@fonoster/common 0.27.0 → 0.30.0

package/dist/protos/voice.proto

@@ -1,557 +0,0 @@
-/*
- * Copyright (C) 2025 by Fonoster Inc (https://fonoster.com)
- * http://github.com/fonoster/fonoster
- *
- * This file is part of Fonoster
- *
- * Licensed under the MIT License (the "License");
- * you may not use this file except in compliance with
- * the License. You may obtain a copy of the License at
- *
- *    https://opensource.org/licenses/MIT
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-syntax = "proto3";
-
-package fonoster.voice.v1beta2;
-
-import "google/protobuf/struct.proto";
-
-// The voice service definition
-service Voice {
-  rpc CreateSession(stream VoiceInStream) returns (stream VoiceOutStream) {}
-}
-
-// CreateSessionRequest is the request message for the CreateSession method
-message CreateSessionRequest {
-  // The account making the request on behalf of the caller
-  string access_key_id = 1;
-
-  // Reference to an application living in the platform
-  string app_ref = 2;
-
-  // gRPC endpoint for the voice service
-  string endpoint = 3;
-
-  // PSTN number or SIP URI associated with the session
-  string ingress_number = 4;
-
-  // The name of the caller
-  string caller_name = 5;
-
-  // The caller number for the phone calling from the
-  string caller_number = 6;
-
-  // The session ID generated by the Media Server
-  string media_session_ref = 7;
-
-  // The call reference identifier
-  string call_ref = 11;
-
-  // Use this token to validate the provenance of the request
-  // If the app_ref is provided, the token must include the app_ref as a claim
-  string session_token = 8;
-
-  // The direction of the call
-  enum CallDirection {
-    FROM_PSTN = 0;
-    TO_PSTN = 1;
-    INTRA_NETWORK = 2;
-  }
-  CallDirection call_direction = 9;
-
-  // Metadata associated with the session
-  map<string, string> metadata = 10;
-
-  // Answering Machine Detection result, computed on early call audio before
-  // this session is dispatched. Absent/UNSPECIFIED when the probe is disabled.
-  Amd amd = 12;
-}
-
-// Answering Machine Detection verdict for the answered call
-message Amd {
-  enum Status {
-    // The probe was disabled or did not run
-    AMD_STATUS_UNSPECIFIED = 0;
-    // A live person answered
-    HUMAN = 1;
-    // A generic answering machine answered
-    MACHINE = 2;
-    // A personal or carrier voicemail greeting answered
-    VOICEMAIL = 3;
-    // An auto-attendant / IVR menu answered
-    IVR = 4;
-    // The probe ran but was inconclusive (low confidence, silence, or timeout)
-    UNKNOWN = 5;
-  }
-
-  // The detected class of answerer
-  Status status = 1;
-
-  // Model confidence for the reported status, 0..1
-  float confidence = 2;
-
-  // Identifier of the detector that produced the verdict (e.g. "asterisk-amd@1")
-  string detector = 3;
-
-  // Wall-clock time spent producing the verdict, in milliseconds
-  int32 latency_ms = 4;
-}
-
-// Common request for verbs
-message VerbRequest {
-  string media_session_ref = 1;
-}
-
-// Common response for verbs
-message VerbResponse {
-  string media_session_ref = 1;
-}
-
-// Request to play a sound
-message PlayRequest {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The URL of the sound to play
-  string url = 2;
-}
-
-// Request to play a sound
-message PlayResponse {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The playback reference generated by the Media Server
-  string playback_ref = 2;
-}
-
-// The request message for the PlaybackControl method
-message PlaybackControlRequest {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The playback reference generated by the Media Server
-  string playback_ref = 2;
-
-  // The action to take
-  enum Action {
-    STOP = 0;
-    RESTART = 1;
-    PAUSE = 2;
-    UNPAUSE = 3;
-    FORWARD = 4;
-  }
-  Action action = 3;
-}
-
-// PlayDtmfRequest is the request message for the PlayDtmf method
-message PlayDtmfRequest {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The digits to play
-  string digits = 2;
-}
-
-// Request to mute or unmute the call
-message MuteRequest {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The direction to mute
-  enum MuteDirection {
-    IN = 0;
-    OUT = 1;
-    BOTH = 2;
-  }
-  MuteDirection direction = 2;
-}
-
-// Request to gather speech or DTMF
-message GatherRequest {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The source of the gather
-  enum GatherSource {
