telnyx 7.3.0 → 7.6.0

package/src/resources/calls/actions.ts

@@ -14,25 +14,45 @@ import { path } from '../../internal/utils/path';
  */
 export 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(
+  startAIAssistant(
     callControlID: string,
-    body: ActionAddAIAssistantMessagesParams,
+    body: ActionStartAIAssistantParams,
     options?: RequestOptions,
-  ): APIPromise<ActionAddAIAssistantMessagesResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/ai_assistant_add_messages`, {
-      body,
-      ...options,
-    });
+  ): APIPromise<ActionStartAIAssistantResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/ai_assistant_start`, { body, ...options });
+  }
+
+  /**
+   * Stop an AI assistant on the call.
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.stopAIAssistant(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  stopAIAssistant(
+    callControlID: string,
+    body: ActionStopAIAssistantParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStopAIAssistantResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/ai_assistant_stop`, { body, ...options });
   }
 
   /**
@@ -98,6 +118,26 @@ export class Actions extends APIResource {
     });
   }
 
+  /**
+   * Updates client state
+   *
+   * @example
+   * ```ts
+   * const response =
+   *   await client.calls.actions.updateClientState(
+   *     'call_control_id',
+   *     { client_state: 'aGF2ZSBhIG5pY2UgZGF5ID1d' },
+   *   );
+   * ```
+   */
+  updateClientState(
+    callControlID: string,
+    body: ActionUpdateClientStateParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionUpdateClientStateResponse> {
+    return this._client.put(path`/calls/${callControlID}/actions/client_state_update`, { body, ...options });
+  }
+
   /**
    * Put the call in a queue.
    *
@@ -117,6 +157,55 @@ export class Actions extends APIResource {
     return this._client.post(path`/calls/${callControlID}/actions/enqueue`, { body, ...options });
   }
 
+  /**
+   * Call forking allows you to stream the media from a call to a specific target in
+   * realtime. This stream can be used to enable realtime audio analysis to support a
+   * variety of use cases, including fraud detection, or the creation of AI-generated
+   * audio responses. Requests must specify either the `target` attribute or the `rx`
+   * and `tx` attributes.
+   *
+   * **Expected Webhooks:**
+   *
+   * - `call.fork.started`
+   * - `call.fork.stopped`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.startForking(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  startForking(
+    callControlID: string,
+    body: ActionStartForkingParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStartForkingResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/fork_start`, { body, ...options });
+  }
+
+  /**
+   * Stop forking a call.
+   *
+   * **Expected Webhooks:**
+   *
+   * - `call.fork.stopped`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.stopForking(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  stopForking(
+    callControlID: string,
+    body: ActionStopForkingParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStopForkingResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/fork_stop`, { body, ...options });
+  }
+
   /**
    * Gather DTMF signals to build interactive menus.
    *
@@ -143,6 +232,28 @@ export class Actions extends APIResource {
     return this._client.post(path`/calls/${callControlID}/actions/gather`, { body, ...options });
   }
 
+  /**
+   * 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> {
+    return this._client.post(path`/calls/${callControlID}/actions/gather_stop`, { body, ...options });
+  }
+
   /**
    * Gather parameters defined in the request payload using a voice assistant.
    *
@@ -265,29 +376,6 @@ export class Actions extends APIResource {
     return this._client.post(path`/calls/${callControlID}/actions/hangup`, { body, ...options });
   }
 
-  /**
-   * 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.joinAIAssistant(
-   *   'call_control_id',
-   *   {
-   *     conversation_id: 'v3:abc123',
-   *     participant: { id: 'v3:abc123def456', role: 'user' },
-   *   },
-   * );
-   * ```
-   */
-  joinAIAssistant(
-    callControlID: string,
-    body: ActionJoinAIAssistantParams,
-    options?: RequestOptions,
-  ): APIPromise<ActionJoinAIAssistantResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/ai_assistant_join`, { body, ...options });
-  }
-
   /**
    * Removes the call from a queue.
    *
@@ -307,76 +395,77 @@ export class Actions extends APIResource {
   }
 
   /**
-   * Pause recording the call. Recording can be resumed via Resume recording command.
+   * 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:**
    *
-   * There are no webhooks associated with this command.
+   * - `call.playback.started`
+   * - `call.playback.ended`
    *
    * @example
    * ```ts
-   * const response = await client.calls.actions.pauseRecording(
+   * const response = await client.calls.actions.startPlayback(
    *   'call_control_id',
    * );
    * ```
    */
-  pauseRecording(
+  startPlayback(
     callControlID: string,
-    body: ActionPauseRecordingParams,
+    body: ActionStartPlaybackParams,
     options?: RequestOptions,
-  ): APIPromise<ActionPauseRecordingResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/record_pause`, { body, ...options });
+  ): APIPromise<ActionStartPlaybackResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/playback_start`, { body, ...options });
   }
 
   /**
-   * Initiate a SIP Refer on a Call Control call. You can initiate a SIP Refer at any
-   * point in the duration of a call.
+   * Stop audio being played on the call.
    *
    * **Expected Webhooks:**
    *
-   * - `call.refer.started`
-   * - `call.refer.completed`
-   * - `call.refer.failed`
+   * - `call.playback.ended` or `call.speak.ended`
    *
    * @example
    * ```ts
-   * const response = await client.calls.actions.refer(
+   * const response = await client.calls.actions.stopPlayback(
    *   'call_control_id',
-   *   {
-   *     sip_address: 'sip:username@sip.non-telnyx-address.com',
-   *   },
    * );
    * ```
    */
-  refer(
+  stopPlayback(
     callControlID: string,
-    body: ActionReferParams,
+    body: ActionStopPlaybackParams,
     options?: RequestOptions,
-  ): APIPromise<ActionReferResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/refer`, { body, ...options });
+  ): APIPromise<ActionStopPlaybackResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/playback_stop`, { body, ...options });
   }
 
   /**
-   * Reject an incoming call.
+   * Pause recording the call. Recording can be resumed via Resume recording command.
    *
    * **Expected Webhooks:**
    *
-   * - `call.hangup`
+   * There are no webhooks associated with this command.
    *
    * @example
    * ```ts
-   * const response = await client.calls.actions.reject(
+   * const response = await client.calls.actions.pauseRecording(
    *   'call_control_id',
-   *   { cause: 'USER_BUSY' },
    * );
    * ```
    */
-  reject(
+  pauseRecording(
     callControlID: string,
-    body: ActionRejectParams,
+    body: ActionPauseRecordingParams,
     options?: RequestOptions,
-  ): APIPromise<ActionRejectResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/reject`, { body, ...options });
+  ): APIPromise<ActionPauseRecordingResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/record_pause`, { body, ...options });
   }
 
   /**
@@ -402,261 +491,224 @@ export class Actions extends APIResource {
   }
 
   /**
-   * Sends DTMF tones from this leg. DTMF tones will be heard by the other end of the
-   * call.
+   * Start recording the call. Recording will stop on call hang-up, or can be
+   * initiated via the Stop Recording command.
    *
    * **Expected Webhooks:**
    *
-   * There are no webhooks associated with this command.
+   * - `call.recording.saved`
+   * - `call.recording.transcription.saved`
+   * - `call.recording.error`
    *
    * @example
    * ```ts
-   * const response = await client.calls.actions.sendDtmf(
+   * const response = await client.calls.actions.startRecording(
    *   'call_control_id',
-   *   { digits: '1www2WABCDw9' },
+   *   { channels: 'single', format: 'wav' },
    * );
    * ```
    */
-  sendDtmf(
+  startRecording(
     callControlID: string,
-    body: ActionSendDtmfParams,
+    body: ActionStartRecordingParams,
     options?: RequestOptions,
-  ): APIPromise<ActionSendDtmfResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/send_dtmf`, { body, ...options });
+  ): APIPromise<ActionStartRecordingResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/record_start`, { body, ...options });
   }
 
   /**
-   * Sends SIP info from this leg.
+   * Stop recording the call.
    *
    * **Expected Webhooks:**
    *
-   * - `call.sip_info.received` (to be received on the target call leg)
+   * - `call.recording.saved`
    *
    * @example
    * ```ts
-   * const response = await client.calls.actions.sendSipInfo(
+   * const response = await client.calls.actions.stopRecording(
    *   'call_control_id',
-   *   {
-   *     body: '{"key": "value", "numValue": 100}',
-   *     content_type: 'application/json',
-   *   },
    * );
    * ```
    */
-  sendSipInfo(
+  stopRecording(
     callControlID: string,
-    body: ActionSendSipInfoParams,
+    body: ActionStopRecordingParams,
     options?: RequestOptions,
-  ): APIPromise<ActionSendSipInfoResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/send_sip_info`, { body, ...options });
+  ): APIPromise<ActionStopRecordingResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/record_stop`, { body, ...options });
   }
 
   /**
-   * Convert text to speech and play it back on the call. If multiple speak text
-   * commands are issued consecutively, the audio files will be placed in a queue
-   * awaiting playback.
+   * Initiate a SIP Refer on a Call Control call. You can initiate a SIP Refer at any
+   * point in the duration of a call.
    *
    * **Expected Webhooks:**
    *
-   * - `call.speak.started`
-   * - `call.speak.ended`
-   *
-   * @example
+   * - `call.refer.started`
+   * - `call.refer.completed`
+   * - `call.refer.failed`
+   *
+   * @example
    * ```ts
-   * const response = await client.calls.actions.speak(
+   * const response = await client.calls.actions.refer(
    *   'call_control_id',
-   *   { payload: 'Say this on the call', voice: 'female' },
+   *   {
+   *     sip_address: 'sip:username@sip.non-telnyx-address.com',
+   *   },
    * );
    * ```
    */
-  speak(
+  refer(
     callControlID: string,
-    body: ActionSpeakParams,
+    body: ActionReferParams,
     options?: RequestOptions,
-  ): APIPromise<ActionSpeakResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/speak`, { body, ...options });
+  ): APIPromise<ActionReferResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/refer`, { body, ...options });
   }
 
   /**
-   * Start an AI assistant on the call.
+   * Reject an incoming call.
    *
    * **Expected Webhooks:**
    *
-   * - `call.conversation.ended`
-   * - `call.conversation_insights.generated`
+   * - `call.hangup`
    *
    * @example
    * ```ts
-   * const response =
-   *   await client.calls.actions.startAIAssistant(
-   *     'call_control_id',
-   *   );
+   * const response = await client.calls.actions.reject(
+   *   'call_control_id',
+   *   { cause: 'USER_BUSY' },
+   * );
    * ```
    */
-  startAIAssistant(
+  reject(
     callControlID: string,
-    body: ActionStartAIAssistantParams,
+    body: ActionRejectParams,
     options?: RequestOptions,
-  ): APIPromise<ActionStartAIAssistantResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/ai_assistant_start`, { body, ...options });
+  ): APIPromise<ActionRejectResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/reject`, { body, ...options });
   }
 
   /**
-   * 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.
+   * Sends DTMF tones from this leg. DTMF tones will be heard by the other end of the
+   * call.
    *
    * **Expected Webhooks:**
    *
-   * - `call.conversation.ended` - Sent when the Conversation Relay session ends. If
-   *   the customer WebSocket disconnects, the webhook payload `reason` is
-   *   `customer_disconnect`.
+   * There are no webhooks associated with this command.
    *
    * @example
    * ```ts
-   * const response =
-   *   await client.calls.actions.startConversationRelay(
-   *     'call_control_id',
-   *   );
+   * const response = await client.calls.actions.sendDtmf(
+   *   'call_control_id',
+   *   { digits: '1www2WABCDw9' },
+   * );
    * ```
    */
-  startConversationRelay(
+  sendDtmf(
     callControlID: string,
-    body: ActionStartConversationRelayParams,
+    body: ActionSendDtmfParams,
     options?: RequestOptions,
-  ): APIPromise<ActionStartConversationRelayResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/conversation_relay_start`, {
-      body,
-      ...options,
-    });
+  ): APIPromise<ActionSendDtmfResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/send_dtmf`, { body, ...options });
   }
 
   /**
-   * Call forking allows you to stream the media from a call to a specific target in
-   * realtime. This stream can be used to enable realtime audio analysis to support a
-   * variety of use cases, including fraud detection, or the creation of AI-generated
-   * audio responses. Requests must specify either the `target` attribute or the `rx`
-   * and `tx` attributes.
+   * Sends SIP info from this leg.
    *
    * **Expected Webhooks:**
    *
-   * - `call.fork.started`
-   * - `call.fork.stopped`
+   * - `call.sip_info.received` (to be received on the target call leg)
    *
    * @example
    * ```ts
-   * const response = await client.calls.actions.startForking(
+   * const response = await client.calls.actions.sendSipInfo(
    *   'call_control_id',
+   *   {
+   *     body: '{"key": "value", "numValue": 100}',
+   *     content_type: 'application/json',
+   *   },
    * );
    * ```
    */
-  startForking(
-    callControlID: string,
-    body: ActionStartForkingParams,
-    options?: RequestOptions,
-  ): APIPromise<ActionStartForkingResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/fork_start`, { body, ...options });
-  }
-
-  /**
-   * Noise Suppression Start (BETA)
-   *
-   * @example
-   * ```ts
-   * const response =
-   *   await client.calls.actions.startNoiseSuppression(
-   *     'call_control_id',
-   *   );
-   * ```
-   */
-  startNoiseSuppression(
+  sendSipInfo(
     callControlID: string,
-    body: ActionStartNoiseSuppressionParams,
+    body: ActionSendSipInfoParams,
     options?: RequestOptions,
-  ): APIPromise<ActionStartNoiseSuppressionResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/suppression_start`, { body, ...options });
+  ): APIPromise<ActionSendSipInfoResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/send_sip_info`, { body, ...options });
   }
 
