@nethesis/phone-island 0.7.78 → 0.7.80

package/dist/index.css

@@ -1 +1 @@
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package/dist/lib/devices/devices.js

@@ -1,2 +1,2 @@
-"use strict";Object.defineProperty(exports,"__esModule",{value:!0}),require("../../node_modules/mic-check/lib/index.js");var e=require("../webrtc/janus.js"),r=require("../../store/index.js"),i=require("../../_virtual/index8.js"),o=e.default;exports.checkMediaPermissions=function(){i.__exports.requestMediaPermissions({audio:!0,video:!1}).then((function(){})).catch((function(e){var s=e.type;e.name,e.message,s===i.__exports.MediaPermissionsErrorType.SystemPermissionDenied?(r.store.dispatch.alerts.setAlert("browser_permissions"),o.error&&o.error("WebRTC: browser does not have permission to access camera or microphone")):s===i.__exports.MediaPermissionsErrorType.UserPermissionDenied?(r.store.dispatch.alerts.setAlert("user_permissions"),o.error&&o.error("WebRTC: user didn't allow app to access camera or microphone")):s===i.__exports.MediaPermissionsErrorType.CouldNotStartVideoSource?(r.store.dispatch.alerts.setAlert("busy_camera"),o.error&&o.error("WebRTC: camera is in use by another application (Zoom, Skype) or browser tab (Google Meet, Messenger Video)")):(r.store.dispatch.alerts.setAlert("unknown_media_permissions"),o.error&&o.error("WebRTC: can't access audio or camere on this device. unknown error"))}))},exports.getSupportedDevices=function(e){var r,i,s=!1,t=!1,n=!1,a=!1;r=function(){var r={audio:s,audioCap:n,video:t,videoCap:a};o.log&&o.log("supportedDevices=",r),e()},i=[],navigator.mediaDevices.enumerateDevices().then((function(e){e.forEach((function(e){var r=!1;i.forEach((function(i){i.deviceId===e.deviceId&&i.kind===e.kind&&(r=!0)})),r||("videoinput"!==e.kind||a||(a=!0),"audioinput"!==e.kind||n||(n=!0),"audioinput"===e.kind&&(s=!0),e.kind,"videoinput"===e.kind&&(t=!0),i.push(e))})),r&&r()}))};
+"use strict";Object.defineProperty(exports,"__esModule",{value:!0}),require("../../node_modules/mic-check/lib/index.js");var e=require("../webrtc/janus.js"),r=require("../../store/index.js"),i=require("../../_virtual/index9.js"),o=e.default;exports.checkMediaPermissions=function(){i.__exports.requestMediaPermissions({audio:!0,video:!1}).then((function(){})).catch((function(e){var s=e.type;e.name,e.message,s===i.__exports.MediaPermissionsErrorType.SystemPermissionDenied?(r.store.dispatch.alerts.setAlert("browser_permissions"),o.error&&o.error("WebRTC: browser does not have permission to access camera or microphone")):s===i.__exports.MediaPermissionsErrorType.UserPermissionDenied?(r.store.dispatch.alerts.setAlert("user_permissions"),o.error&&o.error("WebRTC: user didn't allow app to access camera or microphone")):s===i.__exports.MediaPermissionsErrorType.CouldNotStartVideoSource?(r.store.dispatch.alerts.setAlert("busy_camera"),o.error&&o.error("WebRTC: camera is in use by another application (Zoom, Skype) or browser tab (Google Meet, Messenger Video)")):(r.store.dispatch.alerts.setAlert("unknown_media_permissions"),o.error&&o.error("WebRTC: can't access audio or camere on this device. unknown error"))}))},exports.getSupportedDevices=function(e){var r,i,s=!1,t=!1,n=!1,a=!1;r=function(){var r={audio:s,audioCap:n,video:t,videoCap:a};o.log&&o.log("supportedDevices=",r),e()},i=[],navigator.mediaDevices.enumerateDevices().then((function(e){e.forEach((function(e){var r=!1;i.forEach((function(i){i.deviceId===e.deviceId&&i.kind===e.kind&&(r=!0)})),r||("videoinput"!==e.kind||a||(a=!0),"audioinput"!==e.kind||n||(n=!0),"audioinput"===e.kind&&(s=!0),e.kind,"videoinput"===e.kind&&(t=!0),i.push(e))})),r&&r()}))};
 //# sourceMappingURL=devices.js.map

package/dist/lib/phone/call.d.ts

@@ -2,7 +2,7 @@
  * Starts a call to a number
  * @param number The number string
  */
-export declare function callNumber(number: string): void;
+export declare function callNumber(number: string, sipHost: string): void;
 /**
  * Starts a call to a SIP URI
  *
@@ -34,7 +34,7 @@ export declare function pauseCurrentCall(): void;
  */
 export declare function unpauseCurrentCall(): void;
 /**
- * Transfer the current call through a blind transfer
+ * Transfer the current call through a blind transfer (not in use)
  */
 export declare function blindTransfer(number: string): Promise<boolean | undefined>;
 /**

