+     * Provider-specific structured voice settings. Must be supplied together with
+     * `provider`; Telnyx sends the value as the nested provider configuration for
+     * Conversation Relay.
+     */
+    structured_provider?: {
+        [key: string]: unknown;
+    };
+    /**
+     * @deprecated Not supported for Conversation Relay start requests. Use
+     * `transcription_engine` and `transcription_engine_config` instead.
+     */
+    transcription?: {
+        [key: string]: unknown;
+    };
+    /**
+     * Engine to use for speech recognition. Legacy values `A` - `Google`, `B` -
+     * `Telnyx` are supported for backward compatibility. For Conversation Relay, use
+     * this field with `transcription_engine_config`; the `transcription` object is not
+     * supported.
+     */
+    transcription_engine?: 'Google' | 'Telnyx' | 'Deepgram' | 'Azure' | 'xAI' | 'AssemblyAI' | 'Speechmatics' | 'Soniox' | 'A' | 'B';
+    /**
+     * Engine-specific transcription settings for Conversation Relay. This accepts the
+     * same provider-specific options used by the Call Transcription Start command,
+     * such as `transcription_model`, without requiring the engine discriminator to be
+     * repeated inside this object.
+     */
+    transcription_engine_config?: {
+        [key: string]: unknown;
+    };
+    /**
+     * Text-to-speech provider. If omitted, Telnyx derives it from `voice` or
+     * `provider`.
+     */
+    tts_provider?: string;
+    /**
+     * Public alias for `conversation_relay_url`. Must start with `ws://` or `wss://`.
+     * If both are present, this value wins.
+     */
+    url?: string;
+    /**
+     * 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>`
+     * - **Inworld:** Use `Inworld.<ModelId>.<VoiceId>` (e.g., `Inworld.Mini.Loretta`,
+     *   `Inworld.Max.Oliver`, `Inworld.TTS2.Loretta`). Supported models: `Mini`,
+     *   `Max`, `TTS2`.
+     * - **xAI:** Use `xAI.<VoiceId>` (e.g., `xAI.eve`). Available voices: `eve`,
+     *   `ara`, `rex`, `sal`, `leo`.
+     */
+    voice?: string;
+    /**
+     * The settings associated with the voice selected
+     */
+    voice_settings?: ElevenLabsVoiceSettings | TelnyxVoiceSettings | AwsVoiceSettings | Shared.MinimaxVoiceSettings | Shared.AzureVoiceSettings | Shared.RimeVoiceSettings | Shared.ResembleVoiceSettings | Shared.InworldVoiceSettings | Shared.XaiVoiceSettings;
+}
+export declare namespace ActionStartConversationRelayParams {
+    /**
+     * Custom parameters for the Conversation Relay session. Pass key-value data as
+     * `assistant.dynamic_variables` to make it available to the relay session.
+     */
+    interface Assistant {
+        /**
+         * Custom key-value parameters forwarded to the Conversation Relay session.
+         */
+        dynamic_variables?: {
+            [key: string]: string;
+        };
+        [k: string]: unknown;
+    }
+    /**
+     * Conversation Relay connection settings. This object can provide `url`,
+     * `dtmf_detection`, `interruptible`, `interruptible_greeting`, and `languages`.
+     * Top-level aliases override nested values when both are present.
+     */
+    interface ConversationRelaySettings {
+        /**
+         * WebSocket URL for your Conversation Relay server. Must start with `ws://` or
+         * `wss://`.
+         */
+        url: string;
+        /**
+         * Whether to enable DTMF detection during the relay session.
+         */
+        dtmf_detection?: boolean;
+        /**
+         * Controls when caller input can interrupt assistant speech. `any` allows speech
+         * or DTMF interruptions; `none` disables interruptions; `speech` allows speech
+         * only; `dtmf` allows DTMF only.
+         */
+        interruptible?: ActionsAPI.ConversationRelayInterruptible;
+        /**
+         * Controls when caller input can interrupt assistant speech. `any` allows speech
+         * or DTMF interruptions; `none` disables interruptions; `speech` allows speech
+         * only; `dtmf` allows DTMF only.
+         */
+        interruptible_greeting?: ActionsAPI.ConversationRelayInterruptible;
+        /**
+         * Language-specific TTS and transcription settings.