   /**
-   * Play an audio file on the call. If multiple play audio commands are issued
-   * consecutively, the audio files will be placed in a queue awaiting playback.
-   *
-   * _Notes:_
-   *
-   * - When `overlay` is enabled, `target_legs` is limited to `self`.
-   * - A customer cannot Play Audio with `overlay=true` unless there is a Play Audio
-   *   with `overlay=false` actively playing.
+   * Start siprec session to configured in SIPREC connector SRS.
    *
    * **Expected Webhooks:**
    *
-   * - `call.playback.started`
-   * - `call.playback.ended`
+   * - `siprec.started`
+   * - `siprec.stopped`
+   * - `siprec.failed`
    *
    * @example
    * ```ts
-   * const response = await client.calls.actions.startPlayback(
+   * const response = await client.calls.actions.startSiprec(
    *   'call_control_id',
    * );
    * ```
    */
-  startPlayback(
+  startSiprec(
     callControlID: string,
-    body: ActionStartPlaybackParams,
+    body: ActionStartSiprecParams,
     options?: RequestOptions,
-  ): APIPromise<ActionStartPlaybackResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/playback_start`, { body, ...options });
+  ): APIPromise<ActionStartSiprecResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/siprec_start`, { body, ...options });
   }
 
   /**
-   * Start recording the call. Recording will stop on call hang-up, or can be
-   * initiated via the Stop Recording command.
+   * Stop SIPREC session.
    *
    * **Expected Webhooks:**
    *
-   * - `call.recording.saved`
-   * - `call.recording.transcription.saved`
-   * - `call.recording.error`
+   * - `siprec.stopped`
    *
    * @example
    * ```ts
-   * const response = await client.calls.actions.startRecording(
+   * const response = await client.calls.actions.stopSiprec(
    *   'call_control_id',
-   *   { channels: 'single', format: 'wav' },
    * );
    * ```
    */
-  startRecording(
+  stopSiprec(
     callControlID: string,
-    body: ActionStartRecordingParams,
+    body: ActionStopSiprecParams,
     options?: RequestOptions,
-  ): APIPromise<ActionStartRecordingResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/record_start`, { body, ...options });
+  ): APIPromise<ActionStopSiprecResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/siprec_stop`, { body, ...options });
   }
 
   /**
-   * Start siprec session to configured in SIPREC connector SRS.
+   * Convert text to speech and play it back on the call. If multiple speak text
+   * commands are issued consecutively, the audio files will be placed in a queue
+   * awaiting playback.
    *
    * **Expected Webhooks:**
    *
-   * - `siprec.started`
-   * - `siprec.stopped`
-   * - `siprec.failed`
+   * - `call.speak.started`
+   * - `call.speak.ended`
    *
    * @example
    * ```ts
-   * const response = await client.calls.actions.startSiprec(
+   * const response = await client.calls.actions.speak(
    *   'call_control_id',
+   *   { payload: 'Say this on the call', voice: 'female' },
    * );
    * ```
    */
-  startSiprec(
+  speak(
     callControlID: string,
-    body: ActionStartSiprecParams,
+    body: ActionSpeakParams,
     options?: RequestOptions,
-  ): APIPromise<ActionStartSiprecResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/siprec_start`, { body, ...options });
+  ): APIPromise<ActionSpeakResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/speak`, { body, ...options });
   }
 
   /**
@@ -683,319 +735,267 @@ export class Actions extends APIResource {
   }
 
   /**
-   * Start real-time transcription. Transcription will stop on call hang-up, or can
-   * be initiated via the Transcription stop command.
+   * Stop streaming a call to a WebSocket.
    *
    * **Expected Webhooks:**
    *
-   * - `call.transcription`
+   * - `streaming.stopped`
+   *
+   * @example
+   * ```ts
+   * const response = await client.calls.actions.stopStreaming(
+   *   'call_control_id',
+   * );
+   * ```
+   */
+  stopStreaming(
+    callControlID: string,
+    body: ActionStopStreamingParams,
+    options?: RequestOptions,
+  ): APIPromise<ActionStopStreamingResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/streaming_stop`, { body, ...options });
+  }
+
+  /**
+   * Noise Suppression Start (BETA)
    *
    * @example
    * ```ts
    * const response =
-   *   await client.calls.actions.startTranscription(
+   *   await client.calls.actions.startNoiseSuppression(
    *     'call_control_id',
    *   );
    * ```
    */
-  startTranscription(
+  startNoiseSuppression(
     callControlID: string,
-    body: ActionStartTranscriptionParams,
+    body: ActionStartNoiseSuppressionParams,
     options?: RequestOptions,
-  ): APIPromise<ActionStartTranscriptionResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/transcription_start`, { body, ...options });
+  ): APIPromise<ActionStartNoiseSuppressionResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/suppression_start`, { body, ...options });
   }
 
   /**
-   * Stop an AI assistant on the call.
+   * Noise Suppression Stop (BETA)
    *
    * @example
    * ```ts
-   * const response = await client.calls.actions.stopAIAssistant(
-   *   'call_control_id',
-   * );
+   * const response =
+   *   await client.calls.actions.stopNoiseSuppression(
+   *     'call_control_id',
+   *   );
    * ```
    */
-  stopAIAssistant(
+  stopNoiseSuppression(
     callControlID: string,
-    body: ActionStopAIAssistantParams,
+    body: ActionStopNoiseSuppressionParams,
     options?: RequestOptions,
-  ): APIPromise<ActionStopAIAssistantResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/ai_assistant_stop`, { body, ...options });
+  ): APIPromise<ActionStopNoiseSuppressionResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/suppression_stop`, { body, ...options });
   }
 
   /**
-   * Stop the active Conversation Relay session on a call.
+   * 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.stopConversationRelay(
+   *   await client.calls.actions.switchSupervisorRole(
    *     'call_control_id',
+   *     { role: 'barge' },
    *   );
    * ```
    */
-  stopConversationRelay(
+  switchSupervisorRole(
     callControlID: string,
-    body: ActionStopConversationRelayParams,
+    body: ActionSwitchSupervisorRoleParams,
     options?: RequestOptions,
-  ): APIPromise<ActionStopConversationRelayResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/conversation_relay_stop`, {
+  ): APIPromise<ActionSwitchSupervisorRoleResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/switch_supervisor_role`, {
       body,
       ...options,
     });
   }
 
   /**
-   * Stop forking a call.
+   * Start real-time transcription. Transcription will stop on call hang-up, or can
+   * be initiated via the Transcription stop command.
    *
    * **Expected Webhooks:**
    *
-   * - `call.fork.stopped`
+   * - `call.transcription`
    *
    * @example
    * ```ts
-   * const response = await client.calls.actions.stopForking(
-   *   'call_control_id',
-   * );
+   * const response =
+   *   await client.calls.actions.startTranscription(
+   *     'call_control_id',
+   *   );
    * ```
    */
-  stopForking(
+  startTranscription(
     callControlID: string,
-    body: ActionStopForkingParams,
+    body: ActionStartTranscriptionParams,
     options?: RequestOptions,
-  ): APIPromise<ActionStopForkingResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/fork_stop`, { body, ...options });
-  }
-
-  /**
-   * Stop current gather.
-   *
-   * **Expected Webhooks:**
-   *
-   * - `call.gather.ended`
-   *
-   * @example
-   * ```ts
-   * const response = await client.calls.actions.stopGather(
-   *   'call_control_id',
-   * );
-   * ```
-   */
-  stopGather(
-    callControlID: string,
-    body: ActionStopGatherParams,
-    options?: RequestOptions,
-  ): APIPromise<ActionStopGatherResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/gather_stop`, { body, ...options });
+  ): APIPromise<ActionStartTranscriptionResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/transcription_start`, { body, ...options });
   }
 
   /**
-   * Noise Suppression Stop (BETA)
+   * Stop real-time transcription.
    *
    * @example
    * ```ts
    * const response =
-   *   await client.calls.actions.stopNoiseSuppression(
+   *   await client.calls.actions.stopTranscription(
    *     'call_control_id',
    *   );
    * ```
    */
-  stopNoiseSuppression(
+  stopTranscription(
     callControlID: string,
-    body: ActionStopNoiseSuppressionParams,
+    body: ActionStopTranscriptionParams,
     options?: RequestOptions,
-  ): APIPromise<ActionStopNoiseSuppressionResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/suppression_stop`, { body, ...options });
+  ): APIPromise<ActionStopTranscriptionResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/transcription_stop`, { body, ...options });
   }
 
   /**
-   * Stop audio being played on the call.
+   * Transfer a call to a new destination. If the transfer is unsuccessful, a
+   * `call.hangup` webhook for the other call (Leg B) will be sent indicating that
+   * the transfer could not be completed. The original call will remain active and
+   * may be issued additional commands, potentially transfering the call to an
+   * alternate destination.
    *
    * **Expected Webhooks:**
    *
-   * - `call.playback.ended` or `call.speak.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.stopPlayback(
+   * const response = await client.calls.actions.transfer(
    *   'call_control_id',
+   *   {
+   *     to: '+18005550100 or sip:username@sip.telnyx.com;secure=srtp',
+   *   },
    * );
    * ```
    */
-  stopPlayback(
+  transfer(
     callControlID: string,
-    body: ActionStopPlaybackParams,
+    body: ActionTransferParams,
     options?: RequestOptions,
-  ): APIPromise<ActionStopPlaybackResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/playback_stop`, { body, ...options });
+  ): APIPromise<ActionTransferResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/transfer`, { body, ...options });
   }
 
   /**
-   * Stop recording the call.
-   *
-   * **Expected Webhooks:**
-   *
-   * - `call.recording.saved`
+   * Add messages to the conversation started by an AI assistant on the call.
    *
    * @example
    * ```ts
-   * const response = await client.calls.actions.stopRecording(
-   *   'call_control_id',
-   * );
+   * const response =
+   *   await client.calls.actions.addAIAssistantMessages(
+   *     'call_control_id',
+   *   );
    * ```
    */
-  stopRecording(
+  addAIAssistantMessages(
     callControlID: string,
-    body: ActionStopRecordingParams,
+    body: ActionAddAIAssistantMessagesParams,
     options?: RequestOptions,
-  ): APIPromise<ActionStopRecordingResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/record_stop`, { body, ...options });
+  ): APIPromise<ActionAddAIAssistantMessagesResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/ai_assistant_add_messages`, {
+      body,
+      ...options,
+    });
   }
 
   /**
-   * 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(
+  joinAIAssistant(
     callControlID: string,
-    body: ActionStopSiprecParams,
+    body: ActionJoinAIAssistantParams,
     options?: RequestOptions,
-  ): APIPromise<ActionStopSiprecResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/siprec_stop`, { body, ...options });
+  ): APIPromise<ActionJoinAIAssistantResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/ai_assistant_join`, { body, ...options });
   }
 
   /**
-   * 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> {
-    return this._client.post(path`/calls/${callControlID}/actions/streaming_stop`, { body, ...options });
-  }
-
-  /**
-   * Stop real-time transcription.
-   *
-   * @example
-   * ```ts
-   * const response =
-   *   await client.calls.actions.stopTranscription(
-   *     'call_control_id',
-   *   );
-   * ```
-   */
-  stopTranscription(
-    callControlID: string,
-    body: ActionStopTranscriptionParams,
-    options?: RequestOptions,
-  ): APIPromise<ActionStopTranscriptionResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/transcription_stop`, { body, ...options });
-  }
-
-  /**
-   * Switch the supervisor role for a bridged call. This allows switching between
-   * different supervisor modes during an active call
+   * - `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.switchSupervisorRole(
+   *   await client.calls.actions.startConversationRelay(
    *     'call_control_id',
-   *     { role: 'barge' },
    *   );
    * ```
    */
-  switchSupervisorRole(
+  startConversationRelay(
     callControlID: string,
-    body: ActionSwitchSupervisorRoleParams,
+    body: ActionStartConversationRelayParams,
     options?: RequestOptions,
-  ): APIPromise<ActionSwitchSupervisorRoleResponse> {
-    return this._client.post(path`/calls/${callControlID}/actions/switch_supervisor_role`, {
+  ): APIPromise<ActionStartConversationRelayResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/conversation_relay_start`, {
       body,
       ...options,
     });
   }
 