package/dist/lib/phone/call.js

@@ -1,2 +1,2 @@
-"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("../../node_modules/tslib/tslib.es6.js"),t=require("../devices/devices.js"),r=require("../webrtc/janus.js"),n=require("../webrtc/messages.js"),a=require("../../store/index.js"),o=require("../user/default_device.js"),s=require("../../services/astproxy.js"),i=require("../../static/dtmf/index.js");function u(a){var o=this;t.getSupportedDevices((function(){return e.__awaiter(o,void 0,void 0,(function(){return e.__generator(this,(function(e){switch(e.label){case 0:return r.default.log("This is a SIP call"),[4,n.call(a,{audio:{mandatory:{echoCancellation:!1,googEchoCancellation:!1,googAutoGainControl:!1,googAutoGainControl2:!1,googNoiseSuppression:!1,googHighpassFilter:!1,googTypingNoiseDetection:!1}},audioSend:!0,audioRecv:!0,videoSend:!1,videoRecv:!1})];case 1:return e.sent(),[2]}}))}))}))}exports.answerIncomingCall=function(){o.isWebRTC()&&n.answerWebRTC()},exports.attendedTransfer=function(t){return e.__awaiter(this,void 0,void 0,(function(){var r,n;return e.__generator(this,(function(e){switch(e.label){case 0:return r=a.store.getState().currentCall.conversationId,n=a.store.getState().currentUser.default_device,r&&(null==n?void 0:n.id)&&t?[4,s.attendedTransfer({convid:r,to:t,endpointId:n.id})]:[3,2];case 1:return[2,e.sent()];case 2:return[2]}}))}))},exports.blindTransfer=function(t){return e.__awaiter(this,void 0,void 0,(function(){var r,n;return e.__generator(this,(function(e){switch(e.label){case 0:return r=a.store.getState().currentCall.conversationId,n=a.store.getState().currentUser.default_device,r&&(null==n?void 0:n.id)&&t?[4,s.blindTransfer({convid:r,to:t,endpointId:n.id})]:[3,2];case 1:return[2,e.sent()];case 2:return[2]}}))}))},exports.callNumber=function(e){u("sip:".concat(e,"@").concat("127.0.0.1"))},exports.callSipURI=u,exports.hangupCurrentCall=function(){var e=a.store.getState().currentCall,t=e.outgoing,r=e.accepted;t||r?n.hangup():n.decline(),a.store.dispatch.player.stopAudioPlayer(),a.store.dispatch.currentCall.reset()},exports.muteCurrentCall=function(){o.isWebRTC()&&n.muteWebRTC()&&a.store.dispatch.currentCall.updateCurrentCall({muted:!0})},exports.pauseCurrentCall=function(){o.isWebRTC()&&n.pauseWebRTC()&&(a.store.dispatch.currentCall.updateCurrentCall({paused:!0}),a.store.dispatch.player.pauseRemoteAudio())},exports.playDtmfAudio=function(e){"*"===e&&(e="star"),"#"===e&&(e="pound"),a.store.dispatch.player.updateStartAudioPlayer({src:i.default["dtmf_".concat(e)]})},exports.unmuteCurrentCall=function(){o.isWebRTC()&&n.unmuteWebRTC()&&a.store.dispatch.currentCall.updateCurrentCall({muted:!1})},exports.unpauseCurrentCall=function(){o.isWebRTC()&&n.unpauseWebRTC()&&(a.store.dispatch.currentCall.updateCurrentCall({paused:!1}),a.store.dispatch.player.playRemoteAudio())};
+"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("../../node_modules/tslib/tslib.es6.js"),t=require("../devices/devices.js"),r=require("../webrtc/janus.js"),a=require("../webrtc/messages.js"),n=require("../../store/index.js"),o=require("../user/default_device.js"),s=require("../../services/astproxy.js"),u=require("../../static/dtmf/index.js");function i(n){var o=this;t.getSupportedDevices((function(){return e.__awaiter(o,void 0,void 0,(function(){return e.__generator(this,(function(e){switch(e.label){case 0:return r.default.log("This is a SIP call"),[4,a.call(n,{audio:{mandatory:{echoCancellation:!1,googEchoCancellation:!1,googAutoGainControl:!1,googAutoGainControl2:!1,googNoiseSuppression:!1,googHighpassFilter:!1,googTypingNoiseDetection:!1}},audioSend:!0,audioRecv:!0,videoSend:!1,videoRecv:!1})];case 1:return e.sent(),[2]}}))}))}))}exports.answerIncomingCall=function(){o.isWebRTC()&&a.answerWebRTC()},exports.attendedTransfer=function(t){return e.__awaiter(this,void 0,void 0,(function(){var r,a;return e.__generator(this,(function(e){switch(e.label){case 0:return r=n.store.getState().currentCall.conversationId,a=n.store.getState().currentUser.default_device,r&&(null==a?void 0:a.id)&&t?[4,s.attendedTransfer({convid:r,to:t,endpointId:a.id})]:[3,2];case 1:return[2,e.sent()];case 2:return[2]}}))}))},exports.callNumber=function(e,t){i("sip:".concat(e,"@").concat(t))},exports.callSipURI=i,exports.hangupCurrentCall=function(){var e=n.store.getState().currentCall,t=e.outgoing,r=e.accepted;t||r?a.hangup():a.decline(),n.store.dispatch.player.stopAudioPlayer(),n.store.dispatch.currentCall.reset()},exports.muteCurrentCall=function(){o.isWebRTC()&&a.muteWebRTC()&&n.store.dispatch.currentCall.updateCurrentCall({muted:!0})},exports.pauseCurrentCall=function(){o.isWebRTC()&&a.pauseWebRTC()&&(n.store.dispatch.currentCall.updateCurrentCall({paused:!0}),n.store.dispatch.player.pauseRemoteAudio())},exports.playDtmfAudio=function(e){"*"===e&&(e="star"),"#"===e&&(e="pound"),n.store.dispatch.player.updateStartAudioPlayer({src:u.default["dtmf_".concat(e)]})},exports.unmuteCurrentCall=function(){o.isWebRTC()&&a.unmuteWebRTC()&&n.store.dispatch.currentCall.updateCurrentCall({muted:!1})},exports.unpauseCurrentCall=function(){o.isWebRTC()&&a.unpauseWebRTC()&&(n.store.dispatch.currentCall.updateCurrentCall({paused:!1}),n.store.dispatch.player.playRemoteAudio())};
 //# sourceMappingURL=call.js.map

