retell-sdk 5.23.0 → 5.25.0

package/resources/call.d.mts

@@ -1,5 +1,4 @@
 import { APIResource } from "../core/resource.mjs";
-import * as CallAPI from "./call.mjs";
 import { APIPromise } from "../core/api-promise.mjs";
 import { RequestOptions } from "../internal/request-options.mjs";
 export declare class Call extends APIResource {
@@ -158,7 +157,7 @@ export interface PhoneCallResponse {
      * Dynamic variables collected from the call. Only available after the call ends.
      */
     collected_dynamic_variables?: {
-        [key: string]: unknown;
+        [key: string]: string;
     };
     /**
      * Custom SIP headers to be added to the call.
@@ -237,7 +236,7 @@ export interface PhoneCallResponse {
      * Engine.
      */
     retell_llm_dynamic_variables?: {
-        [key: string]: unknown;
+        [key: string]: string;
     };
     /**
      * Recording of the call without PII, with each party's audio stored in a separate
@@ -1027,7 +1026,7 @@ export interface WebCallResponse {
      * Dynamic variables collected from the call. Only available after the call ends.
      */
     collected_dynamic_variables?: {
-        [key: string]: unknown;
+        [key: string]: string;
     };
     /**
      * Custom SIP headers to be added to the call.
@@ -1106,7 +1105,7 @@ export interface WebCallResponse {
      * Engine.
      */
     retell_llm_dynamic_variables?: {
-        [key: string]: unknown;
+        [key: string]: string;
     };
     /**
      * Recording of the call without PII, with each party's audio stored in a separate
@@ -1750,107 +1749,1257 @@ export declare namespace WebCallResponse {
     }
     interface ToolCallInvocationUtterance {
         /**
-         * Arguments for this tool call, it's a stringified JSON object.
+         * Arguments for this tool call, it's a stringified JSON object.
+         */
+        arguments: string;
+        /**
+         * Name of the function in this tool call.
+         */
+        name: string;
+        /**
+         * This is a tool call invocation.
+         */
+        role: 'tool_call_invocation';
+        /**
+         * Tool call id, globally unique.
+         */
+        tool_call_id: string;
+        /**
+         * Optional thought signature from Google Gemini thinking models. This is used
+         * internally to maintain reasoning chain in multi-turn function calling.
+         */
+        thought_signature?: string;
+    }
+    interface ToolCallResultUtterance {
+        /**
+         * Result of the tool call, can be a string, a stringified json, etc.
+         */
+        content: string;
+        /**
+         * This is the result of a tool call.
+         */
+        role: 'tool_call_result';
+        /**
+         * Tool call id, globally unique.
+         */
+        tool_call_id: string;
+        /**
+         * Whether the tool call was successful.
+         */
+        successful?: boolean;
+    }
+    interface NodeTransitionUtterance {
+        /**
+         * Former node id
+         */
+        former_node_id: string;
+        /**
+         * Former node name
+         */
+        former_node_name: string;
+        /**
+         * New node id
+         */
+        new_node_id: string;
+        /**
+         * New node name
+         */
+        new_node_name: string;
+        /**
+         * This is result of a node transition
+         */
+        role: 'node_transition';
+        /**
+         * How this node was reached. "global" means a global node transition,
+         * "global_go_back" means returning from a global node, "interrupt_go_back" means
+         * going back due to user interruption, and "normal" means a regular edge
+         * transition.
+         */
+        transition_type?: 'global' | 'global_go_back' | 'interrupt_go_back' | 'normal';
+    }
+    interface DtmfUtterance {
+        /**
+         * The digit pressed by the user. Will be a single digit string like "1", "2", "3",
+         * "\*", "#" etc.
+         */
+        digit: string;
+        /**
+         * Digit pressed by the user from their phone keypad.
+         */
+        role: 'dtmf';
+    }
+}
+export interface CallListResponse {
+    /**
+     * Whether more results are available.
+     */
+    has_more?: boolean;
+    items?: Array<CallListResponse.V3WebCallResponse | CallListResponse.V3PhoneCallResponse>;
+    /**
+     * Pagination key for the next page.
+     */
+    pagination_key?: string;
+}
+export declare namespace CallListResponse {
+    interface V3WebCallResponse {
+        /**
+         * Access token to enter the web call room. This needs to be passed to your
+         * frontend to join the call.
+         */
+        access_token: string;
+        /**
+         * Corresponding agent id of this call.
+         */
+        agent_id: string;
+        /**
+         * The version of the agent.
+         */
+        agent_version: number;
+        /**
+         * Unique id of the call. Used to identify the call in the LLM websocket and used
+         * to authenticate in the audio websocket.
+         */
+        call_id: string;
+        /**
+         * Status of call.
+         *
+         * - `registered`: Call id issued, starting to make a call using this id.
+         * - `ongoing`: Call connected and ongoing.
+         * - `ended`: The underlying websocket has ended for the call. Either user or agent
+         *   hung up, or call transferred.
+         * - `error`: Call encountered error.
+         */
+        call_status: 'registered' | 'not_connected' | 'ongoing' | 'ended' | 'error';
+        /**
+         * Type of the call. Used to distinguish between web call and phone call.
+         */
+        call_type: 'web_call';
+        /**
+         * Name of the agent.
+         */
+        agent_name?: string;
+        /**
+         * Post call analysis that includes information such as sentiment, status, summary,
+         * and custom defined data to extract. Available after call ends. Subscribe to
+         * `call_analyzed` webhook event type to receive it once ready.
+         */
+        call_analysis?: V3WebCallResponse.CallAnalysis;
+        /**
+         * Cost of the call, including all the products and their costs and discount.
+         */
+        call_cost?: V3WebCallResponse.CallCost;
+        /**
+         * Dynamic variables collected from the call. Only available after the call ends.
+         */
+        collected_dynamic_variables?: {
+            [key: string]: string;
+        };
+        /**
+         * Custom SIP headers to be added to the call.
+         */
+        custom_sip_headers?: {
+            [key: string]: string;
+        };
+        /**
+         * Data storage setting for this call's agent. "everything" stores all data,
+         * "everything_except_pii" excludes PII when possible, "basic_attributes_only"
+         * stores only metadata.