   /**
-   * Transfer a call to a new destination. If the transfer is unsuccessful, a
-   * `call.hangup` webhook for the other call (Leg B) will be sent indicating that
-   * the transfer could not be completed. The original call will remain active and
-   * may be issued additional commands, potentially transfering the call to an
-   * alternate destination.
-   *
-   * **Expected Webhooks:**
-   *
-   * - `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> {
-    return this._client.post(path`/calls/${callControlID}/actions/transfer`, { body, ...options });
-  }
-
-  /**
-   * Updates client state
+   * Stop the active Conversation Relay session on a call.
    *
    * @example
    * ```ts
    * const response =
-   *   await client.calls.actions.updateClientState(
+   *   await client.calls.actions.stopConversationRelay(
    *     'call_control_id',
-   *     { client_state: 'aGF2ZSBhIG5pY2UgZGF5ID1d' },
    *   );
    * ```
    */
-  updateClientState(
+  stopConversationRelay(
     callControlID: string,
-    body: ActionUpdateClientStateParams,
+    body: ActionStopConversationRelayParams,
     options?: RequestOptions,
-  ): APIPromise<ActionUpdateClientStateResponse> {
-    return this._client.put(path`/calls/${callControlID}/actions/client_state_update`, { body, ...options });
+  ): APIPromise<ActionStopConversationRelayResponse> {
+    return this._client.post(path`/calls/${callControlID}/actions/conversation_relay_stop`, {
+      body,
+      ...options,
+    });
   }
 }
 
@@ -2334,7 +2334,14 @@ 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.
@@ -2348,28 +2355,117 @@ 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;
 
-export interface ActionAnswerParams {
   /**
-   * AI Assistant configuration. All fields except `id` are optional — the
-   * assistant's stored configuration will be used as fallback for any omitted
-   * fields.
+   * Settings for handling user interruptions during assistant speech
    */
-  assistant?: CallsAPI.CallAssistantRequest;
+  interruption_settings?: InterruptionSettings;
 
   /**
-   * Use this field to set the Billing Group ID for the call. Must be a valid and
-   * existing Billing Group ID.
+   * 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.
    */
-  billing_group_id?: string;
+  message_history?: Array<UserMessage | AssistantMessage | ToolMessage | SystemMessage | DeveloperMessage>;
 
   /**
-   * Use this field to add state to every subsequent webhook. It must be a valid
-   * Base-64 encoded string.
+   * 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 {
+  /**
+   * 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 set the Billing Group ID for the call. Must be a valid and
+   * existing Billing Group ID.
+   */
+  billing_group_id?: string;
+
+  /**
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
    */
   client_state?: string;
 
@@ -2750,6 +2846,14 @@ 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
@@ -2788,6 +2892,59 @@ 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
@@ -2845,6 +3002,20 @@ 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
@@ -3266,14 +3437,7 @@ export interface ActionHangupParams {
   custom_headers?: Array<CallsAPI.CustomSipHeader>;
 }
 
-export interface ActionJoinAIAssistantParams {
-  /**
-   * The ID of the AI assistant conversation to join.
-   */
-  conversation_id: string;
-
-  participant: AIAssistantJoinParticipant;
-
+export interface ActionLeaveQueueParams {
   /**
    * Use this field to add state to every subsequent webhook. It must be a valid
    * Base-64 encoded string.
@@ -3287,21 +3451,25 @@ export interface ActionJoinAIAssistantParams {
   command_id?: string;
 }
 
-export interface ActionLeaveQueueParams {
+export interface ActionStartPlaybackParams {
   /**
-   * Use this field to add state to every subsequent webhook. It must be a valid
-   * Base-64 encoded string.
+   * Specifies the type of audio provided in `audio_url` or `playback_content`.
    */
-  client_state?: string;
+  audio_type?: 'mp3' | 'wav';
 
   /**
-   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
-   * the same `command_id` for the same `call_control_id`.
+   * The URL of a file to be played back on the call. The URL can point to either a
+   * WAV or MP3 file. media_name and audio_url cannot be used together in one
+   * request.
    */
-  command_id?: string;
-}
+  audio_url?: string;
+
+  /**
+   * Caches the audio file. Useful when playing the same audio file multiple times
+   * during the call.
+   */
+  cache_audio?: boolean;
 
-export interface ActionPauseRecordingParams {
   /**
    * Use this field to add state to every subsequent webhook. It must be a valid
    * Base-64 encoded string.
@@ -3315,72 +3483,74 @@ export interface ActionPauseRecordingParams {
   command_id?: string;
 
   /**
-   * Uniquely identifies the resource.
+   * The number of times the audio file should be played. If supplied, the value must
+   * be an integer between 1 and 100, or the special string `infinity` for an endless
+   * loop.
    */
-  recording_id?: string;
-}
+  loop?: Loopcount;
 
-export interface ActionReferParams {
   /**
-   * The SIP URI to which the call will be referred to.
+   * The media_name of a file to be played back on the call. The media_name must
+   * point to a file previously uploaded to api.telnyx.com/v2/media by the same
+   * user/organization. The file must either be a WAV or MP3 file.
    */
-  sip_address: string;
+  media_name?: string;
 
   /**
-   * Use this field to add state to every subsequent webhook. It must be a valid
-   * Base-64 encoded string.
+   * When enabled, audio will be mixed on top of any other audio that is actively
+   * being played back. Note that `overlay: true` will only work if there is another
+   * audio file already being played on the call.
    */
-  client_state?: string;
+  overlay?: boolean;
 
   /**
-   * 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.
+   * Allows a user to provide base64 encoded mp3 or wav. Note: when using this
+   * parameter, `media_url` and `media_name` in the `playback_started` and
+   * `playback_ended` webhooks will be empty
    */
-  command_id?: string;
+  playback_content?: string;
 
   /**
-   * Custom headers to be added to the SIP INVITE.
+   * 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.
    */
-  custom_headers?: Array<CallsAPI.CustomSipHeader>;
+  stop?: string;
 
   /**
-   * SIP Authentication password used for SIP challenges.
+   * Specifies the leg or legs on which audio will be played. If supplied, the value
+   * must be either `self`, `opposite` or `both`.
    */
-  sip_auth_password?: string;
+  target_legs?: string;
+}
 
+export interface ActionStopPlaybackParams {
   /**
-   * SIP Authentication username used for SIP challenges.
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
    */
-  sip_auth_username?: string;
+  client_state?: string;
 
   /**
-   * SIP headers to be added to the request. Currently only User-to-User header is
-   * supported.
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
    */
-  sip_headers?: Array<CallsAPI.SipHeader>;
-}
+  command_id?: string;
 
-export interface ActionRejectParams {
   /**
-   * Cause for call rejection.
+   * When enabled, it stops the audio being played in the overlay queue.
    */
-  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;
+  overlay?: boolean;
 
   /**
-   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
-   * the same `command_id` for the same `call_control_id`.
+   * 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.
    */
-  command_id?: string;
+  stop?: string;
 }
 
-export interface ActionResumeRecordingParams {
+export interface ActionPauseRecordingParams {
   /**
    * Use this field to add state to every subsequent webhook. It must be a valid
    * Base-64 encoded string.
@@ -3399,13 +3569,7 @@ export interface ActionResumeRecordingParams {
   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;
-
+export interface ActionResumeRecordingParams {
   /**
    * Use this field to add state to every subsequent webhook. It must be a valid
    * Base-64 encoded string.
@@ -3419,23 +3583,23 @@ export interface ActionSendDtmfParams {
   command_id?: string;
 
   /**
-   * Specifies for how many milliseconds each digit will be played in the audio
-   * stream. Ranges from 100 to 500ms
+   * Uniquely identifies the resource.
    */
-  duration_millis?: number;
+  recording_id?: string;
 }
 
-export interface ActionSendSipInfoParams {
+export interface ActionStartRecordingParams {
   /**
-   * Content of the SIP INFO
+   * When `dual`, final audio file will be stereo recorded with the first leg on
+   * channel A, and the rest on channel B.
    */
-  body: string;
+  channels: 'single' | 'dual';
 
   /**
-   * Content type of the INFO body. Must be MIME type compliant. There is a 1,400
-   * bytes limit
+   * The audio file format used when storing the call recording. Can be either `mp3`
+   * or `wav`.
    */
-  content_type: string;
+  format: 'wav' | 'mp3';
 
   /**
    * Use this field to add state to every subsequent webhook. It must be a valid
@@ -3448,165 +3612,333 @@ export interface ActionSendSipInfoParams {
    * 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.
+   * The custom recording file name to be used instead of the default `call_leg_id`.
+   * Telnyx will still add a Unix timestamp suffix.
    */
-  payload: string;
+  custom_file_name?: 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).
+   * Defines the maximum length for the recording in seconds. The minimum value is 0.
+   * The maximum value is 14400. The default value is 0 (infinite)
    */
-  voice: string;
+  max_length?: number;
 
   /**
-   * Use this field to add state to every subsequent webhook. It must be a valid
-   * Base-64 encoded string.
+   * If enabled, a beep sound will be played at the start of a recording.
    */
-  client_state?: string;
+  play_beep?: boolean;
 
   /**
-   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
-   * the same `command_id` for the same `call_control_id`.
+   * The audio track to be recorded. Can be either `both`, `inbound` or `outbound`.
+   * If only single track is specified (`inbound`, `outbound`), `channels`
+   * configuration is ignored and it will be recorded as mono (single channel).
    */
-  command_id?: string;
+  recording_track?: 'both' | 'inbound' | 'outbound';
 
   /**
-   * The language you want spoken. This parameter is ignored when a `Polly.*` voice
-   * is specified.
+   * The number of seconds that Telnyx will wait for the recording to be stopped if
+   * silence is detected. The timer only starts when the speech is detected. Please
+   * note that call transcription is used to detect silence and the related charge
+   * will be applied. The minimum value is 0. The default value is 0 (infinite)
    */
-  language?:
-    | 'arb'
-    | 'cmn-CN'
-    | 'cy-GB'
+  timeout_secs?: number;
+
+  /**
+   * Enable post recording transcription. The default value is false.
+   */
+  transcription?: boolean;
+
+  /**
+   * Engine to use for speech recognition. `A` - `Google`, `B` - `Telnyx`,
+   * `deepgram/nova-3` - `Deepgram Nova-3`. Note: `deepgram/nova-3` supports only
+   * `en` and `en-{Region}` languages.
+   */
+  transcription_engine?: 'A' | 'B' | 'deepgram/nova-3';
+
+  /**
+   * Language code for transcription. Note: Not all languages are supported by all
+   * transcription engines (google, telnyx, deepgram). See engine-specific
+   * documentation for supported values.
+   */
+  transcription_language?:
+    | 'af'
+    | 'af-ZA'
+    | 'am'
+    | 'am-ET'
+    | 'ar'
+    | 'ar-AE'
+    | 'ar-BH'
+    | 'ar-DZ'
+    | 'ar-EG'
+    | 'ar-IL'
+    | 'ar-IQ'
+    | 'ar-JO'
+    | 'ar-KW'
+    | 'ar-LB'
+    | 'ar-MA'
+    | 'ar-MR'
+    | 'ar-OM'
+    | 'ar-PS'
+    | 'ar-QA'
+    | 'ar-SA'
+    | 'ar-TN'
+    | 'ar-YE'
+    | 'as'
+    | 'auto_detect'
+    | 'az'
+    | 'az-AZ'
+    | 'ba'
+    | 'be'
+    | 'bg'
+    | 'bg-BG'
+    | 'bn'
+    | 'bn-BD'
+    | 'bn-IN'
+    | 'bo'
+    | 'br'
+    | 'bs'
+    | 'bs-BA'
+    | 'ca'
+    | 'ca-ES'
+    | 'cs'
+    | 'cs-CZ'
+    | 'cy'
+    | 'da'
     | 'da-DK'
+    | 'de'
+    | 'de-AT'
+    | 'de-CH'
     | 'de-DE'
+    | 'el'
+    | 'el-GR'
+    | 'en'
     | 'en-AU'
+    | 'en-CA'
     | 'en-GB'
-    | 'en-GB-WLS'
+    | 'en-GH'
+    | 'en-HK'
+    | 'en-IE'
     | 'en-IN'
+    | 'en-KE'
+    | 'en-NG'
+    | 'en-NZ'
+    | 'en-PH'
+    | 'en-PK'
+    | 'en-SG'
+    | 'en-TZ'
     | 'en-US'
+    | 'en-ZA'
+    | 'es'
+    | 'es-419'
+    | 'es-AR'
+    | 'es-BO'
+    | 'es-CL'
+    | 'es-CO'
+    | 'es-CR'
+    | 'es-DO'
+    | 'es-EC'
     | 'es-ES'
+    | 'es-GT'
+    | 'es-HN'
     | 'es-MX'
+    | 'es-NI'
+    | 'es-PA'
+    | 'es-PE'
+    | 'es-PR'
+    | 'es-PY'
+    | 'es-SV'
     | 'es-US'
+    | 'es-UY'
+    | 'es-VE'
+    | 'et'
+    | 'et-EE'
+    | 'eu'
+    | 'eu-ES'
+    | 'fa'
+    | 'fa-IR'
+    | 'fi'
+    | 'fi-FI'
+    | 'fil-PH'
+    | 'fo'
+    | 'fr'
+    | 'fr-BE'
     | 'fr-CA'
+    | 'fr-CH'
     | 'fr-FR'
+    | 'gl'
+    | 'gl-ES'
+    | 'gu'
+    | 'gu-IN'
+    | 'ha'
+    | 'haw'
+    | 'he'
+    | 'hi'
     | 'hi-IN'
+    | 'hr'
+    | 'hr-HR'
+    | 'ht'
+    | 'hu'
+    | 'hu-HU'
+    | 'hy'
+    | 'hy-AM'
+    | 'id'
+    | 'id-ID'
+    | 'is'
     | 'is-IS'
+    | 'it'
+    | 'it-CH'
     | 'it-IT'
-    | 'ja-JP'
+    | 'iw-IL'
+    | 'ja'
+    | 'ja-JP'
+    | 'jv-ID'
+    | 'jw'
+    | 'ka'
+    | 'ka-GE'
+    | 'kk'
+    | 'kk-KZ'
+    | 'km'
+    | 'km-KH'
+    | 'kn'
+    | 'kn-IN'
+    | 'ko'
     | 'ko-KR'
-    | 'nb-NO'
+    | 'la'
+    | 'lb'
+    | 'ln'
+    | 'lo'
+    | 'lo-LA'
+    | 'lt'
+    | 'lt-LT'
+    | 'lv'
+    | 'lv-LV'
+    | 'mg'
+    | 'mi'
+    | 'mk'
+    | 'mk-MK'
+    | 'ml'
+    | 'ml-IN'
+    | 'mn'
+    | 'mn-MN'
+    | 'mr'
+    | 'mr-IN'
+    | 'ms'
+    | 'ms-MY'
+    | 'mt'
+    | 'my'
+    | 'my-MM'
+    | 'ne'
+    | 'ne-NP'
+    | 'nl'
+    | 'nl-BE'
     | 'nl-NL'
+    | 'nn'
+    | 'no'
+    | 'no-NO'
+    | 'oc'
+    | 'pa'
+    | 'pa-Guru-IN'
+    | 'pl'
     | 'pl-PL'
+    | 'ps'
+    | 'pt'
     | 'pt-BR'
     | 'pt-PT'
+    | 'ro'
     | 'ro-RO'
+    | 'ru'
     | 'ru-RU'
+    | 'rw-RW'
+    | 'sa'
+    | 'sd'
+    | 'si'
+    | 'si-LK'
+    | 'sk'
+    | 'sk-SK'
+    | 'sl'
+    | 'sl-SI'
+    | 'sn'
+    | 'so'
+    | 'sq'
+    | 'sq-AL'
+    | 'sr'
+    | 'sr-RS'
+    | 'ss-latn-za'
+    | 'st-ZA'
+    | 'su'
+    | 'su-ID'
+    | 'sv'
     | 'sv-SE'
-    | 'tr-TR';
-
-  /**
-   * The number of times to play the audio file. Use `infinity` to loop indefinitely.
-   * Defaults to 1.
-   */
-  loop?: Loopcount;
+    | 'sw'
+    | 'sw-KE'
+    | 'sw-TZ'
+    | 'ta'
+    | 'ta-IN'
+    | 'ta-LK'
+    | 'ta-MY'
+    | 'ta-SG'
+    | 'te'
+    | 'te-IN'
+    | 'tg'
+    | 'th'
+    | 'th-TH'
+    | 'tk'
+    | 'tl'
+    | 'tn-latn-za'
+    | 'tr'
+    | 'tr-TR'
+    | 'ts-ZA'
+    | 'tt'
+    | 'uk'
+    | 'uk-UA'
+    | 'ur'
+    | 'ur-IN'
+    | 'ur-PK'
+    | 'uz'
+    | 'uz-UZ'
+    | 've-ZA'
+    | 'vi'
+    | 'vi-VN'
+    | 'xh-ZA'
+    | 'yi'
+    | 'yo'
+    | 'yue-Hant-HK'
+    | 'zh'
+    | 'zh-TW'
+    | 'zu-ZA';
 