+         */
+        languages?: Array<CallsAPI.ConversationRelayLanguage>;
+    }
+}
+export interface ActionStopConversationRelayParams {
+    /**
+     * Use this field to add state to subsequent webhooks. 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 declare namespace Actions {
-    export { type AIAssistantJoinParticipant as AIAssistantJoinParticipant, type AssistantMessage as AssistantMessage, type AwsVoiceSettings as AwsVoiceSettings, type CallControlCommandResult as CallControlCommandResult, type CallControlCommandResultWithConversationID as CallControlCommandResultWithConversationID, type ConversationRelayInterruptible as ConversationRelayInterruptible, type DeepgramNova2Config as DeepgramNova2Config, type DeepgramNova3Config as DeepgramNova3Config, type DeveloperMessage as DeveloperMessage, type ElevenLabsVoiceSettings as ElevenLabsVoiceSettings, type GoogleTranscriptionLanguage as GoogleTranscriptionLanguage, type InterruptionSettings as InterruptionSettings, type Loopcount as Loopcount, type StopRecordingRequest as StopRecordingRequest, type SystemMessage as SystemMessage, type TelnyxTranscriptionLanguage as TelnyxTranscriptionLanguage, type TelnyxVoiceSettings as TelnyxVoiceSettings, type ToolMessage as ToolMessage, type TranscriptionConfig as TranscriptionConfig, type TranscriptionEngineAConfig as TranscriptionEngineAConfig, type TranscriptionEngineAssemblyaiConfig as TranscriptionEngineAssemblyaiConfig, type TranscriptionEngineAzureConfig as TranscriptionEngineAzureConfig, type TranscriptionEngineBConfig as TranscriptionEngineBConfig, type TranscriptionEngineDeepgramConfig as TranscriptionEngineDeepgramConfig, type TranscriptionEngineGoogleConfig as TranscriptionEngineGoogleConfig, type TranscriptionEngineParakeetConfig as TranscriptionEngineParakeetConfig, type TranscriptionEngineSonioxConfig as TranscriptionEngineSonioxConfig, type TranscriptionEngineSpeechmaticsConfig as TranscriptionEngineSpeechmaticsConfig, type TranscriptionEngineTelnyxConfig as TranscriptionEngineTelnyxConfig, type TranscriptionEngineXaiConfig as TranscriptionEngineXaiConfig, type TranscriptionStartRequest as TranscriptionStartRequest, type UserMessage as UserMessage, type ActionAddAIAssistantMessagesResponse as ActionAddAIAssistantMessagesResponse, 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 ActionJoinAIAssistantResponse as ActionJoinAIAssistantResponse, 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 ActionStartConversationRelayResponse as ActionStartConversationRelayResponse, 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 ActionStopConversationRelayResponse as ActionStopConversationRelayResponse, 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 ActionAddAIAssistantMessagesParams as ActionAddAIAssistantMessagesParams, 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 ActionJoinAIAssistantParams as ActionJoinAIAssistantParams, 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 ActionStartConversationRelayParams as ActionStartConversationRelayParams, 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 ActionStopConversationRelayParams as ActionStopConversationRelayParams, 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, };
+    export { type AIAssistantJoinParticipant as AIAssistantJoinParticipant, type AssistantMessage as AssistantMessage, type AwsVoiceSettings as AwsVoiceSettings, type CallControlCommandResult as CallControlCommandResult, type CallControlCommandResultWithConversationID as CallControlCommandResultWithConversationID, type ConversationRelayInterruptible as ConversationRelayInterruptible, type DeepgramNova2Config as DeepgramNova2Config, type DeepgramNova3Config as DeepgramNova3Config, type DeveloperMessage as DeveloperMessage, type ElevenLabsVoiceSettings as ElevenLabsVoiceSettings, type GoogleTranscriptionLanguage as GoogleTranscriptionLanguage, type InterruptionSettings as InterruptionSettings, type Loopcount as Loopcount, type StopRecordingRequest as StopRecordingRequest, type SystemMessage as SystemMessage, type TelnyxTranscriptionLanguage