package/dist/lib/phone/call.js.map

@@ -1 +1 @@
-{"version":3,"file":"call.js","sources":["../../../src/lib/phone/call.ts"],"sourcesContent":["// Copyright (C) 2022 Nethesis S.r.l.\n// SPDX-License-Identifier: AGPL-3.0-or-later\n\nimport { getSupportedDevices } from '../devices/devices'\nimport Janus from '../webrtc/janus'\nimport {\n  call,\n  hangup,\n  decline,\n  answerWebRTC,\n  muteWebRTC,\n  unmuteWebRTC,\n  pauseWebRTC,\n  unpauseWebRTC,\n} from '../webrtc/messages'\nimport { store } from '../../store'\nimport { isWebRTC } from '../user/default_device'\nimport {\n  blindTransfer as blindTransferRequest,\n  attendedTransfer as attendedTransferRequest,\n} from '../../services/astproxy'\nimport dtmfAudios from '../../static/dtmf'\n\n/**\n * Starts a call to a number\n * @param number The number string\n */\nexport function callNumber(number: string) {\n  const BASE_HOST_URL: string = '127.0.0.1'\n  const sipURI = `sip:${number}@${BASE_HOST_URL}`\n  callSipURI(sipURI)\n}\n\n/**\n * Starts a call to a SIP URI\n *\n * @param sipURI The SIP URI string\n */\nexport function callSipURI(sipURI: string) {\n  getSupportedDevices(async () => {\n    // @ts-ignore\n    Janus.log('This is a SIP call')\n    await call(sipURI, {\n      audio: {\n        mandatory: {\n          echoCancellation: false,\n          googEchoCancellation: false,\n          googAutoGainControl: false,\n          googAutoGainControl2: false,\n          googNoiseSuppression: false,\n          googHighpassFilter: false,\n          googTypingNoiseDetection: false,\n        },\n      },\n      audioSend: true,\n      audioRecv: true,\n      videoSend: false,\n      videoRecv: false,\n    })\n  })\n}\n\n/**\n * Answer incoming call\n */\nexport function answerIncomingCall() {\n  if (isWebRTC()) {\n    answerWebRTC()\n  }\n}\n\n/**\n * Hangup current call\n */\nexport function hangupCurrentCall() {\n  const { outgoing, accepted } = store.getState().currentCall\n  if (outgoing || accepted) {\n    hangup()\n  } else {\n    decline()\n  }\n  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+{"version":3,"file":"call.js","sources":["../../../src/lib/phone/call.ts"],"sourcesContent":["// Copyright (C) 2022 Nethesis S.r.l.\n// SPDX-License-Identifier: AGPL-3.0-or-later\n\nimport { getSupportedDevices } from '../devices/devices'\nimport Janus from '../webrtc/janus'\nimport {\n  call,\n  hangup,\n  decline,\n  answerWebRTC,\n  muteWebRTC,\n  unmuteWebRTC,\n  pauseWebRTC,\n  unpauseWebRTC,\n} from '../webrtc/messages'\nimport { store } from '../../store'\nimport { isWebRTC } from '../user/default_device'\nimport {\n  blindTransfer as blindTransferRequest,\n  attendedTransfer as attendedTransferRequest,\n} from '../../services/astproxy'\nimport dtmfAudios from '../../static/dtmf'\n\n/**\n * Starts a call to a number\n * @param number The number string\n */\nexport function callNumber(number: string, sipHost: string) {\n  const sipURI = `sip:${number}@${sipHost}`\n  callSipURI(sipURI)\n}\n\n/**\n * Starts a call to a SIP URI\n *\n * @param sipURI The SIP URI string\n */\nexport function callSipURI(sipURI: string) {\n  getSupportedDevices(async () => {\n    // @ts-ignore\n    Janus.log('This is a SIP call')\n    await call(sipURI, {\n      audio: {\n        mandatory: {\n          echoCancellation: false,\n          googEchoCancellation: false,\n          googAutoGainControl: false,\n          googAutoGainControl2: false,\n          googNoiseSuppression: false,\n          googHighpassFilter: false,\n          googTypingNoiseDetection: false,\n        },\n      },\n      audioSend: true,\n      audioRecv: true,\n      videoSend: false,\n      videoRecv: false,\n    })\n  })\n}\n\n/**\n * Answer incoming call\n */\nexport function answerIncomingCall() {\n  if (isWebRTC()) {\n    answerWebRTC()\n  }\n}\n\n/**\n * Hangup current call\n */\nexport function hangupCurrentCall() {\n  const { outgoing, accepted } = store.getState().currentCall\n  if (outgoing || accepted) {\n    hangup()\n  } else {\n    decline()\n  }\n  store.dispatch.player.stopAudioPlayer()\n  store.dispatch.currentCall.reset()\n}\n\n/**\n * Mute the current call\n */\nexport function muteCurrentCall() {\n  // Check the current user default device\n  if (isWebRTC()) {\n    const muted = muteWebRTC()\n    if (muted) {\n      store.dispatch.currentCall.updateCurrentCall({\n        muted: true,\n      })\n    }\n  }\n}\n\n/**\n * Unmute the current call\n */\nexport function unmuteCurrentCall() {\n  // Check the current user default device\n  if (isWebRTC()) {\n    const unmuted = unmuteWebRTC()\n    if (unmuted) {\n      store.dispatch.currentCall.updateCurrentCall({\n        muted: false,\n      })\n    }\n  }\n}\n\n/**\n * Pause the current call\n */\nexport function pauseCurrentCall() {\n  // Check the current user default device\n  if (isWebRTC()) {\n    const paused = pauseWebRTC()\n    if (paused) {\n      store.dispatch.currentCall.updateCurrentCall({\n        paused: true,\n      })\n      // Pause remote audio\n      store.dispatch.player.pauseRemoteAudio()\n    }\n  }\n}\n\n/**\n * Unpause the current call\n */\nexport function unpauseCurrentCall() {\n  // Check the current user default device\n  if (isWebRTC()) {\n    const unpaused = unpauseWebRTC()\n    if (unpaused) {\n      store.dispatch.currentCall.updateCurrentCall({\n        paused: false,\n      })\n      // Play remote audio\n      store.dispatch.player.playRemoteAudio()\n    }\n  }\n}\n\n/**\n * Transfer the current call through a blind transfer (not in use)\n */\nexport async function blindTransfer(number: string) {\n  // Retrieve current conversation info\n  const { conversationId } = store.getState().currentCall\n  const { default_device } = store.getState().currentUser\n  // Transfer the call through blind transfer\n  if (conversationId && default_device?.id && number) {\n    return await blindTransferRequest({\n      convid: conversationId,\n      to: number,\n      endpointId: default_device.id,\n    })\n  }\n}\n\n/**\n * Transfer the current call through a attended transfer\n 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package/dist/lib/webrtc/messages.d.ts

@@ -1,4 +1,9 @@
-export declare function register(sipExten: string, sipSecret: string): void;
+export declare function register({ sipExten, sipSecret, sipHost, sipPort, }: {
+    sipExten: string;
+    sipSecret: string;
+    sipHost: string;
+    sipPort: string;
+}): void;
 export declare function answerWebRTC(): void;
 export declare function decline(): void;
 export declare function hangup(): void;