+         */
+        data_storage_setting?: 'everything' | 'everything_except_pii' | 'basic_attributes_only' | null;
+        /**
+         * The reason for the disconnection of the call. Read detailed description about
+         * reasons listed here at
+         * [Disconnection Reason Doc](/reliability/debug-call-disconnect#understanding-disconnection-reasons).
+         */
+        disconnection_reason?: 'user_hangup' | 'agent_hangup' | 'call_transfer' | 'voicemail_reached' | 'ivr_reached' | 'inactivity' | 'max_duration_reached' | 'concurrency_limit_reached' | 'no_valid_payment' | 'scam_detected' | 'dial_busy' | 'dial_failed' | 'dial_no_answer' | 'invalid_destination' | 'telephony_provider_permission_denied' | 'telephony_provider_unavailable' | 'sip_routing_error' | 'marked_as_spam' | 'user_declined' | 'error_llm_websocket_open' | 'error_llm_websocket_lost_connection' | 'error_llm_websocket_runtime' | 'error_llm_websocket_corrupt_payload' | 'error_no_audio_received' | 'error_asr' | 'error_retell' | 'error_unknown' | 'error_user_not_joined' | 'registered_call_timeout' | 'transfer_bridged' | 'transfer_cancelled' | 'manual_stopped';
+        /**
+         * Duration of the call in milliseconds. Available after call ends.
+         */
+        duration_ms?: number;
+        /**
+         * End timestamp (milliseconds since epoch) of the call. Available after call ends.
+         */
+        end_timestamp?: number;
+        /**
+         * URL to the knowledge base retrieved contents of the call. Available after call
+         * ends if the call utilizes knowledge base feature. It consists of the respond id
+         * and the retrieved contents related to that response. It's already rendered in
+         * call history tab of dashboard, and you can also manually download and check
+         * against the transcript to view the knowledge base retrieval results.
+         */
+        knowledge_base_retrieved_contents_url?: string;
+        /**
+         * Latency tracking of the call, available after call ends. Not all fields here
+         * will be available, as it depends on the type of call and feature used.
+         */
+        latency?: V3WebCallResponse.Latency;
+        /**
+         * LLM token usage of the call, available after call ends. Not populated if using
+         * custom LLM, realtime API, or no LLM call is made.
+         */
+        llm_token_usage?: V3WebCallResponse.LlmTokenUsage;
+        /**
+         * An arbitrary object for storage purpose only. You can put anything here like
+         * your internal customer id associated with the call. Not used for processing. You
+         * can later get this field from the call object.
+         */
+        metadata?: unknown;
+        /**
+         * Whether this agent opts in for signed URLs for public logs and recordings. When
+         * enabled, the generated URLs will include security signatures that restrict
+         * access and automatically expire after 24 hours.
+         */
+        opt_in_signed_url?: boolean;
+        /**
+         * Public log of the call, containing details about all the requests and responses
+         * received in LLM WebSocket, latency tracking for each turntaking, helpful for
+         * debugging and tracing. Available after call ends.
+         */
+        public_log_url?: string;
+        /**
+         * Recording of the call, with each party's audio stored in a separate channel.
+         * Available after the call ends.
+         */
+        recording_multi_channel_url?: string;
+        /**
+         * Recording of the call. Available after call ends.
+         */
+        recording_url?: string;
+        /**
+         * Add optional dynamic variables in key value pairs of string that injects into
+         * your Response Engine prompt and tool description. Only applicable for Response
+         * Engine.
+         */
+        retell_llm_dynamic_variables?: {
+            [key: string]: string;
+        };
+        /**
+         * Recording of the call without PII, with each party's audio stored in a separate
+         * channel. Available after the call ends.
+         */
+        scrubbed_recording_multi_channel_url?: string;
+        /**
+         * Recording of the call without PII. Available after call ends.
+         */
+        scrubbed_recording_url?: string;
+        /**
+         * Begin timestamp (milliseconds since epoch) of the call. Available after call
+         * starts.
+         */
+        start_timestamp?: number;
+        /**
+         * The destination number or identifier where the call was transferred to. Only
+         * populated when the disconnection reason was `call_transfer`. Can be a phone
+         * number or a SIP URI. SIP URIs are prefixed with "sip:" and may include a
+         * ";transport=..." portion (if transport is known) where the transport type can be
+         * "tls", "tcp" or "udp".
+         */
+        transfer_destination?: string | null;
+        /**
+         * Transfer end timestamp (milliseconds since epoch) of the call. Available after
+         * transfer call ends.
+         */
+        transfer_end_timestamp?: number;
+    }
+    namespace V3WebCallResponse {
+        /**
+         * Post call analysis that includes information such as sentiment, status, summary,
+         * and custom defined data to extract. Available after call ends. Subscribe to
+         * `call_analyzed` webhook event type to receive it once ready.
+         */
+        interface CallAnalysis {
+            /**
+             * Whether the agent seems to have a successful call with the user, where the agent
+             * finishes the task, and the call was complete without being cutoff.
+             */
+            call_successful?: boolean;
+            /**
+             * A high level summary of the call.
+             */
+            call_summary?: string;
+            /**
+             * Custom analysis data that was extracted based on the schema defined in agent
+             * post call analysis data. Can be empty if nothing is specified.
+             */
+            custom_analysis_data?: unknown;
+            /**
+             * Whether the call is entered voicemail.
+             */
+            in_voicemail?: boolean;
+            /**
+             * Sentiment of the user in the call.
+             */
+            user_sentiment?: 'Negative' | 'Positive' | 'Neutral' | 'Unknown';
+        }
+        /**
+         * Cost of the call, including all the products and their costs and discount.
+         */
+        interface CallCost {
+            /**
+             * Combined cost of all individual costs in cents
+             */
+            combined_cost: number;
+            /**
+             * List of products with their unit prices and costs in cents
+             */
+            product_costs: Array<CallCost.ProductCost>;
+            /**
+             * Total duration of the call in seconds
+             */
+            total_duration_seconds: number;
+            /**
+             * Total unit duration price of all products in cents per second
+             */
+            total_duration_unit_price: number;
+        }
+        namespace CallCost {
+            interface ProductCost {
+                /**
+                 * Cost for the product in cents for the duration of the call.
+                 */
+                cost: number;
+                /**
+                 * Product name that has a cost associated with it.
+                 */
+                product: string;
+                /**
+                 * True if this cost item is for a transfer segment.