   /**
-   * The type of the provided payload. The payload can either be plain text, or
-   * Speech Synthesis Markup Language (SSML).
+   * Defines maximum number of speakers in the conversation. Applies to `google`
+   * engine only.
    */
-  payload_type?: 'text' | 'ssml';
+  transcription_max_speaker_count?: number;
 
   /**
-   * This parameter impacts speech quality, language options and payload types. When
-   * using `basic`, only the `en-US` language and payload type `text` are allowed.
+   * Defines minimum number of speakers in the conversation. Applies to `google`
+   * engine only.
    */
-  service_level?: 'basic' | 'premium';
+  transcription_min_speaker_count?: number;
 
   /**
-   * 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.
+   * Enables profanity_filter. Applies to `google` engine only.
    */
-  stop?: string;
+  transcription_profanity_filter?: boolean;
 
   /**
-   * Specifies which legs of the call should receive the spoken audio.
+   * Enables speaker diarization. Applies to `google` engine only.
    */
-  target_legs?: 'self' | 'opposite' | 'both';
+  transcription_speaker_diarization?: boolean;
 
   /**
-   * The settings associated with the voice selected
+   * When set to `trim-silence`, silence will be removed from the beginning and end
+   * of the recording.
    */
-  voice_settings?:
-    | ElevenLabsVoiceSettings
-    | TelnyxVoiceSettings
-    | AwsVoiceSettings
-    | Shared.MinimaxVoiceSettings
-    | Shared.AzureVoiceSettings
-    | Shared.RimeVoiceSettings
-    | Shared.ResembleVoiceSettings
-    | Shared.InworldVoiceSettings
-    | Shared.XaiVoiceSettings;
+  trim?: 'trim-silence';
 }
 
-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;
-
+export interface ActionStopRecordingParams {
   /**
    * Use this field to add state to every subsequent webhook. It must be a valid
    * Base-64 encoded string.
@@ -3620,96 +3952,61 @@ export interface ActionStartAIAssistantParams {
   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.
+   * Uniquely identifies the resource.
    */
-  greeting?: string;
+  recording_id?: string;
+}
 
+export interface ActionReferParams {
   /**
-   * Settings for handling user interruptions during assistant speech
+   * The SIP URI to which the call will be referred to.
    */
-  interruption_settings?: InterruptionSettings;
+  sip_address: string;
 
   /**
-   * 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.
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
    */
-  message_history?: Array<UserMessage | AssistantMessage | ToolMessage | SystemMessage | DeveloperMessage>;
+  client_state?: string;
 
   /**
-   * A list of participants to add to the conversation when it starts.
+   * 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.
    */
-  participants?: Array<AIAssistantJoinParticipant>;
+  command_id?: string;
 
   /**
-   * When `true`, a webhook is sent each time the conversation message history is
-   * updated.
+   * Custom headers to be added to the SIP INVITE.
    */
-  send_message_history_updates?: boolean;
+  custom_headers?: Array<CallsAPI.CustomSipHeader>;
 
   /**
-   * 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.
+   * SIP Authentication password used for SIP challenges.
    */
-  transcription?: TranscriptionConfig;
+  sip_auth_password?: 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`.
+   * SIP Authentication username used for SIP challenges.
    */
-  voice?: string;
+  sip_auth_username?: string;
 
   /**
-   * The settings associated with the voice selected
+   * SIP headers to be added to the request. Currently only User-to-User header is
+   * supported.
    */
-  voice_settings?:
-    | ElevenLabsVoiceSettings
-    | TelnyxVoiceSettings
-    | AwsVoiceSettings
-    | Shared.AzureVoiceSettings
-    | Shared.RimeVoiceSettings
-    | Shared.ResembleVoiceSettings
-    | Shared.XaiVoiceSettings;
+  sip_headers?: Array<CallsAPI.SipHeader>;
 }
 
-export interface ActionStartConversationRelayParams {
+export interface ActionRejectParams {
   /**
-   * Custom parameters for the Conversation Relay session. Pass key-value data as
-   * `assistant.dynamic_variables` to make it available to the relay session.
+   * Cause for call rejection.
    */
-  assistant?: ActionStartConversationRelayParams.Assistant;
+  cause: 'CALL_REJECTED' | 'USER_BUSY';
 
   /**
-   * Use this field to add state to subsequent webhooks. It must be a valid Base-64
-   * encoded string.
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
    */
   client_state?: string;
 
@@ -3718,159 +4015,249 @@ export interface ActionStartConversationRelayParams {
    * the same `command_id` for the same `call_control_id`.
    */
   command_id?: string;
+}
 
+export interface ActionSendDtmfParams {
   /**
-   * Enable DTMF detection for the relay session.
+   * DTMF digits to send. Valid digits are 0-9, A-D, \*, and #. Pauses can be added
+   * using w (0.5s) and W (1s).
    */
-  conversation_relay_dtmf_detection?: boolean;
+  digits: string;
 
   /**
-   * 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.
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
    */
-  conversation_relay_settings?: ActionStartConversationRelayParams.ConversationRelaySettings;
+  client_state?: string;
 
   /**
-   * WebSocket URL for your Conversation Relay server. Must start with `ws://` or
-   * `wss://`.
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
    */
-  conversation_relay_url?: string;
+  command_id?: 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.
+   * Specifies for how many milliseconds each digit will be played in the audio
+   * stream. Ranges from 100 to 500ms
    */
-  custom_parameters?: { [key: string]: unknown };
+  duration_millis?: number;
+}
 
+export interface ActionSendSipInfoParams {
   /**
-   * Public alias for `conversation_relay_dtmf_detection`. If both are present, this
-   * value wins.
+   * Content of the SIP INFO
    */
-  dtmf_detection?: boolean;
+  body: string;
 
   /**
-   * Text played when the relay session starts.
+   * Content type of the INFO body. Must be MIME type compliant. There is a 1,400
+   * bytes limit
    */
-  greeting?: string;
+  content_type: 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.
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
    */
-  interruptible?: ConversationRelayInterruptible;
+  client_state?: 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.
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
    */
-  interruptible_greeting?: ConversationRelayInterruptible;
+  command_id?: string;
+}
 
+export interface ActionStartSiprecParams {
   /**
-   * Settings for handling caller interruptions during Conversation Relay speech.
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
    */
-  interruption_settings?: CallsAPI.ConversationRelayInterruptionSettings;
+  client_state?: string;
 
   /**
-   * Default language for the relay session. This value is used for both
-   * text-to-speech and speech recognition.
+   * Name of configured SIPREC connector to be used.
    */
-  language?: string;
+  connector_name?: string;
 
   /**
-   * Per-language TTS and transcription settings.
+   * When set, custom parameters will be added as metadata
+   * (recording.session.ExtensionParameters). Otherwise, they’ll be added to sip
+   * headers.
    */
-  languages?: Array<CallsAPI.ConversationRelayLanguage>;
+  include_metadata_custom_headers?: true | false;
 
   /**
-   * Structured voice provider. Must be supplied together with `structured_provider`.
+   * 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.
    */
-  provider?: string;
+  secure?: true | false;
 
   /**
-   * Provider-specific structured voice settings. Must be supplied together with
-   * `provider`; Telnyx sends the value as the nested provider configuration for
-   * Conversation Relay.
+   * 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.
    */
-  structured_provider?: { [key: string]: unknown };
+  session_timeout_secs?: number;
 
   /**
-   * @deprecated Not supported for Conversation Relay start requests. Use
-   * `transcription_engine` and `transcription_engine_config` instead.
+   * Specifies SIP transport protocol.
    */
-  transcription?: { [key: string]: unknown };
+  sip_transport?: 'udp' | 'tcp' | 'tls';
 
   /**
-   * 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.
+   * Specifies which track should be sent on siprec session.
    */
-  transcription_engine?:
-    | 'Google'
-    | 'Telnyx'
-    | 'Deepgram'
-    | 'Azure'
-    | 'xAI'
-    | 'AssemblyAI'
-    | 'Speechmatics'
-    | 'Soniox'
-    | 'A'
-    | 'B';
+  siprec_track?: 'inbound_track' | 'outbound_track' | 'both_tracks';
+}
 
+export interface ActionStopSiprecParams {
   /**
-   * 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.
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
    */
-  transcription_engine_config?: { [key: string]: unknown };
+  client_state?: string;
 
   /**
-   * Text-to-speech provider. If omitted, Telnyx derives it from `voice` or
-   * `provider`.
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
    */
-  tts_provider?: string;
+  command_id?: string;
+}
 
+export interface ActionSpeakParams {
   /**
-   * Public alias for `conversation_relay_url`. Must start with `ws://` or `wss://`.
-   * If both are present, this value wins.
+   * The text or SSML to be converted into speech. There is a 3,000 character limit.
    */
-  url?: string;
+  payload: string;
 
   /**
-   * The voice to be used by the voice assistant. Currently we support ElevenLabs,
-   * Telnyx and AWS voices.
+   * Specifies the voice used in speech synthesis.
    *
-   * **Supported Providers:**
+   * - Define voices using the format `<Provider>.<Model>.<VoiceId>`. Specifying only
+   *   the provider will give default values for voice_id and model_id.
+   *
+   *   **Supported Providers:**
    *
    * - **AWS:** Use `AWS.Polly.<VoiceId>` (e.g., `AWS.Polly.Joanna`). For neural
    *   voices, which provide more realistic, human-like speech, append `-Neural` to
    *   the `VoiceId` (e.g., `AWS.Polly.Joanna-Neural`). Check the
    *   [available voices](https://docs.aws.amazon.com/polly/latest/dg/available-voices.html)
    *   for compatibility.
-   * - **Azure:** Use `Azure.<VoiceId>. (e.g. Azure.en-CA-ClaraNeural,
-   *   Azure.en-CA-LiamNeural, Azure.en-US-BrianMultilingualNeural,
-   *   Azure.en-US-Ava:DragonHDLatestNeural. For a complete list of voices, go to
-   *   [Azure Voice Gallery](https://speech.microsoft.com/portal/voicegallery).)
+   * - **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.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
+   *   `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>`
+   * - **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`.
+   *   `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;
+  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
@@ -3887,59 +4274,90 @@ export interface ActionStartConversationRelayParams {
     | Shared.XaiVoiceSettings;
 }
 