package/dist/lib/webrtc/messages.js

@@ -1,2 +1,2 @@
-"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("./janus.js"),r=require("../../store/index.js"),t=require("../../node_modules/webrtc-adapter/src/js/adapter_core.js");exports.answerWebRTC=function(){var t=r.store.getState().webrtc,s=t.sipcall,a=t.jsepGlobal;s&&a&&s.createAnswer({jsep:a,media:{audio:!0,videoSend:!1,videoRecv:!1},success:function(e){s.send({message:{request:"accept"},jsep:e})},error:function(r){e.default.error("WebRTC error:",r),s.send({message:{request:"decline",code:480}})}})},exports.call=function(t,s){return new Promise((function(a,n){var o=r.store.getState().webrtc.sipcall;t&&s&&o.createOffer({media:s,success:function(r){e.default.debug("Got SDP!"),e.default.debug(r),o.send({message:{request:"call",uri:t},jsep:r}),a(!0)},error:function(r){e.default.error("ebRTC error...",r),e.default.error("WebRTC error call on createOffer: ",r),n(!1)}})}))},exports.decline=function(){var e=r.store.getState().webrtc.sipcall;e&&e.send({message:{request:"decline"}})},exports.handleRemote=function(e){var t=r.store.getState().webrtc.sipcall;t&&t.handleRemoteJsep({jsep:e,error:function(){t.send({message:{request:"hangup"}}),t.hangup()}})},exports.hangup=function(){var e=r.store.getState().webrtc.sipcall;e&&e.send({message:{request:"hangup"}})},exports.muteWebRTC=function(){var e=r.store.getState().webrtc.sipcall;return e.muteAudio(),e.isAudioMuted()},exports.pauseWebRTC=function(){var e=r.store.getState().webrtc.sipcall;try{return e.send({message:{request:"hold"}}),!0}catch(e){return console.error(e),!1}},exports.register=function(e,t){var s=r.store.getState().webrtc.sipcall,a=r.store.getState().currentUser.name;s&&s.send({message:{request:"register",username:"sip:"+e+"@127.0.0.1",display_name:a||"",secret:t,proxy:"sip:127.0.0.1:5060",sips:!1,refresh:!1}})},exports.sendDTMF=function(e){var s=r.store.getState().webrtc.sipcall;"chrome"===t.default.browserDetails.browser?s.dtmf({dtmf:{tones:e}}):s.send({message:{request:"dtmf_info",digit:e}})},exports.unmuteWebRTC=function(){var e=r.store.getState().webrtc.sipcall;return e.unmuteAudio(),!e.isAudioMuted()},exports.unpauseWebRTC=function(){var e=r.store.getState().webrtc.sipcall;try{return e.send({message:{request:"unhold"}}),!0}catch(e){return console.error(e),!1}},exports.unregister=function(){var e=r.store.getState().webrtc.sipcall;e&&e.send({message:{request:"unregister"}})};
+"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("./janus.js"),t=require("../../store/index.js"),r=require("../../node_modules/webrtc-adapter/src/js/adapter_core.js");exports.answerWebRTC=function(){var r=t.store.getState().webrtc,s=r.sipcall,o=r.jsepGlobal;s&&o&&s.createAnswer({jsep:o,media:{audio:!0,videoSend:!1,videoRecv:!1},success:function(e){s.send({message:{request:"accept"},jsep:e})},error:function(t){e.default.error("WebRTC error:",t),s.send({message:{request:"decline",code:480}})}})},exports.call=function(r,s){return new Promise((function(o,n){var a=t.store.getState().webrtc.sipcall;r&&s&&a.createOffer({media:s,success:function(t){e.default.debug("Got SDP!"),e.default.debug(t),a.send({message:{request:"call",uri:r},jsep:t}),o(!0)},error:function(t){e.default.error("ebRTC error...",t),e.default.error("WebRTC error call on createOffer: ",t),n(!1)}})}))},exports.decline=function(){var e=t.store.getState().webrtc.sipcall;e&&e.send({message:{request:"decline"}})},exports.handleRemote=function(e){var r=t.store.getState().webrtc.sipcall;r&&r.handleRemoteJsep({jsep:e,error:function(){r.send({message:{request:"hangup"}}),r.hangup()}})},exports.hangup=function(){var e=t.store.getState().webrtc.sipcall;e&&e.send({message:{request:"hangup"}})},exports.muteWebRTC=function(){var e=t.store.getState().webrtc.sipcall;return e.muteAudio(),e.isAudioMuted()},exports.pauseWebRTC=function(){var e=t.store.getState().webrtc.sipcall;try{return e.send({message:{request:"hold"}}),!0}catch(e){return console.error(e),!1}},exports.register=function(e){var r=e.sipExten,s=e.sipSecret,o=e.sipHost,n=e.sipPort,a=t.store.getState().webrtc.sipcall,c=t.store.getState().currentUser.name;a&&a.send({message:{request:"register",username:"sip:".concat(r,"@").concat(o),display_name:c||"",secret:s,proxy:"sip:".concat(o,":").concat(n),outbound_proxy:"sip:".concat(o,":").concat(n),sips:!1,refresh:!1}})},exports.sendDTMF=function(e){var s=t.store.getState().webrtc.sipcall;"chrome"===r.default.browserDetails.browser?s.dtmf({dtmf:{tones:e}}):s.send({message:{request:"dtmf_info",digit:e}})},exports.unmuteWebRTC=function(){var e=t.store.getState().webrtc.sipcall;return e.unmuteAudio(),!e.isAudioMuted()},exports.unpauseWebRTC=function(){var e=t.store.getState().webrtc.sipcall;try{return e.send({message:{request:"unhold"}}),!0}catch(e){return console.error(e),!1}},exports.unregister=function(){var e=t.store.getState().webrtc.sipcall;e&&e.send({message:{request:"unregister"}})};
 //# sourceMappingURL=messages.js.map