+                 */
+                is_transfer_leg_cost?: boolean;
+                /**
+                 * Unit price of the product in cents per second.
+                 */
+                unit_price?: number;
+            }
+        }
+        /**
+         * Latency tracking of the call, available after call ends. Not all fields here
+         * will be available, as it depends on the type of call and feature used.
+         */
+        interface Latency {
+            /**
+             * Transcription latency (diff between the duration of the chunks streamed and the
+             * durations of the transcribed part) tracking of the call.
+             */
+            asr?: Latency.Asr;
+            /**
+             * End to end latency (from user stops talking to agent start talking) tracking of
+             * the call. This latency does not account for the network trip time from Retell
+             * server to user frontend. The latency is tracked every time turn change between
+             * user and agent.
+             */
+            e2e?: Latency.E2E;
+            /**
+             * Knowledge base latency (from the triggering of knowledge base retrival to all
+             * relevant context received) tracking of the call. Only populated when using
+             * knowledge base feature for the agent of the call.
+             */
+            knowledge_base?: Latency.KnowledgeBase;
+            /**
+             * LLM latency (from issue of LLM call to first speakable chunk received) tracking
+             * of the call. When using custom LLM. this latency includes LLM websocket
+             * roundtrip time between user server and Retell server.
+             */
+            llm?: Latency.Llm;
+            /**
+             * LLM websocket roundtrip latency (between user server and Retell server) tracking
+             * of the call. Only populated for calls using custom LLM.
+             */
+            llm_websocket_network_rtt?: Latency.LlmWebsocketNetworkRtt;
+            /**
+             * Speech-to-speech latency (from requesting responses of a S2S model to first byte
+             * received) tracking of the call. Only populated for calls that uses S2S model
+             * like Realtime API.
+             */
+            s2s?: Latency.S2s;
+            /**
+             * Text-to-speech latency (from the triggering of TTS to first byte received)
+             * tracking of the call.
+             */
+            tts?: Latency.Tts;
+        }
+        namespace Latency {
+            /**
+             * Transcription latency (diff between the duration of the chunks streamed and the
+             * durations of the transcribed part) tracking of the call.
+             */
+            interface Asr {
+                /**
+                 * Maximum latency in the call, measured in milliseconds.
+                 */
+                max?: number;
+                /**
+                 * Minimum latency in the call, measured in milliseconds.
+                 */
+                min?: number;
+                /**
+                 * Number of data points (number of times latency is tracked).
+                 */
+                num?: number;
+                /**
+                 * 50 percentile of latency, measured in milliseconds.
+                 */
+                p50?: number;
+                /**
+                 * 90 percentile of latency, measured in milliseconds.
+                 */
+                p90?: number;
+                /**
+                 * 95 percentile of latency, measured in milliseconds.
+                 */
+                p95?: number;
+                /**
+                 * 99 percentile of latency, measured in milliseconds.
+                 */
+                p99?: number;
+                /**
+                 * All the latency data points in the call, measured in milliseconds.
+                 */
+                values?: Array<number>;
+            }
+            /**
+             * End to end latency (from user stops talking to agent start talking) tracking of
+             * the call. This latency does not account for the network trip time from Retell
+             * server to user frontend. The latency is tracked every time turn change between
+             * user and agent.
+             */
+            interface E2E {
+                /**
+                 * Maximum latency in the call, measured in milliseconds.
+                 */
+                max?: number;
+                /**
+                 * Minimum latency in the call, measured in milliseconds.
+                 */
+                min?: number;
+                /**
+                 * Number of data points (number of times latency is tracked).
+                 */
+                num?: number;
+                /**
+                 * 50 percentile of latency, measured in milliseconds.
+                 */
+                p50?: number;
+                /**
+                 * 90 percentile of latency, measured in milliseconds.
+                 */
+                p90?: number;
+                /**
+                 * 95 percentile of latency, measured in milliseconds.
+                 */
+                p95?: number;
+                /**
+                 * 99 percentile of latency, measured in milliseconds.
+                 */
+                p99?: number;
+                /**
+                 * All the latency data points in the call, measured in milliseconds.
+                 */
+                values?: Array<number>;
+            }
+            /**
+             * Knowledge base latency (from the triggering of knowledge base retrival to all
+             * relevant context received) tracking of the call. Only populated when using
+             * knowledge base feature for the agent of the call.
+             */
+            interface KnowledgeBase {
+                /**
+                 * Maximum latency in the call, measured in milliseconds.
+                 */
+                max?: number;
+                /**
+                 * Minimum latency in the call, measured in milliseconds.
+                 */
+                min?: number;
+                /**
+                 * Number of data points (number of times latency is tracked).
+                 */
+                num?: number;
+                /**
+                 * 50 percentile of latency, measured in milliseconds.
+                 */
+                p50?: number;
+                /**
+                 * 90 percentile of latency, measured in milliseconds.
+                 */
+                p90?: number;
+                /**
+                 * 95 percentile of latency, measured in milliseconds.
+                 */
+                p95?: number;
+                /**
+                 * 99 percentile of latency, measured in milliseconds.
+                 */
+                p99?: number;
+                /**
+                 * All the latency data points in the call, measured in milliseconds.
+                 */
+                values?: Array<number>;
+            }
+            /**
+             * LLM latency (from issue of LLM call to first speakable chunk received) tracking
+             * of the call. When using custom LLM. this latency includes LLM websocket
+             * roundtrip time between user server and Retell server.
+             */
+            interface Llm {
+                /**
+                 * Maximum latency in the call, measured in milliseconds.
+                 */
+                max?: number;
+                /**
+                 * Minimum latency in the call, measured in milliseconds.
+                 */
+                min?: number;
+                /**
+                 * Number of data points (number of times latency is tracked).
+                 */
+                num?: number;
+                /**
+                 * 50 percentile of latency, measured in milliseconds.
+                 */
+                p50?: number;
+                /**
+                 * 90 percentile of latency, measured in milliseconds.
+                 */
+                p90?: number;
+                /**
+                 * 95 percentile of latency, measured in milliseconds.
+                 */
+                p95?: number;
+                /**
+                 * 99 percentile of latency, measured in milliseconds.
+                 */
+                p99?: number;
+                /**
+                 * All the latency data points in the call, measured in milliseconds.