-export namespace ActionStartConversationRelayParams {
+export interface ActionStartStreamingParams {
   /**
-   * Custom parameters for the Conversation Relay session. Pass key-value data as
-   * `assistant.dynamic_variables` to make it available to the relay session.
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
    */
-  export interface Assistant {
-    /**
-     * Custom key-value parameters forwarded to the Conversation Relay session.
-     */
-    dynamic_variables?: { [key: string]: string };
+  client_state?: string;
 
-    [k: string]: unknown;
-  }
+  /**
+   * 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;
 
   /**
-   * 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.
+   * Custom parameters to be sent as part of the WebSocket connection.
    */
-  export interface ConversationRelaySettings {
-    /**
-     * WebSocket URL for your Conversation Relay server. Must start with `ws://` or
-     * `wss://`.
-     */
-    url: string;
+  custom_parameters?: Array<ActionStartStreamingParams.CustomParameter>;
 
-    /**
-     * Whether to enable DTMF detection during the relay session.
-     */
-    dtmf_detection?: boolean;
+  dialogflow_config?: CallsAPI.DialogflowConfig;
 
-    /**
-     * 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;
+  /**
+   * Enables Dialogflow for the current call. The default value is false.
+   */
+  enable_dialogflow?: boolean;
+
+  /**
+   * An authentication token to be sent as part of the WebSocket connection. Maximum
+   * length is 4000 characters.
+   */
+  stream_auth_token?: string;
+
+  /**
+   * Indicates codec for bidirectional streaming RTP payloads. Used only with
+   * stream_bidirectional_mode=rtp. Case sensitive.
+   */
+  stream_bidirectional_codec?: CallsAPI.StreamBidirectionalCodec;
+
+  /**
+   * Configures method of bidirectional streaming (mp3, rtp).
+   */
+  stream_bidirectional_mode?: CallsAPI.StreamBidirectionalMode;
+
+  /**
+   * Audio sampling rate.
+   */
+  stream_bidirectional_sampling_rate?: CallsAPI.StreamBidirectionalSamplingRate;
+
+  /**
+   * Specifies which call legs should receive the bidirectional stream audio.
+   */
+  stream_bidirectional_target_legs?: CallsAPI.StreamBidirectionalTargetLegs;
+
+  /**
+   * Specifies the codec to be used for the streamed audio. When set to 'default' or
+   * when transcoding is not possible, the codec from the call will be used.
+   */
+  stream_codec?: CallsAPI.StreamCodec;
+
+  /**
+   * Specifies which track should be streamed.
+   */
+  stream_track?: 'inbound_track' | 'outbound_track' | 'both_tracks';
+
+  /**
+   * The destination WebSocket address where the stream is going to be delivered.
+   */
+  stream_url?: string;
+}
 
+export namespace ActionStartStreamingParams {
+  export interface CustomParameter {
     /**
-     * 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.
+     * The name of the custom parameter.
      */
-    interruptible_greeting?: ActionsAPI.ConversationRelayInterruptible;
+    name?: string;
 
     /**
-     * Language-specific TTS and transcription settings.
+     * The value of the custom parameter.
      */
-    languages?: Array<CallsAPI.ConversationRelayLanguage>;
+    value?: string;
   }
 }
 
-export interface ActionStartForkingParams {
+export interface ActionStopStreamingParams {
   /**
    * Use this field to add state to every subsequent webhook. It must be a valid
    * Base-64 encoded string.
@@ -3953,23 +4371,10 @@ export interface ActionStartForkingParams {
   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.
+   * Identifies the stream. If the `stream_id` is not provided the command stops all
+   * streams associated with a given `call_control_id`.
    */
-  tx?: string;
+  stream_id?: string;
 }
 
 export interface ActionStartNoiseSuppressionParams {
@@ -4070,25 +4475,29 @@ export namespace ActionStartNoiseSuppressionParams {
   }
 }
 
-export interface ActionStartPlaybackParams {
+export interface ActionStopNoiseSuppressionParams {
   /**
-   * Specifies the type of audio provided in `audio_url` or `playback_content`.
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
    */
-  audio_type?: 'mp3' | 'wav';
+  client_state?: string;
 
   /**
-   * The URL of a file to be played back on the call. The URL can point to either a
-   * WAV or MP3 file. media_name and audio_url cannot be used together in one
-   * request.
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
    */
-  audio_url?: string;
+  command_id?: string;
+}
 
+export interface ActionSwitchSupervisorRoleParams {
   /**
-   * Caches the audio file. Useful when playing the same audio file multiple times
-   * during the call.
+   * The supervisor role to switch to. 'barge' allows speaking to both parties,
+   * 'whisper' allows speaking to caller only, 'monitor' allows listening only.
    */
-  cache_audio?: boolean;
+  role: 'barge' | 'whisper' | 'monitor';
+}
 
+export interface ActionStartTranscriptionParams {
   /**
    * Use this field to add state to every subsequent webhook. It must be a valid
    * Base-64 encoded string.
@@ -4102,60 +4511,103 @@ export interface ActionStartPlaybackParams {
   command_id?: string;
 
   /**
-   * The number of times the audio file should be played. If supplied, the value must
-   * be an integer between 1 and 100, or the special string `infinity` for an endless
-   * loop.
+   * Engine to use for speech recognition. Legacy values `A` - `Google`, `B` -
+   * `Telnyx` are supported for backward compatibility.
    */
-  loop?: Loopcount;
+  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;
 
   /**
-   * The media_name of a file to be played back on the call. The media_name must
-   * point to a file previously uploaded to api.telnyx.com/v2/media by the same
-   * user/organization. The file must either be a WAV or MP3 file.
+   * 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`.
    */
-  media_name?: string;
+  transcription_tracks?: string;
+}
 
+export interface ActionStopTranscriptionParams {
   /**
-   * When enabled, audio will be mixed on top of any other audio that is actively
-   * being played back. Note that `overlay: true` will only work if there is another
-   * audio file already being played on the call.
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
    */
-  overlay?: boolean;
+  client_state?: string;
 
   /**
-   * Allows a user to provide base64 encoded mp3 or wav. Note: when using this
-   * parameter, `media_url` and `media_name` in the `playback_started` and
-   * `playback_ended` webhooks will be empty
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
    */
-  playback_content?: string;
+  command_id?: string;
+}
 
+export interface ActionTransferParams {
   /**
-   * 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.
+   * The DID or SIP URI to dial out to. For SIP URI destinations, append
+   * `;secure=true` or `;secure=srtp` to enable SRTP media encryption for that
+   * endpoint, or `;secure=dtls` to enable DTLS media encryption for that endpoint.
+   * If `media_encryption` is set to `SRTP` or `DTLS`, it takes precedence over any
+   * per-endpoint `secure` URI parameter. You may also append a comma followed by
+   * DTMF digits (e.g. `+18004247767,200`) to play those digits as DTMF once the
+   * transfer destination answers — equivalent to setting `send_digits_on_answer`
+   * separately. If both are present, the explicit `send_digits_on_answer` parameter
+   * takes precedence.
    */
-  stop?: string;
+  to: string;
 
   /**
-   * Specifies the leg or legs on which audio will be played. If supplied, the value
-   * must be either `self`, `opposite` or `both`.
+   * Enables Answering Machine Detection. When a call is answered, Telnyx runs
+   * real-time detection to determine if it was picked up by a human or a machine and
+   * sends an `call.machine.detection.ended` webhook with the analysis result. If
+   * 'greeting_end' or 'detect_words' is used and a 'machine' is detected, you will
+   * receive another 'call.machine.greeting.ended' webhook when the answering machine
+   * greeting ends with a beep or silence. If `detect_beep` is used, you will only
+   * receive 'call.machine.greeting.ended' if a beep is detected.
    */
-  target_legs?: string;
-}
+  answering_machine_detection?:
+    | 'premium'
+    | 'detect'
+    | 'detect_beep'
+    | 'detect_words'
+    | 'greeting_end'
+    | 'disabled';
 
-export interface ActionStartRecordingParams {
   /**
-   * When `dual`, final audio file will be stereo recorded with the first leg on
-   * channel A, and the rest on channel B.
+   * Optional configuration parameters to modify 'answering_machine_detection'
+   * performance. Only `total_analysis_time_millis` and `greeting_duration_millis`
+   * parameters are applicable when `premium` is selected as
+   * answering_machine_detection.
    */
-  channels: 'single' | 'dual';
+  answering_machine_detection_config?: ActionTransferParams.AnsweringMachineDetectionConfig;
 
   /**
-   * The audio file format used when storing the call recording. Can be either `mp3`
-   * or `wav`.
+   * The URL of a file to be played back when the transfer destination answers before
+   * bridging the call. The URL can point to either a WAV or MP3 file. media_name and
+   * audio_url cannot be used together in one request.
    */
-  format: 'wav' | 'mp3';
+  audio_url?: string;
 
   /**
    * Use this field to add state to every subsequent webhook. It must be a valid
@@ -4170,1039 +4622,587 @@ export interface ActionStartRecordingParams {
   command_id?: string;
 
   /**
-   * The custom recording file name to be used instead of the default `call_leg_id`.
-   * Telnyx will still add a Unix timestamp suffix.
+   * Custom headers to be added to the SIP INVITE.
    */
-  custom_file_name?: string;
+  custom_headers?: Array<CallsAPI.CustomSipHeader>;
 
   /**
-   * Defines the maximum length for the recording in seconds. The minimum value is 0.
-   * The maximum value is 14400. The default value is 0 (infinite)
+   * If set to false, early media will not be passed to the originating leg.
    */
-  max_length?: number;
+  early_media?: boolean;
 