package/dist/lib/webrtc/messages.js.map

@@ -1 +1 @@
-{"version":3,"file":"messages.js","sources":["../../../src/lib/webrtc/messages.ts"],"sourcesContent":["// Copyright (C) 2022 Nethesis S.r.l.\n// SPDX-License-Identifier: AGPL-3.0-or-later\n\nimport Janus from './janus'\nimport { store } from '../../store'\nimport adapter from 'webrtc-adapter'\n\nexport function register(sipExten: string, sipSecret: string) {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  const { name } = store.getState().currentUser\n  if (sipcall) {\n    sipcall.send({\n      message: {\n        request: 'register',\n        username: 'sip:' + sipExten + '@' + '127.0.0.1',\n        display_name: name || '',\n        secret: sipSecret,\n        proxy: 'sip:' + '127.0.0.1' + ':5060',\n        sips: false,\n        refresh: false,\n      },\n    })\n  }\n}\n\nexport function answerWebRTC() {\n  const { sipcall, jsepGlobal }: { sipcall: any; jsepGlobal: any } = store.getState().webrtc\n  if (sipcall && jsepGlobal) {\n    sipcall.createAnswer({\n      jsep: jsepGlobal,\n      media: {\n        audio: true,\n        videoSend: false,\n        videoRecv: false,\n      },\n      success: (jsep) => {\n        sipcall.send({\n          message: {\n            request: 'accept',\n          },\n          jsep: jsep,\n        })\n      },\n      error: (error) => {\n        // @ts-ignore\n        Janus.error('WebRTC error:', error)\n        sipcall.send({\n          message: {\n            request: 'decline',\n            code: 480,\n          },\n        })\n      },\n    })\n  }\n}\n\nexport function decline() {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (sipcall) {\n    sipcall.send({\n      message: {\n        request: 'decline',\n      },\n    })\n  }\n}\n\nexport function hangup() {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (sipcall) {\n    sipcall.send({\n      message: {\n        request: 'hangup',\n      },\n    })\n  }\n}\n\nexport function unregister() {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (sipcall) {\n    sipcall.send({\n      message: {\n        request: 'unregister',\n      },\n    })\n  }\n}\n\nexport function handleRemote(jsep) {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (sipcall) {\n    sipcall.handleRemoteJsep({\n      jsep: jsep,\n      error: function () {\n        var hangup = {\n          request: 'hangup',\n        }\n        sipcall.send({\n          message: hangup,\n        })\n        sipcall.hangup()\n      },\n    })\n  }\n}\n\n/**\n * Starts a call to a sipuri\n *\n * @param sipURI The sipuri to be called\n * @param mediaObj The object containing the media constrains\n * @returns\n */\nexport function call(sipURI: string, mediaObj: object) {\n  return new Promise((resolve, reject) => {\n    const { sipcall }: { sipcall: any } = store.getState().webrtc\n    if (sipURI && mediaObj) {\n      sipcall.createOffer({\n        media: mediaObj,\n        success: function (jsep: any) {\n          // @ts-ignore\n          Janus.debug('Got SDP!')\n          // @ts-ignore\n          Janus.debug(jsep)\n          sipcall.send({\n            message: {\n              request: 'call',\n              uri: sipURI,\n            },\n            jsep: jsep,\n          })\n          resolve(true)\n        },\n        error: function (error) {\n          // @ts-ignore\n          Janus.error('ebRTC error...', error)\n          // @ts-ignore\n          Janus.error('WebRTC error call on createOffer: ', error)\n          reject(false)\n        },\n      })\n    }\n  })\n}\n\n/**\n * Mute current call so the counterpart can't listen the current user\n * @returns The muted status\n */\nexport function muteWebRTC(): boolean {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  // Uset the janus library functions to mute call\n  sipcall.muteAudio()\n  return sipcall.isAudioMuted()\n}\n\n/**\n * Unmute current call so the counterpart can listen the current user\n * @returns The muted status\n */\nexport function unmuteWebRTC(): boolean {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  // Use the janus library functions to unmute call\n  sipcall.unmuteAudio()\n  return !sipcall.isAudioMuted()\n}\n\n/**\n * Pause current call so the counterpart listens the pause ringtone\n * @returns The true if no errors occurs\n */\nexport function pauseWebRTC() {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  // Send hold message\n  try {\n    sipcall.send({\n      message: {\n        request: 'hold',\n      },\n    })\n    return true\n  } catch (err) {\n    console.error(err)\n    return false\n  }\n}\n\n/**\n * Unpause the current call so the counterpart listens the current user\n * @returns The true if no errors occurs\n */\nexport function unpauseWebRTC() {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  // Send hold message\n  try {\n    sipcall.send({\n      message: {\n        request: 'unhold',\n      },\n    })\n    return true\n  } catch (err) {\n    console.error(err)\n    return false\n  }\n}\n\n/**\n * Send DTMF messages to Janus\n */\nexport function sendDTMF(key: string) {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (adapter.browserDetails.browser === 'chrome') {\n    sipcall.dtmf({\n      dtmf: {\n        tones: key,\n      },\n    })\n  } else {\n    sipcall.send({\n      message: {\n        request: 'dtmf_info',\n        digit: key,\n      },\n    })\n  }\n}\n"],"names":["_a","store","getState","webrtc","sipcall","jsepGlobal","createAnswer","jsep","media","audio","videoSend","videoRecv","success","send","message","request","error","Janus","code","sipURI","mediaObj","Promise","resolve","reject","createOffer","debug","uri","handleRemoteJsep","hangup","muteAudio","isAudioMuted","err","console","sipExten","sipSecret","name","currentUser","username","display_name","secret","proxy","sips","refresh","key","adapter","browserDetails","browser","dtmf","tones","digit","unmuteAudio"],"mappings":"wOA0BQ,IAAAA,EAA6DC,EAAAA,MAAMC,WAAWC,OAA5EC,EAAOJ,EAAAI,QAAEC,eACbD,GAAWC,GACbD,EAAQE,aAAa,CACnBC,KAAMF,EACNG,MAAO,CACLC,OAAO,EACPC,WAAW,EACXC,WAAW,GAEbC,QAAS,SAACL,GACRH,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,UAEXR,KAAMA,GAET,EACDS,MAAO,SAACA,GAENC,EAAAA,QAAMD,MAAM,gBAAiBA,GAC7BZ,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,UACTG,KAAM,MAGX,GAGP,eA4DgB,SAAKC,EAAgBC,GACnC,OAAO,IAAIC,SAAQ,SAACC,EAASC,GACnB,IAAAnB,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QACzDe,GAAUC,GACZhB,EAAQoB,YAAY,CAClBhB,MAAOY,EACPR,QAAS,SAAUL,GAEjBU,UAAMQ,MAAM,YAEZR,UAAMQ,MAAMlB,GACZH,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,OACTW,IAAKP,GAEPZ,KAAMA,IAERe,GAAQ,EACT,EACDN,MAAO,SAAUA,GAEfC,EAAAA,QAAMD,MAAM,iBAAkBA,GAE9BC,EAAAA,QAAMD,MAAM,qCAAsCA,GAClDO,GAAO,EACR,GAGP,GACF,6BAvFU,IAAAnB,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QACzDA,GACFA,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,YAIjB,uBAwBM,SAAuBR,GACnB,IAAAH,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QACzDA,GACFA,EAAQuB,iBAAiB,CACvBpB,KAAMA,EACNS,MAAO,WAILZ,EAAQS,KAAK,CACXC,QAJW,CACXC,QAAS,YAKXX,EAAQwB,QACT,GAGP,4BArCU,IAAAxB,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QACzDA,GACFA,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,WAIjB,gCA4EU,IAAAX,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QAG7D,OADAA,EAAQyB,YACDzB,EAAQ0B,cACjB,iCAoBU,IAAA1B,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QAE7D,IAME,OALAA,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,WAGN,CAIR,CAHC,MAAOgB,GAEP,OADAC,QAAQhB,MAAMe,IACP,CACR,CACH,mBAvLgB,SAASE,EAAkBC,GACjC,IAAA9B,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QACrD+B,EAASlC,EAAKA,MAACC,WAAWkC,YAAWD,KACzC/B,GACFA,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,WACTsB,SAAU,OAASJ,EAAT,aACVK,aAAcH,GAAQ,GACtBI,OAAQL,EACRM,MAAO,qBACPC,MAAM,EACNC,SAAS,IAIjB,mBAiMM,SAAmBC,GAEf,IAAAvC,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QACtB,WAAnCwC,UAAQC,eAAeC,QACzB1C,EAAQ2C,KAAK,CACXA,KAAM,CACJC,MAAOL,KAIXvC,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,YACTkC,MAAON,IAIf,kCApEU,IAAAvC,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QAG7D,OADAA,EAAQ8C,eACA9C,EAAQ0B,cAClB,mCA6BU,IAAA1B,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QAE7D,IAME,OALAA,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,aAGN,CAIR,CAHC,MAAOgB,GAEP,OADAC,QAAQhB,MAAMe,IACP,CACR,CACH,gCAnIU,IAAA3B,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QACzDA,GACFA,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,eAIjB"}