-    SPEECH = 0;
-    DTMF = 1;
-    SPEECH_AND_DTMF = 2;
-  }
-  GatherSource source = 2;
-
-  // Optional key to finish the gather
-  string finish_on_key = 3;
-
-  // Optional timeout in seconds
-  int32 timeout = 4;
-
-  // Optional number of digits to gather
-  int32 max_digits = 5;
-}
-
-// Response to a gather request
-message GatherResponse {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The gathered speech or digits
-  oneof content {
-    string speech = 2;
-    string digits = 3;
-  }
-  int32 response_time = 4;
-}
-
-// RecordRequest is the request message to initiate a recording
-message RecordRequest {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The maximum duration of the recording in seconds
-  int32 max_duration = 2;
-
-  // The maximum silence before ending the recording in seconds
-  int32 max_silence = 3;
-
-  // Play a beep before recording
-  bool beep = 4;
-
-  // Finish recording on this key
-  string finish_on_key = 5;
-}
-
-// Response to a record request
-message RecordResponse {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The name of the recording
-  string name = 2;
-
-  // The duration of the recording
-  int32 duration = 3;
-
-  // The format of the recording
-  enum RecordFormat {
-    WAV = 0;
-  }
-  RecordFormat format = 4;
-}
-
-// Request to say a text
-message SayRequest {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The text to say
-  string text = 2;
-
-  // Assign a playback reference to the request
-  string playback_ref = 3 [deprecated = true];
-
-  // TTS options
-  google.protobuf.Struct options = 4;
-}
-
-message StopSayRequest {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-}
-
-// Dial request message
-message DialRequest {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The number to dial
-  string destination = 2;
-
-  // The caller ID to use (reserved for future use)
-  string caller_id = 3;
-
-  // The timeout in seconds
-  int32 timeout = 4;
-
-  // Record direction
-  enum RecordDirection {
-    IN = 0;
-    OUT = 1;
-    BOTH = 2;
-  }
-  RecordDirection record_direction = 5;
-}
-
-// Dial status message
-message DialStatus {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The status of the dial
-  enum Status {
-    TRYING = 0;
-    CANCEL = 1;
-    ANSWER = 2;
-    BUSY = 3;
-    PROGRESS = 4;
-    NOANSWER = 5;
-    // Maps from Asterisk's CHANUNAVAIL and CONGESTION
-    FAILED = 6;
-  }
-  Status status = 2;
-}
-
-// Request to start an audio stream
-message StartStreamRequest {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The direction of the stream
-  enum StreamDirection {
-    IN = 0;
-    OUT = 1;
-    BOTH = 2;
-  }
-  StreamDirection direction = 2;
-
-  // Output format
-  enum StreamAudioFormat {
-    WAV = 0;
-  }
-  StreamAudioFormat format = 3;
-}
-
-// Response to a start stream request
-message StartStreamResponse {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The stream reference generated by the Media Server
-  string stream_ref = 2;
-}
-
-// Request to stop an audio stream
-message StopStreamRequest {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The stream reference generated by the Media Server
-  string stream_ref = 2;
-}
-
-// Message for the bidirectional streams
-message StreamPayload {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The stream reference generated by the Media Server 
-  string stream_ref = 2;
-
-  // The message type
-  enum StreamMessageType {
-    AUDIO_IN = 0;
-    AUDIO_OUT = 1;
-    ERROR = 2;
-  }
-  StreamMessageType type = 3;
-
-  // The audio data
-  bytes data = 4;
-
-  // (Q. should we use the same error codes as gRPC?)
-  // Error code if any
-  string code = 5;
-
-  // Error message if any
-  string message = 6;
-}
-
-// Request for Stream Gather
-message StartStreamGatherRequest {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The source of the gather
-  enum StreamGatherSource {
-    SPEECH = 0;
-    DTMF = 1;
-    SPEECH_AND_DTMF = 2;
-  }
-  StreamGatherSource source = 2;
-}
-
-// Request to stop a Stream Gather
-message StopStreamGatherRequest {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-}
-
-// Response to Stream Gather request
-message StreamGatherPayload {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // The gathered speech or a single digit
-  oneof content {
-    string speech = 2;
-    string digit = 3;
-  }
-  int32 response_time = 4;
-}
-