   /**
-   * If enabled, a beep sound will be played at the start of a recording.
-   */
-  play_beep?: boolean;
-
-  /**
-   * The audio track to be recorded. Can be either `both`, `inbound` or `outbound`.
-   * If only single track is specified (`inbound`, `outbound`), `channels`
-   * configuration is ignored and it will be recorded as mono (single channel).
-   */
-  recording_track?: 'both' | 'inbound' | 'outbound';
-
-  /**
-   * The number of seconds that Telnyx will wait for the recording to be stopped if
-   * silence is detected. The timer only starts when the speech is detected. Please
-   * note that call transcription is used to detect silence and the related charge
-   * will be applied. The minimum value is 0. The default value is 0 (infinite)
-   */
-  timeout_secs?: number;
-
-  /**
-   * Enable post recording transcription. The default value is false.
-   */
-  transcription?: boolean;
-
-  /**
-   * Engine to use for speech recognition. `A` - `Google`, `B` - `Telnyx`,
-   * `deepgram/nova-3` - `Deepgram Nova-3`. Note: `deepgram/nova-3` supports only
-   * `en` and `en-{Region}` languages.
-   */
-  transcription_engine?: 'A' | 'B' | 'deepgram/nova-3';
-
-  /**
-   * Language code for transcription. Note: Not all languages are supported by all
-   * transcription engines (google, telnyx, deepgram). See engine-specific
-   * documentation for supported values.
-   */
-  transcription_language?:
-    | 'af'
-    | 'af-ZA'
-    | 'am'
-    | 'am-ET'
-    | 'ar'
-    | 'ar-AE'
-    | 'ar-BH'
-    | 'ar-DZ'
-    | 'ar-EG'
-    | 'ar-IL'
-    | 'ar-IQ'
-    | 'ar-JO'
-    | 'ar-KW'
-    | 'ar-LB'
-    | 'ar-MA'
-    | 'ar-MR'
-    | 'ar-OM'
-    | 'ar-PS'
-    | 'ar-QA'
-    | 'ar-SA'
-    | 'ar-TN'
-    | 'ar-YE'
-    | 'as'
-    | 'auto_detect'
-    | 'az'
-    | 'az-AZ'
-    | 'ba'
-    | 'be'
-    | 'bg'
-    | 'bg-BG'
-    | 'bn'
-    | 'bn-BD'
-    | 'bn-IN'
-    | 'bo'
-    | 'br'
-    | 'bs'
-    | 'bs-BA'
-    | 'ca'
-    | 'ca-ES'
-    | 'cs'
-    | 'cs-CZ'
-    | 'cy'
-    | 'da'
-    | 'da-DK'
-    | 'de'
-    | 'de-AT'
-    | 'de-CH'
-    | 'de-DE'
-    | 'el'
-    | 'el-GR'
-    | 'en'
-    | 'en-AU'
-    | 'en-CA'
-    | 'en-GB'
-    | 'en-GH'
-    | 'en-HK'
-    | 'en-IE'
-    | 'en-IN'
-    | 'en-KE'
-    | 'en-NG'
-    | 'en-NZ'
-    | 'en-PH'
-    | 'en-PK'
-    | 'en-SG'
-    | 'en-TZ'
-    | 'en-US'
-    | 'en-ZA'
-    | 'es'
-    | 'es-419'
-    | 'es-AR'
-    | 'es-BO'
-    | 'es-CL'
-    | 'es-CO'
-    | 'es-CR'
-    | 'es-DO'
-    | 'es-EC'
-    | 'es-ES'
-    | 'es-GT'
-    | 'es-HN'
-    | 'es-MX'
-    | 'es-NI'
-    | 'es-PA'
-    | 'es-PE'
-    | 'es-PR'
-    | 'es-PY'
-    | 'es-SV'
-    | 'es-US'
-    | 'es-UY'
-    | 'es-VE'
-    | 'et'
-    | 'et-EE'
-    | 'eu'
-    | 'eu-ES'
-    | 'fa'
-    | 'fa-IR'
-    | 'fi'
-    | 'fi-FI'
-    | 'fil-PH'
-    | 'fo'
-    | 'fr'
-    | 'fr-BE'
-    | 'fr-CA'
-    | 'fr-CH'
-    | 'fr-FR'
-    | 'gl'
-    | 'gl-ES'
-    | 'gu'
-    | 'gu-IN'
-    | 'ha'
-    | 'haw'
-    | 'he'
-    | 'hi'
-    | 'hi-IN'
-    | 'hr'
-    | 'hr-HR'
-    | 'ht'
-    | 'hu'
-    | 'hu-HU'
-    | 'hy'
-    | 'hy-AM'
-    | 'id'
-    | 'id-ID'
-    | 'is'
-    | 'is-IS'
-    | 'it'
-    | 'it-CH'
-    | 'it-IT'
-    | 'iw-IL'
-    | 'ja'
-    | 'ja-JP'
-    | 'jv-ID'
-    | 'jw'
-    | 'ka'
-    | 'ka-GE'
-    | 'kk'
-    | 'kk-KZ'
-    | 'km'
-    | 'km-KH'
-    | 'kn'
-    | 'kn-IN'
-    | 'ko'
-    | 'ko-KR'
-    | 'la'
-    | 'lb'
-    | 'ln'
-    | 'lo'
-    | 'lo-LA'
-    | 'lt'
-    | 'lt-LT'
-    | 'lv'
-    | 'lv-LV'
-    | 'mg'
-    | 'mi'
-    | 'mk'
-    | 'mk-MK'
-    | 'ml'
-    | 'ml-IN'
-    | 'mn'
-    | 'mn-MN'
-    | 'mr'
-    | 'mr-IN'
-    | 'ms'
-    | 'ms-MY'
-    | 'mt'
-    | 'my'
-    | 'my-MM'
-    | 'ne'
-    | 'ne-NP'
-    | 'nl'
-    | 'nl-BE'
-    | 'nl-NL'
-    | 'nn'
-    | 'no'
-    | 'no-NO'
-    | 'oc'
-    | 'pa'
-    | 'pa-Guru-IN'
-    | 'pl'
-    | 'pl-PL'
-    | 'ps'
-    | 'pt'
-    | 'pt-BR'
-    | 'pt-PT'
-    | 'ro'
-    | 'ro-RO'
-    | 'ru'
-    | 'ru-RU'
-    | 'rw-RW'
-    | 'sa'
-    | 'sd'
-    | 'si'
-    | 'si-LK'
-    | 'sk'
-    | 'sk-SK'
-    | 'sl'
-    | 'sl-SI'
-    | 'sn'
-    | 'so'
-    | 'sq'
-    | 'sq-AL'
-    | 'sr'
-    | 'sr-RS'
-    | 'ss-latn-za'
-    | 'st-ZA'
-    | 'su'
-    | 'su-ID'
-    | 'sv'
-    | 'sv-SE'
-    | 'sw'
-    | 'sw-KE'
-    | 'sw-TZ'
-    | 'ta'
-    | 'ta-IN'
-    | 'ta-LK'
-    | 'ta-MY'
-    | 'ta-SG'
-    | 'te'
-    | 'te-IN'
-    | 'tg'
-    | 'th'
-    | 'th-TH'
-    | 'tk'
-    | 'tl'
-    | 'tn-latn-za'
-    | 'tr'
-    | 'tr-TR'
-    | 'ts-ZA'
-    | 'tt'
-    | 'uk'
-    | 'uk-UA'
-    | 'ur'
-    | 'ur-IN'
-    | 'ur-PK'
-    | 'uz'
-    | 'uz-UZ'
-    | 've-ZA'
-    | 'vi'
-    | 'vi-VN'
-    | 'xh-ZA'
-    | 'yi'
-    | 'yo'
-    | 'yue-Hant-HK'
-    | 'zh'
-    | 'zh-TW'
-    | 'zu-ZA';
-
-  /**
-   * Defines maximum number of speakers in the conversation. Applies to `google`
-   * engine only.
-   */
-  transcription_max_speaker_count?: number;
-
-  /**
-   * Defines minimum number of speakers in the conversation. Applies to `google`
-   * engine only.
-   */
-  transcription_min_speaker_count?: number;
-
-  /**
-   * Enables profanity_filter. Applies to `google` engine only.
-   */
-  transcription_profanity_filter?: boolean;
-
-  /**
-   * Enables speaker diarization. Applies to `google` engine only.
-   */
-  transcription_speaker_diarization?: boolean;
-
-  /**
-   * When set to `trim-silence`, silence will be removed from the beginning and end
-   * of the recording.
-   */
-  trim?: 'trim-silence';
-}
-
-export interface ActionStartSiprecParams {
-  /**
-   * Use this field to add state to every subsequent webhook. It must be a valid
-   * Base-64 encoded string.
-   */
-  client_state?: string;
-
-  /**
-   * Name of configured SIPREC connector to be used.
-   */
-  connector_name?: string;
-
-  /**
-   * When set, custom parameters will be added as metadata
-   * (recording.session.ExtensionParameters). Otherwise, they’ll be added to sip
-   * headers.
-   */
-  include_metadata_custom_headers?: true | false;
-
-  /**
-   * Controls whether to encrypt media sent to your SRS using SRTP and TLS. When set
-   * you need to configure SRS port in your connector to 5061.
-   */
-  secure?: true | false;
-
-  /**
-   * Sets `Session-Expires` header to the INVITE. A reinvite is sent every half the
-   * value set. Usefull for session keep alive. Minimum value is 90, set to 0 to
-   * disable.
-   */
-  session_timeout_secs?: number;
-
-  /**
-   * Specifies SIP transport protocol.
-   */
-  sip_transport?: 'udp' | 'tcp' | 'tls';
-
-  /**
-   * Specifies which track should be sent on siprec session.
-   */
-  siprec_track?: 'inbound_track' | 'outbound_track' | 'both_tracks';
-}
-
-export interface ActionStartStreamingParams {
-  /**
-   * Use this field to add state to every subsequent webhook. It must be a valid
-   * Base-64 encoded string.
-   */
-  client_state?: string;
-
-  /**
-   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
-   * the same `command_id` for the same `call_control_id`.
-   */
-  command_id?: string;
-
-  /**
-   * Custom parameters to be sent as part of the WebSocket connection.
-   */
-  custom_parameters?: Array<ActionStartStreamingParams.CustomParameter>;
-
-  dialogflow_config?: CallsAPI.DialogflowConfig;
-
-  /**
-   * Enables Dialogflow for the current call. The default value is false.
-   */
-  enable_dialogflow?: boolean;
-
-  /**
-   * An authentication token to be sent as part of the WebSocket connection. Maximum
-   * length is 4000 characters.
-   */
-  stream_auth_token?: string;
-
-  /**
-   * Indicates codec for bidirectional streaming RTP payloads. Used only with
-   * stream_bidirectional_mode=rtp. Case sensitive.
-   */
-  stream_bidirectional_codec?: CallsAPI.StreamBidirectionalCodec;
-
-  /**
-   * Configures method of bidirectional streaming (mp3, rtp).
-   */
-  stream_bidirectional_mode?: CallsAPI.StreamBidirectionalMode;
-
-  /**
-   * Audio sampling rate.
-   */
-  stream_bidirectional_sampling_rate?: CallsAPI.StreamBidirectionalSamplingRate;
-
-  /**
-   * Specifies which call legs should receive the bidirectional stream audio.
-   */
-  stream_bidirectional_target_legs?: CallsAPI.StreamBidirectionalTargetLegs;
-
-  /**
-   * Specifies the codec to be used for the streamed audio. When set to 'default' or
-   * when transcoding is not possible, the codec from the call will be used.
-   */
-  stream_codec?: CallsAPI.StreamCodec;
-
-  /**
-   * Specifies which track should be streamed.
-   */
-  stream_track?: 'inbound_track' | 'outbound_track' | 'both_tracks';
-
-  /**
-   * The destination WebSocket address where the stream is going to be delivered.
-   */
-  stream_url?: string;
-}
-
-export namespace ActionStartStreamingParams {
-  export interface CustomParameter {
-    /**
-     * The name of the custom parameter.
-     */
-    name?: string;
-
-    /**
-     * The value of the custom parameter.
-     */
-    value?: string;
-  }
-}
-
-export interface ActionStartTranscriptionParams {
-  /**
-   * Use this field to add state to every subsequent webhook. It must be a valid
-   * Base-64 encoded string.
-   */
-  client_state?: string;
-
-  /**
-   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
-   * the same `command_id` for the same `call_control_id`.
-   */
-  command_id?: string;
-
-  /**
-   * Engine to use for speech recognition. 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`.
-   */
-  transcription_tracks?: string;
-}
-
-export interface ActionStopAIAssistantParams {
-  /**
-   * Use this field to add state to every subsequent webhook. It must be a valid
-   * Base-64 encoded string.
-   */
-  client_state?: string;
-
-  /**
-   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
-   * the same `command_id` for the same `call_control_id`.
-   */
-  command_id?: string;
-}
-
-export interface 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.
-   */
-  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`.
+   * The `from` number to be used as the caller id presented to the destination (`to`
+   * number). The number should be in +E164 format. This attribute will default to
+   * the `to` number of the original call if omitted.
    */
-  command_id?: string;
+  from?: string;
 
   /**
-   * Optionally specify a `stream_type`. This should match the `stream_type` that was
-   * used in `fork_start` command to properly stop the fork.
+   * The `from_display_name` string to be used as the caller id name (SIP From
+   * Display Name) presented to the destination (`to` number). The string should have
+   * a maximum of 128 characters, containing only letters, numbers, spaces, and
+   * -\_~!.+ special characters. If ommited, the display name will be the same as the
+   * number in the `from` field.
    */
-  stream_type?: 'raw' | 'decrypted';
-}
+  from_display_name?: string;
 
-export interface ActionStopGatherParams {
   /**
-   * Use this field to add state to every subsequent webhook. It must be a valid
-   * Base-64 encoded string.
+   * Defines whether media should be encrypted on the new call leg. For SIP URI
+   * destinations, media encryption can also be requested per endpoint with the
+   * `secure` URI parameter: `;secure=true` or `;secure=srtp` enables SRTP, and
+   * `;secure=dtls` enables DTLS. This parameter, when set to `SRTP` or `DTLS`, takes
+   * precedence over the per-endpoint `secure` value.
    */
-  client_state?: string;
+  media_encryption?: 'disabled' | 'SRTP' | 'DTLS';
 
   /**
-   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
-   * the same `command_id` for the same `call_control_id`.
+   * The media_name of a file to be played back when the transfer destination answers
+   * before bridging the call. The media_name must point to a file previously
+   * uploaded to api.telnyx.com/v2/media by the same user/organization. The file must
+   * either be a WAV or MP3 file.
    */
-  command_id?: string;
-}
+  media_name?: string;
 
-export interface ActionStopNoiseSuppressionParams {
   /**
-   * Use this field to add state to every subsequent webhook. It must be a valid
-   * Base-64 encoded string.
+   * When enabled, DTMF tones are not passed to the call participant. The webhooks
+   * containing the DTMF information will be sent.
    */
-  client_state?: string;
+  mute_dtmf?: 'none' | 'both' | 'self' | 'opposite';
 
   /**
-   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
-   * the same `command_id` for the same `call_control_id`.
+   * Specifies behavior after the bridge ends (i.e. the opposite leg either hangs up
+   * or is transferred). If supplied with the value `self`, the current leg will be
+   * parked after unbridge. If not set, the default behavior is to hang up the leg.
    */
-  command_id?: string;
-}
+  park_after_unbridge?: string;
 
-export interface ActionStopPlaybackParams {
   /**
-   * Use this field to add state to every subsequent webhook. It must be a valid
-   * Base-64 encoded string.
+   * The list of comma-separated codecs in order of preference to be used during the
+   * call. The codecs supported are `G722`, `PCMU`, `PCMA`, `G729`, `OPUS`, `VP8`,
+   * `H264`, `AMR-WB`.
    */
-  client_state?: string;
+  preferred_codecs?: string;
 
   /**
-   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
-   * the same `command_id` for the same `call_control_id`.
+   * Indicates the privacy level to be used for the call. When set to `id`, caller ID
+   * information (name and number) will be hidden from the called party. When set to
+   * `none` or omitted, caller ID will be shown normally.
    */
-  command_id?: string;
+  privacy?: 'id' | 'none';
 
   /**
-   * When enabled, it stops the audio being played in the overlay queue.
+   * Start recording automatically after an event. Disabled by default.
    */
-  overlay?: boolean;
+  record?: 'record-from-answer';
 