+{"version":3,"file":"messages.js","sources":["../../../src/lib/webrtc/messages.ts"],"sourcesContent":["// Copyright (C) 2022 Nethesis S.r.l.\n// SPDX-License-Identifier: AGPL-3.0-or-later\n\nimport Janus from './janus'\nimport { store } from '../../store'\nimport adapter from 'webrtc-adapter'\n\nexport function register({\n  sipExten,\n  sipSecret,\n  sipHost,\n  sipPort,\n}: {\n  sipExten: string\n  sipSecret: string\n  sipHost: string\n  sipPort: string\n}) {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  const { name } = store.getState().currentUser\n  if (sipcall) {\n    sipcall.send({\n      message: {\n        request: 'register',\n        username: `sip:${sipExten}@${sipHost}`,\n        display_name: name || '',\n        secret: sipSecret,\n        proxy: `sip:${sipHost}:${sipPort}`,\n        outbound_proxy: `sip:${sipHost}:${sipPort}`,\n        sips: false,\n        refresh: false,\n      },\n    })\n  }\n}\n\nexport function answerWebRTC() {\n  const { sipcall, jsepGlobal }: { sipcall: any; jsepGlobal: any } = store.getState().webrtc\n  if (sipcall && jsepGlobal) {\n    sipcall.createAnswer({\n      jsep: jsepGlobal,\n      media: {\n        audio: true,\n        videoSend: false,\n        videoRecv: false,\n      },\n      success: (jsep) => {\n        sipcall.send({\n          message: {\n            request: 'accept',\n          },\n          jsep: jsep,\n        })\n      },\n      error: (error) => {\n        // @ts-ignore\n        Janus.error('WebRTC error:', error)\n        sipcall.send({\n          message: {\n            request: 'decline',\n            code: 480,\n          },\n        })\n      },\n    })\n  }\n}\n\nexport function decline() {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (sipcall) {\n    sipcall.send({\n      message: {\n        request: 'decline',\n      },\n    })\n  }\n}\n\nexport function hangup() {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (sipcall) {\n    sipcall.send({\n      message: {\n        request: 'hangup',\n      },\n    })\n  }\n}\n\nexport function unregister() {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (sipcall) {\n    sipcall.send({\n      message: {\n        request: 'unregister',\n      },\n    })\n  }\n}\n\nexport function handleRemote(jsep) {\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (sipcall) {\n    sipcall.handleRemoteJsep({\n      jsep: jsep,\n      error: function () {\n        var hangup = {\n          request: 'hangup',\n        }\n        sipcall.send({\n          message: hangup,\n        })\n        sipcall.hangup()\n      },\n    })\n  }\n}\n\n/**\n * Starts a call to a sipuri\n *\n * @param sipURI The sipuri to be called\n * @param mediaObj The object containing the media constrains\n * @returns\n */\nexport function call(sipURI: string, mediaObj: object) {\n  return new Promise((resolve, reject) => {\n    const { sipcall }: { sipcall: any } = store.getState().webrtc\n    if (sipURI && mediaObj) {\n      sipcall.createOffer({\n        media: mediaObj,\n        success: function (jsep: any) {\n          // @ts-ignore\n          Janus.debug('Got SDP!')\n          // @ts-ignore\n          Janus.debug(jsep)\n          sipcall.send({\n            message: {\n              request: 'call',\n              uri: sipURI,\n            },\n            jsep: jsep,\n          })\n          resolve(true)\n        },\n        error: function (error) {\n          // @ts-ignore\n          Janus.error('ebRTC error...', error)\n          // @ts-ignore\n          Janus.error('WebRTC error call on createOffer: ', error)\n          reject(false)\n        },\n      })\n    }\n  })\n}\n\n/**\n * Mute current call so the counterpart can't listen the current user\n * @returns The muted status\n */\nexport function muteWebRTC(): boolean {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  // Uset the janus library functions to mute call\n  sipcall.muteAudio()\n  return sipcall.isAudioMuted()\n}\n\n/**\n * Unmute current call so the counterpart can listen the current user\n * @returns The muted status\n */\nexport function unmuteWebRTC(): boolean {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  // Use the janus library functions to unmute call\n  sipcall.unmuteAudio()\n  return !sipcall.isAudioMuted()\n}\n\n/**\n * Pause current call so the counterpart listens the pause ringtone\n * @returns The true if no errors occurs\n */\nexport function pauseWebRTC() {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  // Send hold message\n  try {\n    sipcall.send({\n      message: {\n        request: 'hold',\n      },\n    })\n    return true\n  } catch (err) {\n    console.error(err)\n    return false\n  }\n}\n\n/**\n * Unpause the current call so the counterpart listens the current user\n * @returns The true if no errors occurs\n */\nexport function unpauseWebRTC() {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  // Send hold message\n  try {\n    sipcall.send({\n      message: {\n        request: 'unhold',\n      },\n    })\n    return true\n  } catch (err) {\n    console.error(err)\n    return false\n  }\n}\n\n/**\n * Send DTMF messages to Janus\n */\nexport function sendDTMF(key: string) {\n  // Initialize sipcall\n  const { sipcall }: { sipcall: any } = store.getState().webrtc\n  if (adapter.browserDetails.browser === 'chrome') {\n    sipcall.dtmf({\n      dtmf: {\n        tones: key,\n      },\n    })\n  } else {\n    sipcall.send({\n      message: {\n        request: 'dtmf_info',\n        digit: key,\n      },\n    })\n  }\n}\n"],"names":["_a","store","getState","webrtc","sipcall","jsepGlobal","createAnswer","jsep","media","audio","videoSend","videoRecv","success","send","message","request","error","Janus","code","sipURI","mediaObj","Promise","resolve","reject","createOffer","debug","uri","handleRemoteJsep","hangup","muteAudio","isAudioMuted","err","console","sipExten","sipSecret","sipHost","sipPort","name","currentUser","username","concat","display_name","secret","proxy","outbound_proxy","sips","refresh","key","adapter","browserDetails","browser","dtmf","tones","digit","unmuteAudio"],"mappings":"wOAqCQ,IAAAA,EAA6DC,EAAAA,MAAMC,WAAWC,OAA5EC,EAAOJ,EAAAI,QAAEC,eACbD,GAAWC,GACbD,EAAQE,aAAa,CACnBC,KAAMF,EACNG,MAAO,CACLC,OAAO,EACPC,WAAW,EACXC,WAAW,GAEbC,QAAS,SAACL,GACRH,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,UAEXR,KAAMA,GAET,EACDS,MAAO,SAACA,GAENC,EAAAA,QAAMD,MAAM,gBAAiBA,GAC7BZ,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,UACTG,KAAM,MAGX,GAGP,eA4DgB,SAAKC,EAAgBC,GACnC,OAAO,IAAIC,SAAQ,SAACC,EAASC,GACnB,IAAAnB,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QACzDe,GAAUC,GACZhB,EAAQoB,YAAY,CAClBhB,MAAOY,EACPR,QAAS,SAAUL,GAEjBU,UAAMQ,MAAM,YAEZR,UAAMQ,MAAMlB,GACZH,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,OACTW,IAAKP,GAEPZ,KAAMA,IAERe,GAAQ,EACT,EACDN,MAAO,SAAUA,GAEfC,EAAAA,QAAMD,MAAM,iBAAkBA,GAE9BC,EAAAA,QAAMD,MAAM,qCAAsCA,GAClDO,GAAO,EACR,GAGP,GACF,6BAvFU,IAAAnB,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QACzDA,GACFA,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,YAIjB,uBAwBM,SAAuBR,GACnB,IAAAH,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QACzDA,GACFA,EAAQuB,iBAAiB,CACvBpB,KAAMA,EACNS,MAAO,WAILZ,EAAQS,KAAK,CACXC,QAJW,CACXC,QAAS,YAKXX,EAAQwB,QACT,GAGP,4BArCU,IAAAxB,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QACzDA,GACFA,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,WAIjB,gCA4EU,IAAAX,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QAG7D,OADAA,EAAQyB,YACDzB,EAAQ0B,cACjB,iCAoBU,IAAA1B,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QAE7D,IAME,OALAA,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,WAGN,CAIR,CAHC,MAAOgB,GAEP,OADAC,QAAQhB,MAAMe,IACP,CACR,CACH,mBAlMM,SAAmB/B,OACvBiC,EAAQjC,EAAAiC,SACRC,EAASlC,EAAAkC,UACTC,EAAOnC,EAAAmC,QACPC,EAAOpC,EAAAoC,QAOChC,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QACrDiC,EAASpC,EAAKA,MAACC,WAAWoC,YAAWD,KACzCjC,GACFA,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,WACTwB,SAAU,OAAAC,OAAOP,EAAQ,KAAAO,OAAIL,GAC7BM,aAAcJ,GAAQ,GACtBK,OAAQR,EACRS,MAAO,OAAAH,OAAOL,EAAO,KAAAK,OAAIJ,GACzBQ,eAAgB,OAAAJ,OAAOL,EAAO,KAAAK,OAAIJ,GAClCS,MAAM,EACNC,SAAS,IAIjB,mBAiMM,SAAmBC,GAEf,IAAA3C,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QACtB,WAAnC4C,UAAQC,eAAeC,QACzB9C,EAAQ+C,KAAK,CACXA,KAAM,CACJC,MAAOL,KAIX3C,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,YACTsC,MAAON,IAIf,kCApEU,IAAA3C,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QAG7D,OADAA,EAAQkD,eACAlD,EAAQ0B,cAClB,mCA6BU,IAAA1B,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QAE7D,IAME,OALAA,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,aAGN,CAIR,CAHC,MAAOgB,GAEP,OADAC,QAAQhB,MAAMe,IACP,CACR,CACH,gCAnIU,IAAA3B,EAA8BH,EAAKA,MAACC,WAAWC,OAAMC,QACzDA,GACFA,EAAQS,KAAK,CACXC,QAAS,CACPC,QAAS,eAIjB"}