+                 */
+                values?: Array<number>;
+            }
+            /**
+             * LLM websocket roundtrip latency (between user server and Retell server) tracking
+             * of the call. Only populated for calls using custom LLM.
+             */
+            interface LlmWebsocketNetworkRtt {
+                /**
+                 * Maximum latency in the call, measured in milliseconds.
+                 */
+                max?: number;
+                /**
+                 * Minimum latency in the call, measured in milliseconds.
+                 */
+                min?: number;
+                /**
+                 * Number of data points (number of times latency is tracked).
+                 */
+                num?: number;
+                /**
+                 * 50 percentile of latency, measured in milliseconds.
+                 */
+                p50?: number;
+                /**
+                 * 90 percentile of latency, measured in milliseconds.
+                 */
+                p90?: number;
+                /**
+                 * 95 percentile of latency, measured in milliseconds.
+                 */
+                p95?: number;
+                /**
+                 * 99 percentile of latency, measured in milliseconds.
+                 */
+                p99?: number;
+                /**
+                 * All the latency data points in the call, measured in milliseconds.
+                 */
+                values?: Array<number>;
+            }
+            /**
+             * Speech-to-speech latency (from requesting responses of a S2S model to first byte
+             * received) tracking of the call. Only populated for calls that uses S2S model
+             * like Realtime API.
+             */
+            interface S2s {
+                /**
+                 * Maximum latency in the call, measured in milliseconds.
+                 */
+                max?: number;
+                /**
+                 * Minimum latency in the call, measured in milliseconds.
+                 */
+                min?: number;
+                /**
+                 * Number of data points (number of times latency is tracked).
+                 */
+                num?: number;
+                /**
+                 * 50 percentile of latency, measured in milliseconds.
+                 */
+                p50?: number;
+                /**
+                 * 90 percentile of latency, measured in milliseconds.
+                 */
+                p90?: number;
+                /**
+                 * 95 percentile of latency, measured in milliseconds.
+                 */
+                p95?: number;
+                /**
+                 * 99 percentile of latency, measured in milliseconds.
+                 */
+                p99?: number;
+                /**
+                 * All the latency data points in the call, measured in milliseconds.
+                 */
+                values?: Array<number>;
+            }
+            /**
+             * Text-to-speech latency (from the triggering of TTS to first byte received)
+             * tracking of the call.
+             */
+            interface Tts {
+                /**
+                 * Maximum latency in the call, measured in milliseconds.
+                 */
+                max?: number;
+                /**
+                 * Minimum latency in the call, measured in milliseconds.
+                 */
+                min?: number;
+                /**
+                 * Number of data points (number of times latency is tracked).
+                 */
+                num?: number;
+                /**
+                 * 50 percentile of latency, measured in milliseconds.
+                 */
+                p50?: number;
+                /**
+                 * 90 percentile of latency, measured in milliseconds.
+                 */
+                p90?: number;
+                /**
+                 * 95 percentile of latency, measured in milliseconds.
+                 */
+                p95?: number;
+                /**
+                 * 99 percentile of latency, measured in milliseconds.
+                 */
+                p99?: number;
+                /**
+                 * All the latency data points in the call, measured in milliseconds.
+                 */
+                values?: Array<number>;
+            }
+        }
+        /**
+         * LLM token usage of the call, available after call ends. Not populated if using
+         * custom LLM, realtime API, or no LLM call is made.
+         */
+        interface LlmTokenUsage {
+            /**
+             * Average token count of the call.
+             */
+            average: number;
+            /**
+             * Number of requests made to the LLM.
+             */
+            num_requests: number;
+            /**
+             * All the token count values in the call.
+             */
+            values: Array<number>;
+        }
+    }
+    interface V3PhoneCallResponse {
+        /**
+         * Corresponding agent id of this call.
+         */
+        agent_id: string;
+        /**
+         * The version of the agent.
+         */
+        agent_version: number;
+        /**
+         * Unique id of the call. Used to identify the call in the LLM websocket and used
+         * to authenticate in the audio websocket.
+         */
+        call_id: string;
+        /**
+         * Status of call.
+         *
+         * - `registered`: Call id issued, starting to make a call using this id.
+         * - `ongoing`: Call connected and ongoing.
+         * - `ended`: The underlying websocket has ended for the call. Either user or agent
+         *   hung up, or call transferred.
+         * - `error`: Call encountered error.
+         */
+        call_status: 'registered' | 'not_connected' | 'ongoing' | 'ended' | 'error';
+        /**
+         * Type of the call. Used to distinguish between web call and phone call.
+         */
+        call_type: 'phone_call';
+        /**
+         * Direction of the phone call.
+         */
+        direction: 'inbound' | 'outbound';
+        /**
+         * The caller number.
+         */
+        from_number: string;
+        /**
+         * The callee number.
+         */
+        to_number: string;
+        /**
+         * Name of the agent.
+         */
+        agent_name?: string;
+        /**
+         * Post call analysis that includes information such as sentiment, status, summary,
+         * and custom defined data to extract. Available after call ends. Subscribe to
+         * `call_analyzed` webhook event type to receive it once ready.
+         */
+        call_analysis?: V3PhoneCallResponse.CallAnalysis;
+        /**
+         * Cost of the call, including all the products and their costs and discount.
+         */
+        call_cost?: V3PhoneCallResponse.CallCost;
+        /**
+         * Dynamic variables collected from the call. Only available after the call ends.
+         */
+        collected_dynamic_variables?: {
+            [key: string]: string;
+        };
+        /**
+         * Custom SIP headers to be added to the call.
+         */
+        custom_sip_headers?: {
+            [key: string]: string;
+        };
+        /**
+         * Data storage setting for this call's agent. "everything" stores all data,
+         * "everything_except_pii" excludes PII when possible, "basic_attributes_only"
+         * stores only metadata.
+         */
+        data_storage_setting?: 'everything' | 'everything_except_pii' | 'basic_attributes_only' | null;
+        /**
+         * The reason for the disconnection of the call. Read detailed description about
+         * reasons listed here at
+         * [Disconnection Reason Doc](/reliability/debug-call-disconnect#understanding-disconnection-reasons).