   /**
-   * 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.
+   * Defines which channel should be recorded ('single' or 'dual') when `record` is
+   * specified.
    */
-  stop?: string;
-}
+  record_channels?: 'single' | 'dual';
 
-export interface ActionStopRecordingParams {
   /**
-   * Use this field to add state to every subsequent webhook. It must be a valid
-   * Base-64 encoded string.
+   * The custom recording file name to be used instead of the default `call_leg_id`.
+   * Telnyx will still add a Unix timestamp suffix.
    */
-  client_state?: string;
+  record_custom_file_name?: string;
 
   /**
-   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
-   * the same `command_id` for the same `call_control_id`.
+   * Defines the format of the recording ('wav' or 'mp3') when `record` is specified.
    */
-  command_id?: string;
+  record_format?: 'wav' | 'mp3';
 
   /**
-   * Uniquely identifies the resource.
+   * Defines the maximum length for the recording in seconds when `record` is
+   * specified. The minimum value is 0. The maximum value is 43200. The default value
+   * is 0 (infinite).
    */
-  recording_id?: string;
-}
+  record_max_length?: number;
 
-export interface ActionStopSiprecParams {
   /**
-   * Use this field to add state to every subsequent webhook. It must be a valid
-   * Base-64 encoded string.
+   * The number of seconds that Telnyx will wait for the recording to be stopped if
+   * silence is detected when `record` is specified. The timer only starts when the
+   * speech is detected. Please note that call transcription is used to detect
+   * silence and the related charge will be applied. The minimum value is 0. The
+   * default value is 0 (infinite).
    */
-  client_state?: string;
+  record_timeout_secs?: number;
 
   /**
-   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
-   * the same `command_id` for the same `call_control_id`.
+   * The audio track to be recorded. Can be either `both`, `inbound` or `outbound`.
+   * If only single track is specified (`inbound`, `outbound`), `channels`
+   * configuration is ignored and it will be recorded as mono (single channel).
    */
-  command_id?: string;
-}
+  record_track?: 'both' | 'inbound' | 'outbound';
 
-export interface ActionStopStreamingParams {
   /**
-   * Use this field to add state to every subsequent webhook. It must be a valid
-   * Base-64 encoded string.
+   * When set to `trim-silence`, silence will be removed from the beginning and end
+   * of the recording.
    */
-  client_state?: string;
+  record_trim?: 'trim-silence';
 
   /**
-   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
-   * the same `command_id` for the same `call_control_id`.
+   * DTMF digits to send automatically after the transfer destination answers. Useful
+   * for reaching an extension behind an IVR (e.g. `"200"` to dial extension 200 once
+   * the called party picks up). Allowed characters: `0-9`, `A-D`, `w` (0.5s pause),
+   * `W` (1s pause), `*`, `#`. Maximum 64 characters. When omitted, no automatic DTMF
+   * is sent. May also be supplied inline by appending `,<digits>` to `to` (e.g.
+   * `to=+18004247767,200`); if both forms are present, this explicit field takes
+   * precedence.
    */
-  command_id?: string;
+  send_digits_on_answer?: string;
 
   /**
-   * Identifies the stream. If the `stream_id` is not provided the command stops all
-   * streams associated with a given `call_control_id`.
+   * SIP Authentication password used for SIP challenges.
    */
-  stream_id?: string;
-}
+  sip_auth_password?: string;
 
-export interface ActionStopTranscriptionParams {
   /**
-   * Use this field to add state to every subsequent webhook. It must be a valid
-   * Base-64 encoded string.
+   * SIP Authentication username used for SIP challenges.
    */
-  client_state?: string;
+  sip_auth_username?: string;
 
   /**
-   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
-   * the same `command_id` for the same `call_control_id`.
+   * SIP headers to be added to the SIP INVITE. Currently only User-to-User header is
+   * supported.
    */
-  command_id?: string;
-}
+  sip_headers?: Array<CallsAPI.SipHeader>;
 
-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.
+   * Defines the SIP region to be used for the call.
    */
-  role: 'barge' | 'whisper' | 'monitor';
-}
+  sip_region?: 'US' | 'Europe' | 'Canada' | 'Australia' | 'Middle East';
 
-export interface ActionTransferParams {
   /**
-   * The DID or SIP URI to dial out to. For SIP URI destinations, append
-   * `;secure=true` or `;secure=srtp` to enable SRTP media encryption for that
-   * endpoint, or `;secure=dtls` to enable DTLS media encryption for that endpoint.
-   * If `media_encryption` is set to `SRTP` or `DTLS`, it takes precedence over any
-   * per-endpoint `secure` URI parameter. You may also append a comma followed by
-   * DTMF digits (e.g. `+18004247767,200`) to play those digits as DTMF once the
-   * transfer destination answers — equivalent to setting `send_digits_on_answer`
-   * separately. If both are present, the explicit `send_digits_on_answer` parameter
-   * takes precedence.
+   * Defines SIP transport protocol to be used on the call.
    */
-  to: string;
+  sip_transport_protocol?: 'UDP' | 'TCP' | 'TLS';
 
   /**
-   * Enables Answering Machine Detection. When a call is answered, Telnyx runs
-   * real-time detection to determine if it was picked up by a human or a machine and
-   * sends an `call.machine.detection.ended` webhook with the analysis result. If
-   * 'greeting_end' or 'detect_words' is used and a 'machine' is detected, you will
-   * receive another 'call.machine.greeting.ended' webhook when the answering machine
-   * greeting ends with a beep or silence. If `detect_beep` is used, you will only
-   * receive 'call.machine.greeting.ended' if a beep is detected.
+   * Use this field to modify sound effects, for example adjust the pitch.
    */
-  answering_machine_detection?:
-    | 'premium'
-    | 'detect'
-    | 'detect_beep'
-    | 'detect_words'
-    | 'greeting_end'
-    | 'disabled';
+  sound_modifications?: CallsAPI.SoundModifications;
 
   /**
-   * Optional configuration parameters to modify 'answering_machine_detection'
-   * performance. Only `total_analysis_time_millis` and `greeting_duration_millis`
-   * parameters are applicable when `premium` is selected as
-   * answering_machine_detection.
+   * Use this field to add state to every subsequent webhook for the new leg. It must
+   * be a valid Base-64 encoded string.
    */
-  answering_machine_detection_config?: ActionTransferParams.AnsweringMachineDetectionConfig;
+  target_leg_client_state?: string;
 
   /**
-   * The URL of a file to be played back when the transfer destination answers before
-   * bridging the call. The URL can point to either a WAV or MP3 file. media_name and
-   * audio_url cannot be used together in one request.
+   * Sets the maximum duration of a Call Control Leg in seconds. If the time limit is
+   * reached, the call will hangup and a `call.hangup` webhook with a `hangup_cause`
+   * of `time_limit` will be sent. For example, by setting a time limit of 120
+   * seconds, a Call Leg will be automatically terminated two minutes after being
+   * answered. The default time limit is 14400 seconds or 4 hours and this is also
+   * the maximum allowed call length.
    */
-  audio_url?: string;
+  time_limit_secs?: number;
 
   /**
-   * Use this field to add state to every subsequent webhook. It must be a valid
-   * Base-64 encoded string.
+   * The number of seconds that Telnyx will wait for the call to be answered by the
+   * destination to which it is being transferred. If the timeout is reached before
+   * an answer is received, the call will hangup and a `call.hangup` webhook with a
+   * `hangup_cause` of `timeout` will be sent. Minimum value is 5 seconds. Maximum
+   * value is 600 seconds.
    */
-  client_state?: string;
+  timeout_secs?: number;
 
   /**
-   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
-   * the same `command_id` for the same `call_control_id`.
+   * A map of event types to retry policies. Each retry policy contains an array of
+   * `retries_ms` specifying the delays between retry attempts in milliseconds.
+   * Maximum 5 retries, total delay cannot exceed 60 seconds.
    */
-  command_id?: string;
+  webhook_retries_policies?: { [key: string]: ActionTransferParams.WebhookRetriesPolicies };
 
   /**
-   * Custom headers to be added to the SIP INVITE.
+   * Use this field to override the URL for which Telnyx will send subsequent
+   * webhooks to for this call.
    */
-  custom_headers?: Array<CallsAPI.CustomSipHeader>;
+  webhook_url?: string;
 
   /**
-   * If set to false, early media will not be passed to the originating leg.
+   * HTTP request type used for `webhook_url`.
    */
-  early_media?: boolean;
+  webhook_url_method?: 'POST' | 'GET';
 
   /**
-   * The `from` number to be used as the caller id presented to the destination (`to`
-   * number). The number should be in +E164 format. This attribute will default to
-   * the `to` number of the original call if omitted.
+   * A map of event types to webhook URLs. When an event of the specified type
+   * occurs, the webhook URL associated with that event type will be called instead
+   * of `webhook_url`. Events not mapped here will use the default `webhook_url`.
    */
-  from?: string;
+  webhook_urls?: { [key: string]: string };
 
   /**
-   * The `from_display_name` string to be used as the caller id name (SIP From
-   * Display Name) presented to the destination (`to` number). The string should have
-   * a maximum of 128 characters, containing only letters, numbers, spaces, and
-   * -\_~!.+ special characters. If ommited, the display name will be the same as the
-   * number in the `from` field.
+   * HTTP request method to invoke `webhook_urls`.
    */
-  from_display_name?: string;
+  webhook_urls_method?: 'POST' | 'GET';
+}
 
+export namespace ActionTransferParams {
   /**
-   * Defines whether media should be encrypted on the new call leg. For SIP URI
-   * destinations, media encryption can also be requested per endpoint with the
-   * `secure` URI parameter: `;secure=true` or `;secure=srtp` enables SRTP, and
-   * `;secure=dtls` enables DTLS. This parameter, when set to `SRTP` or `DTLS`, takes
-   * precedence over the per-endpoint `secure` value.
+   * Optional configuration parameters to modify 'answering_machine_detection'
+   * performance. Only `total_analysis_time_millis` and `greeting_duration_millis`
+   * parameters are applicable when `premium` is selected as
+   * answering_machine_detection.
    */
-  media_encryption?: 'disabled' | 'SRTP' | 'DTLS';
+  export interface AnsweringMachineDetectionConfig {
+    /**
+     * Silence duration threshold after a greeting message or voice for it be
+     * considered human.
+     */
+    after_greeting_silence_millis?: number;
+
+    /**
+     * Maximum threshold for silence between words.
+     */
+    between_words_silence_millis?: number;
+
+    /**
+     * Maximum threshold of a human greeting. If greeting longer than this value,
+     * considered machine.
+     */
+    greeting_duration_millis?: number;
+
+    /**
+     * If machine already detected, maximum threshold for silence between words. If
+     * exceeded, the greeting is considered ended.
+     */
+    greeting_silence_duration_millis?: number;
+
+    /**
+     * If machine already detected, maximum timeout threshold to determine the end of
+     * the machine greeting.
+     */
+    greeting_total_analysis_time_millis?: number;
+
+    /**
+     * If initial silence duration is greater than this value, consider it a machine.
+     */
+    initial_silence_millis?: number;
+
+    /**
+     * If number of detected words is greater than this value, consder it a machine.
+     */
+    maximum_number_of_words?: number;
+
+    /**
+     * If a single word lasts longer than this threshold, consider it a machine.
+     */
+    maximum_word_length_millis?: number;
+
+    /**
+     * Minimum noise threshold for any analysis.
+     */
+    silence_threshold?: number;
+
+    /**
+     * Maximum timeout threshold for overall detection.
+     */
+    total_analysis_time_millis?: number;
+  }
 
+  export interface WebhookRetriesPolicies {
+    /**
+     * Array of delays in milliseconds between retry attempts. Total sum cannot exceed
+     * 60000ms.
+     */
+    retries_ms?: Array<number>;
+  }
+}
+
+export interface ActionAddAIAssistantMessagesParams {
   /**
-   * The media_name of a file to be played back when the transfer destination answers
-   * before bridging the call. The media_name must point to a file previously
-   * uploaded to api.telnyx.com/v2/media by the same user/organization. The file must
-   * either be a WAV or MP3 file.
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
    */
-  media_name?: string;
+  client_state?: string;
 
   /**
-   * When enabled, DTMF tones are not passed to the call participant. The webhooks
-   * containing the DTMF information will be sent.
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
    */
-  mute_dtmf?: 'none' | 'both' | 'self' | 'opposite';
+  command_id?: string;
 
   /**
-   * Specifies behavior after the bridge ends (i.e. the opposite leg either hangs up
-   * or is transferred). If supplied with the value `self`, the current leg will be
-   * parked after unbridge. If not set, the default behavior is to hang up the leg.
+   * The messages to add to the conversation.
    */
-  park_after_unbridge?: string;
+  messages?: Array<UserMessage | AssistantMessage | ToolMessage | SystemMessage | DeveloperMessage>;
+}
 
+export interface ActionJoinAIAssistantParams {
   /**
-   * The list of comma-separated codecs in order of preference to be used during the
-   * call. The codecs supported are `G722`, `PCMU`, `PCMA`, `G729`, `OPUS`, `VP8`,
-   * `H264`, `AMR-WB`.
+   * The ID of the AI assistant conversation to join.
    */
-  preferred_codecs?: string;
+  conversation_id: string;
+
+  participant: AIAssistantJoinParticipant;
 