-// An audio filter the application wants applied to the caller's audio
-message AudioFilter {
-  // Name of a filter registered in the Media Server (e.g. "aiCoustics")
-  string name = 1;
-
-  // Filter-specific options. Carries tuning values only; credentials are read
-  // from the Media Server's environment and never travel on this stream.
-  google.protobuf.Struct options = 2;
-}
-
-// Request to set the audio filters for the session. Sending it again replaces
-// the previous filters; an empty list removes them.
-message SetAudioFiltersRequest {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // Filters to run, in order
-  repeated AudioFilter filters = 2;
-}
-
-// Result of setting the session's audio filters. `error` is set when the
-// filters could not be applied; the call continues with unfiltered audio.
-message SetAudioFiltersResponse {
-  // The session reference generated by the Media Server
-  string media_session_ref = 1;
-
-  // Empty when the filters were applied
-  string error = 2;
-}
-
-// VoiceInStream is the input stream for the voice service
-message VoiceInStream {
-  oneof content {
-    // Initial request to create a new session
-    CreateSessionRequest request = 1;
-
-    // Command to accept the call
-    VerbResponse answer_response = 2;
-
-    // Response to a play request
-    PlayResponse play_response = 3;
-
-    // Response to a hangup request
-    VerbResponse hangup_response = 4;
-
-    // Response to a mute request
-    VerbResponse mute_response = 5;
-
-    // Response to a unmute request
-    VerbResponse unmute_response = 6;
-
-    // Response to a play dtmf request
-    VerbResponse play_dtmf_response = 7;
-
-    // Response to a gather request
-    GatherResponse gather_response = 8;
-
-    // Response to a playback control request
-    VerbResponse playback_control_response = 9;
-
-    // Response to a say request (It shares the same response as play)
-    PlayResponse say_response = 10;
-
-    // Response to stopping the an active say speech
-    VerbResponse stop_say_response = 18;
-
-    // Response to a record request
-    RecordResponse record_response = 11;
-
-    // Status of a dial request progress
-    DialStatus dial_status = 12;
-
-    // Response to a start stream request
-    StartStreamResponse start_stream_response = 13;
-
-    // Message for the bidirectional streams
-    StreamPayload stream_payload = 14;
-
-    // Response to start a stream gather
-    VerbResponse start_stream_gather_response = 15;
-
-    // Response to stop a stream gather
-    VerbResponse stop_stream_gather_response = 16;
-
-    // Message with payload for the stream gather
-    StreamGatherPayload stream_gather_payload = 17;
-
-    // Response to setting the session's audio filters
-    SetAudioFiltersResponse set_audio_filters_response = 19;
-  }
-}
-
-// VoiceOutStream is the output stream for the voice service
-message VoiceOutStream {
-  oneof content {
-    // Request to accept the call
-    VerbRequest answer_request = 1;
-
-    // Request to play a sound
-    PlayRequest play_request = 2;
-
-    // Request to hangup the call
-    VerbRequest hangup_request = 3;
-
-    // Request to mute the call
-    MuteRequest mute_request = 4;
-
-    // Request to unmute the call
-    MuteRequest unmute_request = 5;
-
-    // Request to play dtmf
-    PlayDtmfRequest play_dtmf_request = 6;
-
-    // Request to gather speech or DTMF
-    GatherRequest gather_request = 7;
-
-    // Request to control a playback
-    PlaybackControlRequest playback_control_request = 9;
-
-    // Request to say a text
-    SayRequest say_request = 10;
-
-    // Request to stop any active Say speech
-    StopSayRequest stop_say_request = 19;
-
-    // Request to record
-    RecordRequest record_request = 11;
-
-    // Request to dial a number
-    DialRequest dial_request = 12;
-
-    // Request to start an audio stream
-    StartStreamRequest start_stream_request = 14;
-
-    // Request to stop an audio stream
-    StopStreamRequest stop_stream_request = 15;
-
-    // Message for the bidirectional streams
-    StreamPayload stream_payload = 16;
-
-    // Request to start a stream gather
-    StartStreamGatherRequest start_stream_gather_request = 17;
-
-    // Request to stop a stream gather
-    StopStreamGatherRequest stop_stream_gather_request = 18;
-
-    // Request to set the audio filters for the session
-    SetAudioFiltersRequest set_audio_filters_request = 20;
-  }
-}