+         */
+        disconnection_reason?: 'user_hangup' | 'agent_hangup' | 'call_transfer' | 'voicemail_reached' | 'ivr_reached' | 'inactivity' | 'max_duration_reached' | 'concurrency_limit_reached' | 'no_valid_payment' | 'scam_detected' | 'dial_busy' | 'dial_failed' | 'dial_no_answer' | 'invalid_destination' | 'telephony_provider_permission_denied' | 'telephony_provider_unavailable' | 'sip_routing_error' | 'marked_as_spam' | 'user_declined' | 'error_llm_websocket_open' | 'error_llm_websocket_lost_connection' | 'error_llm_websocket_runtime' | 'error_llm_websocket_corrupt_payload' | 'error_no_audio_received' | 'error_asr' | 'error_retell' | 'error_unknown' | 'error_user_not_joined' | 'registered_call_timeout' | 'transfer_bridged' | 'transfer_cancelled' | 'manual_stopped';
+        /**
+         * Duration of the call in milliseconds. Available after call ends.
+         */
+        duration_ms?: number;
+        /**
+         * End timestamp (milliseconds since epoch) of the call. Available after call ends.
+         */
+        end_timestamp?: number;
+        /**
+         * URL to the knowledge base retrieved contents of the call. Available after call
+         * ends if the call utilizes knowledge base feature. It consists of the respond id
+         * and the retrieved contents related to that response. It's already rendered in
+         * call history tab of dashboard, and you can also manually download and check
+         * against the transcript to view the knowledge base retrieval results.
+         */
+        knowledge_base_retrieved_contents_url?: string;
+        /**
+         * Latency tracking of the call, available after call ends. Not all fields here
+         * will be available, as it depends on the type of call and feature used.
+         */
+        latency?: V3PhoneCallResponse.Latency;
+        /**
+         * LLM token usage of the call, available after call ends. Not populated if using
+         * custom LLM, realtime API, or no LLM call is made.
+         */
+        llm_token_usage?: V3PhoneCallResponse.LlmTokenUsage;
+        /**
+         * An arbitrary object for storage purpose only. You can put anything here like
+         * your internal customer id associated with the call. Not used for processing. You
+         * can later get this field from the call object.
          */
-        arguments: string;
+        metadata?: unknown;
         /**
-         * Name of the function in this tool call.
+         * Whether this agent opts in for signed URLs for public logs and recordings. When
+         * enabled, the generated URLs will include security signatures that restrict
+         * access and automatically expire after 24 hours.
          */
-        name: string;
+        opt_in_signed_url?: boolean;
         /**
-         * This is a tool call invocation.
+         * Public log of the call, containing details about all the requests and responses
+         * received in LLM WebSocket, latency tracking for each turntaking, helpful for
+         * debugging and tracing. Available after call ends.
          */
-        role: 'tool_call_invocation';
+        public_log_url?: string;
         /**
-         * Tool call id, globally unique.
+         * Recording of the call, with each party's audio stored in a separate channel.
+         * Available after the call ends.
          */
-        tool_call_id: string;
+        recording_multi_channel_url?: string;
         /**
-         * Optional thought signature from Google Gemini thinking models. This is used
-         * internally to maintain reasoning chain in multi-turn function calling.
+         * Recording of the call. Available after call ends.
          */
-        thought_signature?: string;
-    }
-    interface ToolCallResultUtterance {
+        recording_url?: string;
         /**
-         * Result of the tool call, can be a string, a stringified json, etc.
+         * Add optional dynamic variables in key value pairs of string that injects into
+         * your Response Engine prompt and tool description. Only applicable for Response
+         * Engine.
          */
-        content: string;
+        retell_llm_dynamic_variables?: {
+            [key: string]: string;
+        };
         /**
-         * This is the result of a tool call.
+         * Recording of the call without PII, with each party's audio stored in a separate
+         * channel. Available after the call ends.
          */
-        role: 'tool_call_result';
+        scrubbed_recording_multi_channel_url?: string;
         /**
-         * Tool call id, globally unique.
+         * Recording of the call without PII. Available after call ends.
          */
-        tool_call_id: string;
+        scrubbed_recording_url?: string;
         /**
-         * Whether the tool call was successful.
+         * Begin timestamp (milliseconds since epoch) of the call. Available after call
+         * starts.
          */
-        successful?: boolean;
-    }
-    interface NodeTransitionUtterance {
+        start_timestamp?: number;
         /**
-         * Former node id
+         * Telephony identifier of the call, populated when available. Tracking purposes
+         * only.
          */
-        former_node_id: string;
+        telephony_identifier?: V3PhoneCallResponse.TelephonyIdentifier;
         /**
-         * Former node name
+         * The destination number or identifier where the call was transferred to. Only
+         * populated when the disconnection reason was `call_transfer`. Can be a phone
+         * number or a SIP URI. SIP URIs are prefixed with "sip:" and may include a
+         * ";transport=..." portion (if transport is known) where the transport type can be
+         * "tls", "tcp" or "udp".
          */
-        former_node_name: string;
+        transfer_destination?: string | null;
         /**
-         * New node id
+         * Transfer end timestamp (milliseconds since epoch) of the call. Available after
+         * transfer call ends.
          */
-        new_node_id: string;
+        transfer_end_timestamp?: number;
+    }
+    namespace V3PhoneCallResponse {
         /**
-         * New node name
+         * Post call analysis that includes information such as sentiment, status, summary,
+         * and custom defined data to extract. Available after call ends. Subscribe to
+         * `call_analyzed` webhook event type to receive it once ready.
          */
-        new_node_name: string;
+        interface CallAnalysis {
+            /**
+             * Whether the agent seems to have a successful call with the user, where the agent
+             * finishes the task, and the call was complete without being cutoff.
+             */
+            call_successful?: boolean;
+            /**
+             * A high level summary of the call.
+             */
+            call_summary?: string;
+            /**
+             * Custom analysis data that was extracted based on the schema defined in agent
+             * post call analysis data. Can be empty if nothing is specified.
+             */
+            custom_analysis_data?: unknown;
+            /**
+             * Whether the call is entered voicemail.
+             */
+            in_voicemail?: boolean;
+            /**
+             * Sentiment of the user in the call.
+             */
+            user_sentiment?: 'Negative' | 'Positive' | 'Neutral' | 'Unknown';
+        }
         /**
-         * This is result of a node transition
+         * Cost of the call, including all the products and their costs and discount.