   /**
-   * Indicates the privacy level to be used for the call. When set to `id`, caller ID
-   * information (name and number) will be hidden from the called party. When set to
-   * `none` or omitted, caller ID will be shown normally.
+   * Use this field to add state to every subsequent webhook. It must be a valid
+   * Base-64 encoded string.
    */
-  privacy?: 'id' | 'none';
+  client_state?: string;
 
   /**
-   * Start recording automatically after an event. Disabled by default.
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
    */
-  record?: 'record-from-answer';
+  command_id?: string;
+}
 
+export interface ActionStartConversationRelayParams {
   /**
-   * Defines which channel should be recorded ('single' or 'dual') when `record` is
-   * specified.
+   * Custom parameters for the Conversation Relay session. Pass key-value data as
+   * `assistant.dynamic_variables` to make it available to the relay session.
    */
-  record_channels?: 'single' | 'dual';
+  assistant?: ActionStartConversationRelayParams.Assistant;
 
   /**
-   * The custom recording file name to be used instead of the default `call_leg_id`.
-   * Telnyx will still add a Unix timestamp suffix.
+   * Use this field to add state to subsequent webhooks. It must be a valid Base-64
+   * encoded string.
    */
-  record_custom_file_name?: string;
+  client_state?: string;
 
   /**
-   * Defines the format of the recording ('wav' or 'mp3') when `record` is specified.
+   * Use this field to avoid duplicate commands. Telnyx will ignore any command with
+   * the same `command_id` for the same `call_control_id`.
    */
-  record_format?: 'wav' | 'mp3';
+  command_id?: string;
 
   /**
-   * Defines the maximum length for the recording in seconds when `record` is
-   * specified. The minimum value is 0. The maximum value is 43200. The default value
-   * is 0 (infinite).
+   * Enable DTMF detection for the relay session.
    */
-  record_max_length?: number;
+  conversation_relay_dtmf_detection?: boolean;
 
   /**
-   * The number of seconds that Telnyx will wait for the recording to be stopped if
-   * silence is detected when `record` is specified. The timer only starts when the
-   * speech is detected. Please note that call transcription is used to detect
-   * silence and the related charge will be applied. The minimum value is 0. The
-   * default value is 0 (infinite).
+   * 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.
    */
-  record_timeout_secs?: number;
+  conversation_relay_settings?: ActionStartConversationRelayParams.ConversationRelaySettings;
 
   /**
-   * The audio track to be recorded. Can be either `both`, `inbound` or `outbound`.
-   * If only single track is specified (`inbound`, `outbound`), `channels`
-   * configuration is ignored and it will be recorded as mono (single channel).
+   * WebSocket URL for your Conversation Relay server. Must start with `ws://` or
+   * `wss://`.
    */
-  record_track?: 'both' | 'inbound' | 'outbound';
+  conversation_relay_url?: string;
 
   /**
-   * When set to `trim-silence`, silence will be removed from the beginning and end
-   * of the recording.
+   * 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.
    */
-  record_trim?: 'trim-silence';
+  custom_parameters?: { [key: string]: unknown };
 
   /**
-   * DTMF digits to send automatically after the transfer destination answers. Useful
-   * for reaching an extension behind an IVR (e.g. `"200"` to dial extension 200 once
-   * the called party picks up). Allowed characters: `0-9`, `A-D`, `w` (0.5s pause),
-   * `W` (1s pause), `*`, `#`. Maximum 64 characters. When omitted, no automatic DTMF
-   * is sent. May also be supplied inline by appending `,<digits>` to `to` (e.g.
-   * `to=+18004247767,200`); if both forms are present, this explicit field takes
-   * precedence.
+   * Public alias for `conversation_relay_dtmf_detection`. If both are present, this
+   * value wins.
    */
-  send_digits_on_answer?: string;
+  dtmf_detection?: boolean;
 
   /**
-   * SIP Authentication password used for SIP challenges.
+   * Text played when the relay session starts.
    */
-  sip_auth_password?: string;
+  greeting?: string;
 
   /**
-   * SIP Authentication username used for SIP challenges.
+   * 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.
    */
-  sip_auth_username?: string;
+  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;
 
   /**
-   * SIP headers to be added to the SIP INVITE. Currently only User-to-User header is
-   * supported.
+   * Settings for handling caller interruptions during Conversation Relay speech.
    */
-  sip_headers?: Array<CallsAPI.SipHeader>;
+  interruption_settings?: CallsAPI.ConversationRelayInterruptionSettings;
 
   /**
-   * Defines the SIP region to be used for the call.
+   * Default language for the relay session. This value is used for both
+   * text-to-speech and speech recognition.
    */
-  sip_region?: 'US' | 'Europe' | 'Canada' | 'Australia' | 'Middle East';
+  language?: string;
 
   /**
-   * Defines SIP transport protocol to be used on the call.
+   * Per-language TTS and transcription settings.
    */
-  sip_transport_protocol?: 'UDP' | 'TCP' | 'TLS';
+  languages?: Array<CallsAPI.ConversationRelayLanguage>;
 
   /**
-   * Use this field to modify sound effects, for example adjust the pitch.
+   * Structured voice provider. Must be supplied together with `structured_provider`.
    */
-  sound_modifications?: CallsAPI.SoundModifications;
+  provider?: string;
 
   /**
-   * Use this field to add state to every subsequent webhook for the new leg. It must
-   * be a valid Base-64 encoded string.
+   * Provider-specific structured voice settings. Must be supplied together with
+   * `provider`; Telnyx sends the value as the nested provider configuration for
+   * Conversation Relay.
    */
-  target_leg_client_state?: string;
+  structured_provider?: { [key: string]: unknown };
 
   /**
-   * Sets the maximum duration of a Call Control Leg in seconds. If the time limit is
-   * reached, the call will hangup and a `call.hangup` webhook with a `hangup_cause`
-   * of `time_limit` will be sent. For example, by setting a time limit of 120
-   * seconds, a Call Leg will be automatically terminated two minutes after being
-   * answered. The default time limit is 14400 seconds or 4 hours and this is also
-   * the maximum allowed call length.
+   * @deprecated Not supported for Conversation Relay start requests. Use
+   * `transcription_engine` and `transcription_engine_config` instead.
    */
-  time_limit_secs?: number;
+  transcription?: { [key: string]: unknown };
 
   /**
-   * The number of seconds that Telnyx will wait for the call to be answered by the
-   * destination to which it is being transferred. If the timeout is reached before
-   * an answer is received, the call will hangup and a `call.hangup` webhook with a
-   * `hangup_cause` of `timeout` will be sent. Minimum value is 5 seconds. Maximum
-   * value is 600 seconds.
+   * 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.
    */
-  timeout_secs?: number;
+  transcription_engine?:
+    | 'Google'
+    | 'Telnyx'
+    | 'Deepgram'
+    | 'Azure'
+    | 'xAI'
+    | 'AssemblyAI'
+    | 'Speechmatics'
+    | 'Soniox'
+    | 'A'
+    | 'B';
 
   /**
-   * A map of event types to retry policies. Each retry policy contains an array of
-   * `retries_ms` specifying the delays between retry attempts in milliseconds.
-   * Maximum 5 retries, total delay cannot exceed 60 seconds.
+   * 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.
    */
-  webhook_retries_policies?: { [key: string]: ActionTransferParams.WebhookRetriesPolicies };
+  transcription_engine_config?: { [key: string]: unknown };
 
   /**
-   * Use this field to override the URL for which Telnyx will send subsequent
-   * webhooks to for this call.
+   * Text-to-speech provider. If omitted, Telnyx derives it from `voice` or
+   * `provider`.
    */
-  webhook_url?: string;
+  tts_provider?: string;
 
   /**
-   * HTTP request type used for `webhook_url`.
+   * Public alias for `conversation_relay_url`. Must start with `ws://` or `wss://`.
+   * If both are present, this value wins.
    */
-  webhook_url_method?: 'POST' | 'GET';
+  url?: string;
 
   /**
-   * A map of event types to webhook URLs. When an event of the specified type
-   * occurs, the webhook URL associated with that event type will be called instead
-   * of `webhook_url`. Events not mapped here will use the default `webhook_url`.
+   * 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`.
    */
-  webhook_urls?: { [key: string]: string };
+  voice?: string;
 
   /**
-   * HTTP request method to invoke `webhook_urls`.
+   * The settings associated with the voice selected
    */
-  webhook_urls_method?: 'POST' | 'GET';
+  voice_settings?:
+    | ElevenLabsVoiceSettings
+    | TelnyxVoiceSettings
+    | AwsVoiceSettings
+    | Shared.MinimaxVoiceSettings
+    | Shared.AzureVoiceSettings
+    | Shared.RimeVoiceSettings
+    | Shared.ResembleVoiceSettings
+    | Shared.InworldVoiceSettings
+    | Shared.XaiVoiceSettings;
 }
 
-export namespace ActionTransferParams {
+export namespace ActionStartConversationRelayParams {
   /**
-   * Optional configuration parameters to modify 'answering_machine_detection'
-   * performance. Only `total_analysis_time_millis` and `greeting_duration_millis`
-   * parameters are applicable when `premium` is selected as
-   * answering_machine_detection.
+   * Custom parameters for the Conversation Relay session. Pass key-value data as
+   * `assistant.dynamic_variables` to make it available to the relay session.
    */
-  export interface AnsweringMachineDetectionConfig {
-    /**
-     * Silence duration threshold after a greeting message or voice for it be
-     * considered human.
-     */
-    after_greeting_silence_millis?: number;
-
-    /**
-     * Maximum threshold for silence between words.
-     */
-    between_words_silence_millis?: number;
-
-    /**
-     * Maximum threshold of a human greeting. If greeting longer than this value,
-     * considered machine.
-     */
-    greeting_duration_millis?: number;
-
-    /**
-     * If machine already detected, maximum threshold for silence between words. If
-     * exceeded, the greeting is considered ended.
-     */
-    greeting_silence_duration_millis?: number;
-
+  export interface Assistant {
     /**
-     * If machine already detected, maximum timeout threshold to determine the end of
-     * the machine greeting.
+     * Custom key-value parameters forwarded to the Conversation Relay session.
      */
-    greeting_total_analysis_time_millis?: number;
+    dynamic_variables?: { [key: string]: string };
 
-    /**
-     * If initial silence duration is greater than this value, consider it a machine.
-     */
-    initial_silence_millis?: number;
+    [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.
+   */
+  export interface ConversationRelaySettings {
     /**
-     * If number of detected words is greater than this value, consder it a machine.
+     * WebSocket URL for your Conversation Relay server. Must start with `ws://` or
+     * `wss://`.
      */
-    maximum_number_of_words?: number;
+    url: string;
 
     /**
-     * If a single word lasts longer than this threshold, consider it a machine.
+     * Whether to enable DTMF detection during the relay session.
      */
-    maximum_word_length_millis?: number;
+    dtmf_detection?: boolean;
 
     /**
-     * Minimum noise threshold for any analysis.
+     * 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.
      */
-    silence_threshold?: number;
+    interruptible?: ActionsAPI.ConversationRelayInterruptible;
 
     /**
-     * Maximum timeout threshold for overall detection.
+     * 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.
      */
-    total_analysis_time_millis?: number;
-  }
+    interruptible_greeting?: ActionsAPI.ConversationRelayInterruptible;
 
-  export interface WebhookRetriesPolicies {
     /**
-     * Array of delays in milliseconds between retry attempts. Total sum cannot exceed
-     * 60000ms.
+     * Language-specific TTS and transcription settings.
      */
-    retries_ms?: Array<number>;
+    languages?: Array<CallsAPI.ConversationRelayLanguage>;
   }
 }
 
-export interface ActionUpdateClientStateParams {
+export interface ActionStopConversationRelayParams {
   /**
-   * Use this field to add state to every subsequent webhook. It must be a valid
-   * Base-64 encoded string.
+   * Use this field to add state to subsequent webhooks. 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;
 }
 
 export declare namespace Actions {
@@ -5279,45 +5279,45 @@ export declare namespace Actions {
     type ActionSwitchSupervisorRoleResponse as ActionSwitchSupervisorRoleResponse,
     type ActionTransferResponse as ActionTransferResponse,
     type ActionUpdateClientStateResponse as ActionUpdateClientStateResponse,
-    type ActionAddAIAssistantMessagesParams as ActionAddAIAssistantMessagesParams,
+    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 ActionJoinAIAssistantParams as ActionJoinAIAssistantParams,
     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 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 ActionSpeakParams as ActionSpeakParams,
+    type ActionStartStreamingParams as ActionStartStreamingParams,
     type ActionStopStreamingParams as ActionStopStreamingParams,
-    type ActionStopTranscriptionParams as ActionStopTranscriptionParams,
+    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 ActionUpdateClientStateParams as ActionUpdateClientStateParams,
+    type ActionAddAIAssistantMessagesParams as ActionAddAIAssistantMessagesParams,
+    type ActionJoinAIAssistantParams as ActionJoinAIAssistantParams,
+    type ActionStartConversationRelayParams as ActionStartConversationRelayParams,
+    type ActionStopConversationRelayParams as ActionStopConversationRelayParams,
   };
 }