import { APIResource } from "../../core/resource.mjs";
import * as RealtimeAPI from "./realtime.mjs";
import * as ResponsesAPI from "../responses/responses.mjs";
import { APIPromise } from "../../core/api-promise.mjs";
import { RequestOptions } from "../../internal/request-options.mjs";
export declare class Calls extends APIResource {
    /**
     * Accept an incoming SIP call and configure the realtime session that will handle
     * it.
     *
     * @example
     * ```ts
     * await client.realtime.calls.accept('call_id', {
     *   type: 'realtime',
     * });
     * ```
     */
    accept(callID: string, body: CallAcceptParams, options?: RequestOptions): APIPromise<void>;
    /**
     * End an active Realtime API call, whether it was initiated over SIP or WebRTC.
     *
     * @example
     * ```ts
     * await client.realtime.calls.hangup('call_id');
     * ```
     */
    hangup(callID: string, options?: RequestOptions): APIPromise<void>;
    /**
     * Transfer an active SIP call to a new destination using the SIP REFER verb.
     *
     * @example
     * ```ts
     * await client.realtime.calls.refer('call_id', {
     *   target_uri: 'tel:+14155550123',
     * });
     * ```
     */
    refer(callID: string, body: CallReferParams, options?: RequestOptions): APIPromise<void>;
    /**
     * Decline an incoming SIP call by returning a SIP status code to the caller.
     *
     * @example
     * ```ts
     * await client.realtime.calls.reject('call_id');
     * ```
     */
    reject(callID: string, body?: CallRejectParams | null | undefined, options?: RequestOptions): APIPromise<void>;
}
export interface CallAcceptParams {
    /**
     * The type of session to create. Always `realtime` for the Realtime API.
     */
    type: 'realtime';
    /**
     * Configuration for input and output audio.
     */
    audio?: RealtimeAPI.RealtimeAudioConfig;
    /**
     * Additional fields to include in server outputs.
     *
     * `item.input_audio_transcription.logprobs`: Include logprobs for input audio
     * transcription.
     */
    include?: Array<'item.input_audio_transcription.logprobs'>;
    /**
     * The default system instructions (i.e. system message) prepended to model calls.
     * This field allows the client to guide the model on desired responses. The model
     * can be instructed on response content and format, (e.g. "be extremely succinct",
     * "act friendly", "here are examples of good responses") and on audio behavior
     * (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The
     * instructions are not guaranteed to be followed by the model, but they provide
     * guidance to the model on the desired behavior.
     *
     * Note that the server sets default instructions which will be used if this field
     * is not set and are visible in the `session.created` event at the start of the
     * session.
     */
    instructions?: string;
    /**
     * Maximum number of output tokens for a single assistant response, inclusive of
     * tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
     * `inf` for the maximum available tokens for a given model. Defaults to `inf`.
     */
    max_output_tokens?: number | 'inf';
    /**
     * The Realtime model used for this session.
     */
    model?: (string & {}) | 'gpt-realtime' | 'gpt-realtime-1.5' | 'gpt-realtime-2025-08-28' | 'gpt-4o-realtime-preview' | 'gpt-4o-realtime-preview-2024-10-01' | 'gpt-4o-realtime-preview-2024-12-17' | 'gpt-4o-realtime-preview-2025-06-03' | 'gpt-4o-mini-realtime-preview' | 'gpt-4o-mini-realtime-preview-2024-12-17' | 'gpt-realtime-mini' | 'gpt-realtime-mini-2025-10-06' | 'gpt-realtime-mini-2025-12-15' | 'gpt-audio-1.5' | 'gpt-audio-mini' | 'gpt-audio-mini-2025-10-06' | 'gpt-audio-mini-2025-12-15';
    /**
     * The set of modalities the model can respond with. It defaults to `["audio"]`,
     * indicating that the model will respond with audio plus a transcript. `["text"]`
     * can be used to make the model respond with text only. It is not possible to
     * request both `text` and `audio` at the same time.
     */
    output_modalities?: Array<'text' | 'audio'>;
    /**
     * Reference to a prompt template and its variables.
     * [Learn more](https://platform.openai.com/docs/guides/text?api-mode=responses#reusable-prompts).
     */
    prompt?: ResponsesAPI.ResponsePrompt | null;
    /**
     * How the model chooses tools. Provide one of the string modes or force a specific
     * function/MCP tool.
     */
    tool_choice?: RealtimeAPI.RealtimeToolChoiceConfig;
    /**
     * Tools available to the model.
     */
    tools?: RealtimeAPI.RealtimeToolsConfig;
    /**
     * Realtime API can write session traces to the
     * [Traces Dashboard](https://platform.openai.com/logs?api=traces). Set to null to
     * disable tracing. Once tracing is enabled for a session, the configuration cannot
     * be modified.
     *
     * `auto` will create a trace for the session with default values for the workflow
     * name, group id, and metadata.
     */
    tracing?: RealtimeAPI.RealtimeTracingConfig | null;
    /**
     * When the number of tokens in a conversation exceeds the model's input token
     * limit, the conversation be truncated, meaning messages (starting from the
     * oldest) will not be included in the model's context. A 32k context model with
     * 4,096 max output tokens can only include 28,224 tokens in the context before
     * truncation occurs.
     *
     * Clients can configure truncation behavior to truncate with a lower max token
     * limit, which is an effective way to control token usage and cost.
     *
     * Truncation will reduce the number of cached tokens on the next turn (busting the
     * cache), since messages are dropped from the beginning of the context. However,
     * clients can also configure truncation to retain messages up to a fraction of the
     * maximum context size, which will reduce the need for future truncations and thus
     * improve the cache rate.
     *
     * Truncation can be disabled entirely, which means the server will never truncate
     * but would instead return an error if the conversation exceeds the model's input
     * token limit.
     */
    truncation?: RealtimeAPI.RealtimeTruncation;
}
export interface CallReferParams {
    /**
     * URI that should appear in the SIP Refer-To header. Supports values like
     * `tel:+14155550123` or `sip:agent@example.com`.
     */
    target_uri: string;
}
export interface CallRejectParams {
    /**
     * SIP response code to send back to the caller. Defaults to `603` (Decline) when
     * omitted.
     */
    status_code?: number;
}
export declare namespace Calls {
    export { type CallAcceptParams as CallAcceptParams, type CallReferParams as CallReferParams, type CallRejectParams as CallRejectParams, };
}
//# sourceMappingURL=calls.d.mts.map