          */
-        role: 'node_transition';
+        interface CallCost {
+            /**
+             * Combined cost of all individual costs in cents
+             */
+            combined_cost: number;
+            /**
+             * List of products with their unit prices and costs in cents
+             */
+            product_costs: Array<CallCost.ProductCost>;
+            /**
+             * Total duration of the call in seconds
+             */
+            total_duration_seconds: number;
+            /**
+             * Total unit duration price of all products in cents per second
+             */
+            total_duration_unit_price: number;
+        }
+        namespace CallCost {
+            interface ProductCost {
+                /**
+                 * Cost for the product in cents for the duration of the call.
+                 */
+                cost: number;
+                /**
+                 * Product name that has a cost associated with it.
+                 */
+                product: string;
+                /**
+                 * True if this cost item is for a transfer segment.
+                 */
+                is_transfer_leg_cost?: boolean;
+                /**
+                 * Unit price of the product in cents per second.
+                 */
+                unit_price?: number;
+            }
+        }
         /**
-         * How this node was reached. "global" means a global node transition,
-         * "global_go_back" means returning from a global node, "interrupt_go_back" means
-         * going back due to user interruption, and "normal" means a regular edge
-         * transition.
+         * Latency tracking of the call, available after call ends. Not all fields here
+         * will be available, as it depends on the type of call and feature used.
          */
-        transition_type?: 'global' | 'global_go_back' | 'interrupt_go_back' | 'normal';
-    }
-    interface DtmfUtterance {
+        interface Latency {
+            /**
+             * Transcription latency (diff between the duration of the chunks streamed and the
+             * durations of the transcribed part) tracking of the call.
+             */
+            asr?: Latency.Asr;
+            /**
+             * End to end latency (from user stops talking to agent start talking) tracking of
+             * the call. This latency does not account for the network trip time from Retell
+             * server to user frontend. The latency is tracked every time turn change between
+             * user and agent.
+             */
+            e2e?: Latency.E2E;
+            /**
+             * Knowledge base latency (from the triggering of knowledge base retrival to all
+             * relevant context received) tracking of the call. Only populated when using
+             * knowledge base feature for the agent of the call.
+             */
+            knowledge_base?: Latency.KnowledgeBase;
+            /**
+             * LLM latency (from issue of LLM call to first speakable chunk received) tracking
+             * of the call. When using custom LLM. this latency includes LLM websocket
+             * roundtrip time between user server and Retell server.
+             */
+            llm?: Latency.Llm;
+            /**
+             * LLM websocket roundtrip latency (between user server and Retell server) tracking
+             * of the call. Only populated for calls using custom LLM.
+             */
+            llm_websocket_network_rtt?: Latency.LlmWebsocketNetworkRtt;
+            /**
+             * Speech-to-speech latency (from requesting responses of a S2S model to first byte
+             * received) tracking of the call. Only populated for calls that uses S2S model
+             * like Realtime API.
+             */
+            s2s?: Latency.S2s;
+            /**
+             * Text-to-speech latency (from the triggering of TTS to first byte received)
+             * tracking of the call.
+             */
+            tts?: Latency.Tts;
+        }
+        namespace Latency {
+            /**
+             * Transcription latency (diff between the duration of the chunks streamed and the
+             * durations of the transcribed part) tracking of the call.
+             */
+            interface Asr {
+                /**
+                 * Maximum latency in the call, measured in milliseconds.
+                 */
+                max?: number;
+                /**
+                 * Minimum latency in the call, measured in milliseconds.
+                 */
+                min?: number;
+                /**
+                 * Number of data points (number of times latency is tracked).
+                 */
+                num?: number;
+                /**
+                 * 50 percentile of latency, measured in milliseconds.
+                 */
+                p50?: number;
+                /**
+                 * 90 percentile of latency, measured in milliseconds.
+                 */
+                p90?: number;
+                /**
+                 * 95 percentile of latency, measured in milliseconds.
+                 */
+                p95?: number;
+                /**
+                 * 99 percentile of latency, measured in milliseconds.
+                 */
+                p99?: number;
+                /**
+                 * All the latency data points in the call, measured in milliseconds.
+                 */
+                values?: Array<number>;
+            }
+            /**
+             * End to end latency (from user stops talking to agent start talking) tracking of
+             * the call. This latency does not account for the network trip time from Retell
+             * server to user frontend. The latency is tracked every time turn change between
+             * user and agent.
+             */
+            interface E2E {
+                /**
+                 * Maximum latency in the call, measured in milliseconds.
+                 */
+                max?: number;
+                /**
+                 * Minimum latency in the call, measured in milliseconds.
+                 */
+                min?: number;
+                /**
+                 * Number of data points (number of times latency is tracked).
+                 */
+                num?: number;
+                /**
+                 * 50 percentile of latency, measured in milliseconds.
+                 */
+                p50?: number;
+                /**
+                 * 90 percentile of latency, measured in milliseconds.
+                 */
+                p90?: number;
+                /**
+                 * 95 percentile of latency, measured in milliseconds.
+                 */
+                p95?: number;
+                /**
+                 * 99 percentile of latency, measured in milliseconds.
+                 */
+                p99?: number;
+                /**
+                 * All the latency data points in the call, measured in milliseconds.
+                 */
+                values?: Array<number>;
+            }
+            /**
+             * Knowledge base latency (from the triggering of knowledge base retrival to all
+             * relevant context received) tracking of the call. Only populated when using
+             * knowledge base feature for the agent of the call.
+             */
+            interface KnowledgeBase {
+                /**
+                 * Maximum latency in the call, measured in milliseconds.
+                 */
+                max?: number;
+                /**
+                 * Minimum latency in the call, measured in milliseconds.
+                 */
+                min?: number;
+                /**
+                 * Number of data points (number of times latency is tracked).
+                 */
+                num?: number;
+                /**
+                 * 50 percentile of latency, measured in milliseconds.
+                 */
+                p50?: number;
+                /**
+                 * 90 percentile of latency, measured in milliseconds.
+                 */
+                p90?: number;
+                /**
+                 * 95 percentile of latency, measured in milliseconds.
+                 */
+                p95?: number;
+                /**
+                 * 99 percentile of latency, measured in milliseconds.
+                 */
+                p99?: number;
+                /**
+                 * All the latency data points in the call, measured in milliseconds.