package/dist/models/currentCall.d.ts

@@ -24,6 +24,7 @@ export declare const currentCall: {
         transferringNumber: string;
         transferringStartTime: string;
         transferSwitching: boolean;
+        transferCalls: any[];
     };
 } & {
     effects: (dispatch: import("@rematch/core").RematchDispatch<RootModel>) => {
@@ -56,7 +57,9 @@ export declare const currentCall: {
             transferringNumber: string;
             transferringStartTime: string;
             transferSwitching: boolean;
+            transferCalls: any[];
         }, payload: CurrentCallTypes) => {
+            transferCalls: any[];
             displayName: string;
             username: string;
             number: string;
@@ -103,6 +106,7 @@ export declare const currentCall: {
             transferringNumber: string;
             transferringStartTime: string;
             transferSwitching: boolean;
+            transferCalls: any[];
         }, payload: string) => {
             keypadValue: string;
             displayName: string;
@@ -126,6 +130,7 @@ export declare const currentCall: {
             transferringNumber: string;
             transferringStartTime: string;
             transferSwitching: boolean;
+            transferCalls: any[];
         };
         updateTransferring: (state: {
             displayName: string;
@@ -150,6 +155,7 @@ export declare const currentCall: {
             transferringNumber: string;
             transferringStartTime: string;
             transferSwitching: boolean;
+            transferCalls: any[];
         }, payload: boolean) => {
             transferring: boolean;
             displayName: string;
@@ -173,6 +179,7 @@ export declare const currentCall: {
             transferringNumber: string;
             transferringStartTime: string;
             transferSwitching: boolean;
+            transferCalls: any[];
         };
         updateTransferSwitching: (state: {
             displayName: string;
@@ -197,6 +204,7 @@ export declare const currentCall: {
             transferringNumber: string;
             transferringStartTime: string;
             transferSwitching: boolean;
+            transferCalls: any[];
         }, payload: boolean) => {
             transferSwitching: boolean;
             displayName: string;
@@ -220,6 +228,56 @@ export declare const currentCall: {
             transferringName: string;
             transferringNumber: string;
             transferringStartTime: string;
+            transferCalls: any[];
+        };
+        addTransferCalls: (state: {
+            displayName: string;
+            username: string;
+            number: string;
+            incomingSocket: boolean;
+            incomingWebRTC: boolean;
+            incoming: boolean;
+            acceptedSocket: boolean;
+            acceptedWebRTC: boolean;
+            accepted: boolean;
+            outgoingSocket: boolean;
+            outgoingWebRTC: boolean;
+            outgoing: boolean;
+            startTime: string;
+            muted: boolean;
+            paused: boolean;
+            keypadValue: string;
+            conversationId: string;
+            transferring: boolean;
+            transferringName: string;
+            transferringNumber: string;
+            transferringStartTime: string;
+            transferSwitching: boolean;
+            transferCalls: any[];
+        }, payload: TransferCallsTypes) => {
+            displayName: string;
+            username: string;
+            number: string;
+            incomingSocket: boolean;
+            incomingWebRTC: boolean;
+            incoming: boolean;
+            acceptedSocket: boolean;
+            acceptedWebRTC: boolean;
+            accepted: boolean;
+            outgoingSocket: boolean;
+            outgoingWebRTC: boolean;
+            outgoing: boolean;
+            startTime: string;
+            muted: boolean;
+            paused: boolean;
+            keypadValue: string;
+            conversationId: string;
+            transferring: boolean;
+            transferringName: string;
+            transferringNumber: string;
+            transferringStartTime: string;
+            transferSwitching: boolean;
+            transferCalls: any[];
         };
         reset: () => {
             displayName: string;
@@ -244,6 +302,7 @@ export declare const currentCall: {
             transferringNumber: string;
             transferringStartTime: string;
             transferSwitching: boolean;
+            transferCalls: any[];
         };
     };
 } & ({} | {
@@ -270,8 +329,15 @@ export declare const currentCall: {
         transferringNumber: string;
         transferringStartTime: string;
         transferSwitching: boolean;
+        transferCalls: any[];
     }, import("redux").AnyAction>;
 });
+export type TransferCallsTypes = {
+    type: 'transferred' | 'destination';
+    displayName: string;
+    number: string;
+    startTime: string;
+};
 export interface CurrentCallTypes {
     displayName?: string;
     username?: string;
@@ -294,4 +360,5 @@ export interface CurrentCallTypes {
     transferringNumber?: string;
     transferringStartTime?: string;
     transferSwitching?: boolean;
+    transferCalls?: TransferCallsTypes[];
 }