package/dist/voice/Dial.js

@@ -1,20 +0,0 @@
-"use strict";
-Object.defineProperty(exports, "__esModule", { value: true });
-exports.DialStatus = exports.DialRecordDirection = void 0;
-var DialRecordDirection;
-(function (DialRecordDirection) {
-    DialRecordDirection["IN"] = "IN";
-    DialRecordDirection["OUT"] = "OUT";
-    DialRecordDirection["BOTH"] = "BOTH";
-})(DialRecordDirection || (exports.DialRecordDirection = DialRecordDirection = {}));
-var DialStatus;
-(function (DialStatus) {
-    DialStatus["TRYING"] = "TRYING";
-    DialStatus["CANCEL"] = "CANCEL";
-    DialStatus["ANSWER"] = "ANSWER";
-    DialStatus["BUSY"] = "BUSY";
-    DialStatus["PROGRESS"] = "PROGRESS";
-    DialStatus["NOANSWER"] = "NOANSWER";
-    // Maps from Asterisk's CHANUNAVAIL and CONGESTION
-    DialStatus["FAILED"] = "FAILED";
-})(DialStatus || (exports.DialStatus = DialStatus = {}));

package/dist/voice/Gather.d.ts

@@ -1,37 +0,0 @@
-/**
- * Copyright (C) 2025 by Fonoster Inc (https://fonoster.com)
- * http://github.com/fonoster/fonoster
- *
- * This file is part of Fonoster
- *
- * Licensed under the MIT License (the "License");
- * you may not use this file except in compliance with
- * the License. You may obtain a copy of the License at
- *
- *    https://opensource.org/licenses/MIT
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-import { VerbRequest, VerbResponse } from "./Verb";
-declare enum GatherSource {
-    SPEECH = "speech",
-    DTMF = "dtmf",
-    SPEECH_AND_DTMF = "speech,dtmf"
-}
-type GatherOptions = {
-    finishOnKey?: string;
-    maxDigits?: number;
-    timeout?: number;
-    source?: GatherSource;
-};
-type GatherRequest = VerbRequest & GatherOptions;
-type GatherResponse = VerbResponse & {
-    speech?: string;
-    digits?: string;
-    responseTime: number;
-};
-export { GatherOptions, GatherRequest, GatherResponse, GatherSource };

package/dist/voice/Gather.js

@@ -1,9 +0,0 @@
-"use strict";
-Object.defineProperty(exports, "__esModule", { value: true });
-exports.GatherSource = void 0;
-var GatherSource;
-(function (GatherSource) {
-    GatherSource["SPEECH"] = "speech";
-    GatherSource["DTMF"] = "dtmf";
-    GatherSource["SPEECH_AND_DTMF"] = "speech,dtmf";
-})(GatherSource || (exports.GatherSource = GatherSource = {}));

package/dist/voice/Mute.d.ts

@@ -1,31 +0,0 @@
-/**
- * Copyright (C) 2025 by Fonoster Inc (https://fonoster.com)
- * http://github.com/fonoster/fonoster
- *
- * This file is part of Fonoster
- *
- * Licensed under the MIT License (the "License");
- * you may not use this file except in compliance with
- * the License. You may obtain a copy of the License at
- *
- *    https://opensource.org/licenses/MIT
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-import { VerbRequest } from "./Verb";
-type MuteOptions = {
-    direction?: MuteDirection;
-};
-declare enum MuteDirection {
-    IN = "IN",
-    OUT = "OUT",
-    BOTH = "BOTH"
-}
-type MuteRequest = VerbRequest & {
-    direction: MuteDirection;
-};
-export { MuteDirection, MuteOptions, MuteRequest };

package/dist/voice/Mute.js

@@ -1,9 +0,0 @@
-"use strict";
-Object.defineProperty(exports, "__esModule", { value: true });
-exports.MuteDirection = void 0;
-var MuteDirection;
-(function (MuteDirection) {
-    MuteDirection["IN"] = "IN";
-    MuteDirection["OUT"] = "OUT";
-    MuteDirection["BOTH"] = "BOTH";
-})(MuteDirection || (exports.MuteDirection = MuteDirection = {}));

package/dist/voice/Play.js

@@ -1,2 +0,0 @@
-"use strict";
-Object.defineProperty(exports, "__esModule", { value: true });

package/dist/voice/PlayDtmf.js

@@ -1,2 +0,0 @@
-"use strict";
-Object.defineProperty(exports, "__esModule", { value: true });

package/dist/voice/PlaybackControl.d.ts

@@ -1,31 +0,0 @@
-/**
- * Copyright (C) 2025 by Fonoster Inc (https://fonoster.com)
- * http://github.com/fonoster/fonoster
- *
- * This file is part of Fonoster
- *
- * Licensed under the MIT License (the "License");
- * you may not use this file except in compliance with
- * the License. You may obtain a copy of the License at
- *
- *    https://opensource.org/licenses/MIT
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-import { VerbRequest } from "./Verb";
-declare enum PlaybackControlAction {
-    STOP = "STOP",
-    RESTART = "RESTART",
-    PAUSE = "PAUSE",
-    UNPAUSE = "UNPAUSE",
-    FORWARD = "FORWARD"
-}
-type PlaybackControlRequest = VerbRequest & {
-    playbackRef: string;
-    action: PlaybackControlAction;
-};
-export { PlaybackControlAction, PlaybackControlRequest };