+                 */
+                values?: Array<number>;
+            }
+            /**
+             * LLM latency (from issue of LLM call to first speakable chunk received) tracking
+             * of the call. When using custom LLM. this latency includes LLM websocket
+             * roundtrip time between user server and Retell server.
+             */
+            interface Llm {
+                /**
+                 * Maximum latency in the call, measured in milliseconds.
+                 */
+                max?: number;
+                /**
+                 * Minimum latency in the call, measured in milliseconds.
+                 */
+                min?: number;
+                /**
+                 * Number of data points (number of times latency is tracked).
+                 */
+                num?: number;
+                /**
+                 * 50 percentile of latency, measured in milliseconds.
+                 */
+                p50?: number;
+                /**
+                 * 90 percentile of latency, measured in milliseconds.
+                 */
+                p90?: number;
+                /**
+                 * 95 percentile of latency, measured in milliseconds.
+                 */
+                p95?: number;
+                /**
+                 * 99 percentile of latency, measured in milliseconds.
+                 */
+                p99?: number;
+                /**
+                 * All the latency data points in the call, measured in milliseconds.
+                 */
+                values?: Array<number>;
+            }
+            /**
+             * LLM websocket roundtrip latency (between user server and Retell server) tracking
+             * of the call. Only populated for calls using custom LLM.
+             */
+            interface LlmWebsocketNetworkRtt {
+                /**
+                 * Maximum latency in the call, measured in milliseconds.
+                 */
+                max?: number;
+                /**
+                 * Minimum latency in the call, measured in milliseconds.
+                 */
+                min?: number;
+                /**
+                 * Number of data points (number of times latency is tracked).
+                 */
+                num?: number;
+                /**
+                 * 50 percentile of latency, measured in milliseconds.
+                 */
+                p50?: number;
+                /**
+                 * 90 percentile of latency, measured in milliseconds.
+                 */
+                p90?: number;
+                /**
+                 * 95 percentile of latency, measured in milliseconds.
+                 */
+                p95?: number;
+                /**
+                 * 99 percentile of latency, measured in milliseconds.
+                 */
+                p99?: number;
+                /**
+                 * All the latency data points in the call, measured in milliseconds.
+                 */
+                values?: Array<number>;
+            }
+            /**
+             * Speech-to-speech latency (from requesting responses of a S2S model to first byte
+             * received) tracking of the call. Only populated for calls that uses S2S model
+             * like Realtime API.
+             */
+            interface S2s {
+                /**
+                 * Maximum latency in the call, measured in milliseconds.
+                 */
+                max?: number;
+                /**
+                 * Minimum latency in the call, measured in milliseconds.
+                 */
+                min?: number;
+                /**
+                 * Number of data points (number of times latency is tracked).
+                 */
+                num?: number;
+                /**
+                 * 50 percentile of latency, measured in milliseconds.
+                 */
+                p50?: number;
+                /**
+                 * 90 percentile of latency, measured in milliseconds.
+                 */
+                p90?: number;
+                /**
+                 * 95 percentile of latency, measured in milliseconds.
+                 */
+                p95?: number;
+                /**
+                 * 99 percentile of latency, measured in milliseconds.
+                 */
+                p99?: number;
+                /**
+                 * All the latency data points in the call, measured in milliseconds.
+                 */
+                values?: Array<number>;
+            }
+            /**
+             * Text-to-speech latency (from the triggering of TTS to first byte received)
+             * tracking of the call.
+             */
+            interface Tts {
+                /**
+                 * Maximum latency in the call, measured in milliseconds.
+                 */
+                max?: number;
+                /**
+                 * Minimum latency in the call, measured in milliseconds.
+                 */
+                min?: number;
+                /**
+                 * Number of data points (number of times latency is tracked).
+                 */
+                num?: number;
+                /**
+                 * 50 percentile of latency, measured in milliseconds.
+                 */
+                p50?: number;
+                /**
+                 * 90 percentile of latency, measured in milliseconds.
+                 */
+                p90?: number;
+                /**
+                 * 95 percentile of latency, measured in milliseconds.
+                 */
+                p95?: number;
+                /**
+                 * 99 percentile of latency, measured in milliseconds.
+                 */
+                p99?: number;
+                /**
+                 * All the latency data points in the call, measured in milliseconds.
+                 */
+                values?: Array<number>;
+            }
+        }
         /**
-         * The digit pressed by the user. Will be a single digit string like "1", "2", "3",
-         * "\*", "#" etc.
+         * LLM token usage of the call, available after call ends. Not populated if using
+         * custom LLM, realtime API, or no LLM call is made.
          */
-        digit: string;
+        interface LlmTokenUsage {
+            /**
+             * Average token count of the call.
+             */
+            average: number;
+            /**
+             * Number of requests made to the LLM.
+             */
+            num_requests: number;
+            /**
+             * All the token count values in the call.
+             */
+            values: Array<number>;
+        }
         /**
-         * Digit pressed by the user from their phone keypad.
+         * Telephony identifier of the call, populated when available. Tracking purposes
+         * only.
          */
-        role: 'dtmf';
-    }
-}
-export interface CallListResponse {
-    /**
-     * Whether more results are available.
-     */
-    has_more?: boolean;
-    items?: Array<CallListResponse.CallWebCallResponse | CallListResponse.CallPhoneCallResponse>;
-    /**
-     * Pagination key for the next page.
-     */
-    pagination_key?: string;
-}
-export declare namespace CallListResponse {
-    /**
-     * V3 list calls response. Transcript fields are intentionally omitted.
-     */
-    interface CallWebCallResponse extends CallAPI.WebCallResponse {
-    }
-    /**
-     * V3 list calls response. Transcript fields are intentionally omitted.
-     */
-    interface CallPhoneCallResponse extends CallAPI.PhoneCallResponse {
+        interface TelephonyIdentifier {
+            /**
+             * Twilio call sid.
+             */
+            twilio_call_sid?: string;
+        }
     }
 }
 export interface CallUpdateParams {
@@ -2532,7 +3681,7 @@ export interface CallCreatePhoneCallParams {
      * Engine.