as TelnyxTranscriptionLanguage, type TelnyxVoiceSettings as TelnyxVoiceSettings, type ToolMessage as ToolMessage, type TranscriptionConfig as TranscriptionConfig, type TranscriptionEngineAConfig as TranscriptionEngineAConfig, type TranscriptionEngineAssemblyaiConfig as TranscriptionEngineAssemblyaiConfig, type TranscriptionEngineAzureConfig as TranscriptionEngineAzureConfig, type TranscriptionEngineBConfig as TranscriptionEngineBConfig, type TranscriptionEngineDeepgramConfig as TranscriptionEngineDeepgramConfig, type TranscriptionEngineGoogleConfig as TranscriptionEngineGoogleConfig, type TranscriptionEngineParakeetConfig as TranscriptionEngineParakeetConfig, type TranscriptionEngineSonioxConfig as TranscriptionEngineSonioxConfig, type TranscriptionEngineSpeechmaticsConfig as TranscriptionEngineSpeechmaticsConfig, type TranscriptionEngineTelnyxConfig as TranscriptionEngineTelnyxConfig, type TranscriptionEngineXaiConfig as TranscriptionEngineXaiConfig, type TranscriptionStartRequest as TranscriptionStartRequest, type UserMessage as UserMessage, type ActionAddAIAssistantMessagesResponse as ActionAddAIAssistantMessagesResponse, 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 ActionJoinAIAssistantResponse as ActionJoinAIAssistantResponse, 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 ActionStartConversationRelayResponse as ActionStartConversationRelayResponse, 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 ActionStopConversationRelayResponse as ActionStopConversationRelayResponse, 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 ActionStartAIAssistantParams as ActionStartAIAssistantParams, type ActionStopAIAssistantParams as ActionStopAIAssistantParams, type ActionAnswerParams as ActionAnswerParams, type ActionBridgeParams as ActionBridgeParams, type ActionUpdateClientStateParams as ActionUpdateClientStateParams, type ActionEnqueueParams as ActionEnqueueParams, type ActionStartForkingParams as ActionStartForkingParams, type ActionStopForkingParams as ActionStopForkingParams, type ActionGatherParams as ActionGatherParams, type ActionStopGatherParams as ActionStopGatherParams, type ActionGatherUsingAIParams as ActionGatherUsingAIParams, type ActionGatherUsingAudioParams as ActionGatherUsingAudioParams, type ActionGatherUsingSpeakParams as ActionGatherUsingSpeakParams, type ActionHangupParams as ActionHangupParams, type ActionLeaveQueueParams as ActionLeaveQueueParams, type ActionStartPlaybackParams as ActionStartPlaybackParams, type ActionStopPlaybackParams as ActionStopPlaybackParams, type ActionPauseRecordingParams as ActionPauseRecordingParams, type ActionResumeRecordingParams as ActionResumeRecordingParams, type ActionStartRecordingParams as ActionStartRecordingParams, type ActionStopRecordingParams as ActionStopRecordingParams, type ActionReferParams as ActionReferParams, type ActionRejectParams as ActionRejectParams, type ActionSendDtmfParams as ActionSendDtmfParams, type ActionSendSipInfoParams as ActionSendSipInfoParams, type ActionStartSiprecParams as ActionStartSiprecParams, type ActionStopSiprecParams as ActionStopSiprecParams, type ActionSpeakParams as ActionSpeakParams, type ActionStartStreamingParams as ActionStartStreamingParams, type ActionStopStreamingParams as ActionStopStreamingParams, type ActionStartNoiseSuppressionParams as ActionStartNoiseSuppressionParams, type ActionStopNoiseSuppressionParams as ActionStopNoiseSuppressionParams, type ActionSwitchSupervisorRoleParams as ActionSwitchSupervisorRoleParams, type ActionStartTranscriptionParams as ActionStartTranscriptionParams, type ActionStopTranscriptionParams as ActionStopTranscriptionParams, type ActionTransferParams as ActionTransferParams, type ActionAddAIAssistantMessagesParams as ActionAddAIAssistantMessagesParams, type ActionJoinAIAssistantParams as ActionJoinAIAssistantParams, type ActionStartConversationRelayParams as ActionStartConversationRelayParams, type ActionStopConversationRelayParams as ActionStopConversationRelayParams, };
 }
 //# sourceMappingURL=actions.d.mts.map