      */
     retell_llm_dynamic_variables?: {
-        [key: string]: unknown;
+        [key: string]: string;
     };
 }
 export declare namespace CallCreatePhoneCallParams {
@@ -2865,7 +4014,7 @@ export declare namespace CallCreatePhoneCallParams {
              * available voice models. Set to null to remove voice model selection, and default
              * ones will apply. Check out dashboard for more details of each voice model.
              */
-            voice_model?: 'eleven_turbo_v2' | 'eleven_flash_v2' | 'eleven_turbo_v2_5' | 'eleven_flash_v2_5' | 'eleven_multilingual_v2' | 'eleven_v3' | 'sonic-3' | 'sonic-3-latest' | 'tts-1' | 'gpt-4o-mini-tts' | 'speech-02-turbo' | 'speech-2.8-turbo' | 's1' | 's2-pro' | null;
+            voice_model?: 'eleven_turbo_v2' | 'eleven_flash_v2' | 'eleven_turbo_v2_5' | 'eleven_flash_v2_5' | 'eleven_multilingual_v2' | 'eleven_v3' | 'sonic-3' | 'sonic-3-latest' | 'sonic-3.5' | 'tts-1' | 'gpt-4o-mini-tts' | 'speech-02-turbo' | 'speech-2.8-turbo' | 's1' | 's2-pro' | 's2.1-pro' | null;
             /**
              * Controls speed of voice. Value ranging from [0.5,2]. Lower value means slower
              * speech, while higher value means faster speech rate. If unset, default value 1
@@ -3412,7 +4561,7 @@ export declare namespace CallCreatePhoneCallParams {
              * Select the underlying speech to speech model. Can only set this or model, not
              * both.
              */
-            s2s_model?: 'gpt-realtime-1.5' | 'gpt-realtime' | 'gpt-realtime-mini' | null;
+            s2s_model?: 'gpt-realtime-2' | 'gpt-realtime-1.5' | 'gpt-realtime' | 'gpt-realtime-mini' | null;
             /**
              * The speaker who starts the conversation. Required. Must be either 'user' or
              * 'agent'.
@@ -3481,7 +4630,7 @@ export interface CallCreateWebCallParams {
      * Engine.
      */
     retell_llm_dynamic_variables?: {
-        [key: string]: unknown;
+        [key: string]: string;
     };
 }
 export declare namespace CallCreateWebCallParams {
@@ -3814,7 +4963,7 @@ export declare namespace CallCreateWebCallParams {
              * available voice models. Set to null to remove voice model selection, and default
              * ones will apply. Check out dashboard for more details of each voice model.
              */
-            voice_model?: 'eleven_turbo_v2' | 'eleven_flash_v2' | 'eleven_turbo_v2_5' | 'eleven_flash_v2_5' | 'eleven_multilingual_v2' | 'eleven_v3' | 'sonic-3' | 'sonic-3-latest' | 'tts-1' | 'gpt-4o-mini-tts' | 'speech-02-turbo' | 'speech-2.8-turbo' | 's1' | 's2-pro' | null;
+            voice_model?: 'eleven_turbo_v2' | 'eleven_flash_v2' | 'eleven_turbo_v2_5' | 'eleven_flash_v2_5' | 'eleven_multilingual_v2' | 'eleven_v3' | 'sonic-3' | 'sonic-3-latest' | 'sonic-3.5' | 'tts-1' | 'gpt-4o-mini-tts' | 'speech-02-turbo' | 'speech-2.8-turbo' | 's1' | 's2-pro' | 's2.1-pro' | null;
             /**
              * Controls speed of voice. Value ranging from [0.5,2]. Lower value means slower
              * speech, while higher value means faster speech rate. If unset, default value 1
@@ -4361,7 +5510,7 @@ export declare namespace CallCreateWebCallParams {
              * Select the underlying speech to speech model. Can only set this or model, not
              * both.
              */
-            s2s_model?: 'gpt-realtime-1.5' | 'gpt-realtime' | 'gpt-realtime-mini' | null;
+            s2s_model?: 'gpt-realtime-2' | 'gpt-realtime-1.5' | 'gpt-realtime' | 'gpt-realtime-mini' | null;
             /**
              * The speaker who starts the conversation. Required. Must be either 'user' or
              * 'agent'.
@@ -4425,7 +5574,7 @@ export interface CallRegisterPhoneCallParams {
      * Engine.
      */
     retell_llm_dynamic_variables?: {
-        [key: string]: unknown;
+        [key: string]: string;
     };
     /**
      * The number you want to call, in E.164 format. Stored for tracking purpose.
@@ -4762,7 +5911,7 @@ export declare namespace CallRegisterPhoneCallParams {
              * available voice models. Set to null to remove voice model selection, and default
              * ones will apply. Check out dashboard for more details of each voice model.
              */
-            voice_model?: 'eleven_turbo_v2' | 'eleven_flash_v2' | 'eleven_turbo_v2_5' | 'eleven_flash_v2_5' | 'eleven_multilingual_v2' | 'eleven_v3' | 'sonic-3' | 'sonic-3-latest' | 'tts-1' | 'gpt-4o-mini-tts' | 'speech-02-turbo' | 'speech-2.8-turbo' | 's1' | 's2-pro' | null;
+            voice_model?: 'eleven_turbo_v2' | 'eleven_flash_v2' | 'eleven_turbo_v2_5' | 'eleven_flash_v2_5' | 'eleven_multilingual_v2' | 'eleven_v3' | 'sonic-3' | 'sonic-3-latest' | 'sonic-3.5' | 'tts-1' | 'gpt-4o-mini-tts' | 'speech-02-turbo' | 'speech-2.8-turbo' | 's1' | 's2-pro' | 's2.1-pro' | null;
             /**
              * Controls speed of voice. Value ranging from [0.5,2]. Lower value means slower
              * speech, while higher value means faster speech rate. If unset, default value 1
@@ -5309,7 +6458,7 @@ export declare namespace CallRegisterPhoneCallParams {
              * Select the underlying speech to speech model. Can only set this or model, not
              * both.
              */
-            s2s_model?: 'gpt-realtime-1.5' | 'gpt-realtime' | 'gpt-realtime-mini' | null;
+            s2s_model?: 'gpt-realtime-2' | 'gpt-realtime-1.5' | 'gpt-realtime' | 'gpt-realtime-mini' | null;
             /**
              * The speaker who starts the conversation. Required. Must be either 'user' or
              * 'agent'.