package/dist/models/currentCall.js

@@ -1,2 +1,2 @@
-"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("../node_modules/tslib/tslib.es6.js"),r=require("../node_modules/@rematch/core/dist/core.esm.js"),t=require("../static/incoming_ringtone.js"),n=require("../events/CallEvents.js");require("react"),require("../store/index.js"),require("../node_modules/react-redux/es/index.js");var i={displayName:"",username:"",number:"",incomingSocket:!1,incomingWebRTC:!1,incoming:!1,acceptedSocket:!1,acceptedWebRTC:!1,accepted:!1,outgoingSocket:!1,outgoingWebRTC:!1,outgoing:!1,startTime:"",muted:!1,paused:!1,keypadValue:"",conversationId:"",transferring:!1,transferringName:"",transferringNumber:"",transferringStartTime:"",transferSwitching:!1},a=r.createModel()({state:i,reducers:{updateCurrentCall:function(r,t){return e.__assign(e.__assign({},r),t)},updateKeypadValue:function(r,t){return e.__assign(e.__assign({},r),{keypadValue:t})},updateTransferring:function(r,t){return e.__assign(e.__assign({},r),{transferring:t})},updateTransferSwitching:function(r,t){return e.__assign(e.__assign({},r),{transferSwitching:t})},reset:function(){return i}},effects:function(r){return{checkIncomingUpdatePlay:function(n,i){var a,u;("webrtc"===(null===(a=i.currentUser.default_device)||void 0===a?void 0:a.type)&&n.incomingWebRTC||"physical"===(null===(u=i.currentUser.default_device)||void 0===u?void 0:u.type)&&n.incomingSocket)&&(n.incoming=!0,r.player.updateStartAudioPlayer({src:t.default,loop:!0})),r.currentCall.updateCurrentCall(e.__assign({},n))},checkOutgoingUpdate:function(t,i){var a,u;("webrtc"===(null===(a=i.currentUser.default_device)||void 0===a?void 0:a.type)&&t.outgoingWebRTC||"physical"===(null===(u=i.currentUser.default_device)||void 0===u?void 0:u.type)&&t.outgoingSocket)&&(t.outgoing=!0,n.dispatchOutgoingCallStarted(t.displayName,t.number)),r.currentCall.updateCurrentCall(e.__assign({},t))},checkAcceptedUpdate:function(t,n){var i,a;("webrtc"===(null===(i=n.currentUser.default_device)||void 0===i?void 0:i.type)&&t.acceptedWebRTC||"physical"===(null===(a=n.currentUser.default_device)||void 0===a?void 0:a.type)&&t.acceptedSocket)&&(t.accepted=!0),r.currentCall.updateCurrentCall(e.__assign({},t))}}}});exports.currentCall=a;
+"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("../node_modules/tslib/tslib.es6.js"),r=require("../node_modules/@rematch/core/dist/core.esm.js"),n=require("../static/incoming_ringtone.js"),t=require("../events/CallEvents.js");require("react"),require("../store/index.js"),require("../node_modules/react-redux/es/index.js");var a={displayName:"",username:"",number:"",incomingSocket:!1,incomingWebRTC:!1,incoming:!1,acceptedSocket:!1,acceptedWebRTC:!1,accepted:!1,outgoingSocket:!1,outgoingWebRTC:!1,outgoing:!1,startTime:"",muted:!1,paused:!1,keypadValue:"",conversationId:"",transferring:!1,transferringName:"",transferringNumber:"",transferringStartTime:"",transferSwitching:!1,transferCalls:new Array},i=r.createModel()({state:a,reducers:{updateCurrentCall:function(r,n){return e.__assign(e.__assign(e.__assign({},r),n),{transferCalls:r.transferCalls})},updateKeypadValue:function(r,n){return e.__assign(e.__assign({},r),{keypadValue:n})},updateTransferring:function(r,n){return e.__assign(e.__assign({},r),{transferring:n})},updateTransferSwitching:function(r,n){return e.__assign(e.__assign({},r),{transferSwitching:n})},addTransferCalls:function(r,n){return r.transferCalls.find((function(e){return e.number===n.number}))?r:e.__assign(e.__assign({},r),{transferCalls:e.__spreadArray(e.__spreadArray([],r.transferCalls,!0),[n],!1)})},reset:function(){return a}},effects:function(r){return{checkIncomingUpdatePlay:function(t,a){var i,s;("webrtc"===(null===(i=a.currentUser.default_device)||void 0===i?void 0:i.type)&&t.incomingWebRTC||"physical"===(null===(s=a.currentUser.default_device)||void 0===s?void 0:s.type)&&t.incomingSocket)&&(t.incoming=!0,r.player.updateStartAudioPlayer({src:n.default,loop:!0})),r.currentCall.updateCurrentCall(e.__assign({},t))},checkOutgoingUpdate:function(n,a){var i,s;("webrtc"===(null===(i=a.currentUser.default_device)||void 0===i?void 0:i.type)&&n.outgoingWebRTC||"physical"===(null===(s=a.currentUser.default_device)||void 0===s?void 0:s.type)&&n.outgoingSocket)&&(n.outgoing=!0,t.dispatchOutgoingCallStarted(n.displayName,n.number)),r.currentCall.updateCurrentCall(e.__assign({},n))},checkAcceptedUpdate:function(n,t){var a,i;("webrtc"===(null===(a=t.currentUser.default_device)||void 0===a?void 0:a.type)&&n.acceptedWebRTC||"physical"===(null===(i=t.currentUser.default_device)||void 0===i?void 0:i.type)&&n.acceptedSocket)&&(n.accepted=!0),r.currentCall.updateCurrentCall(e.__assign({},n))}}}});exports.currentCall=i;
 //# sourceMappingURL=currentCall.js.map

package/dist/models/currentCall.js.map

@@ -1 +1 @@
-{"version":3,"file":"currentCall.js","sources":["../../src/models/currentCall.ts"],"sourcesContent":["// Copyright (C) 2022 Nethesis S.r.l.\n// SPDX-License-Identifier: AGPL-3.0-or-later\n\nimport { createModel } from '@rematch/core'\nimport type { RootModel } from '.'\nimport incomingRingtone from '../static/incoming_ringtone'\nimport { dispatchOutgoingCallStarted } from '../events/index'\n\nconst defaultState = {\n  displayName: '',\n  username: '',\n  number: '',\n  incomingSocket: false,\n  incomingWebRTC: false,\n  incoming: false,\n  acceptedSocket: false,\n  acceptedWebRTC: false,\n  accepted: false,\n  outgoingSocket: false,\n  outgoingWebRTC: false,\n  outgoing: false,\n  startTime: '',\n  muted: false,\n  paused: false,\n  keypadValue: '',\n  conversationId: '',\n  transferring: false,\n  transferringName: '',\n  transferringNumber: '',\n  transferringStartTime: '',\n  transferSwitching: false,\n}\n\nexport const currentCall = createModel<RootModel>()({\n  state: defaultState,\n  reducers: {\n    updateCurrentCall: (state, payload: CurrentCallTypes) => {\n      return {\n        ...state,\n        ...payload,\n      }\n    },\n    updateKeypadValue: (state, payload: string) => {\n      return {\n        ...state,\n        keypadValue: payload,\n      }\n    },\n    updateTransferring: (state, payload: boolean) => {\n      return {\n        ...state,\n        transferring: payload,\n      }\n    },\n    updateTransferSwitching: (state, payload: boolean) => {\n      return {\n        ...state,\n        transferSwitching: payload,\n      }\n    },\n    reset: () => {\n      return defaultState\n    },\n  },\n  effects: (dispatch) => ({\n    checkIncomingUpdatePlay: (payload: CurrentCallTypes, rootState) => {\n      // Check call type and incoming confirmation source\n      if (\n        (rootState.currentUser.default_device?.type === 'webrtc' && payload.incomingWebRTC) ||\n        (rootState.currentUser.default_device?.type === 'physical' && payload.incomingSocket)\n      ) {\n        payload.incoming = true\n\n        // Update local player and play the audio\n        dispatch.player.updateStartAudioPlayer({ src: incomingRingtone, loop: true })\n      }\n      // Update the current call values and set incoming\n      dispatch.currentCall.updateCurrentCall({\n        ...payload,\n      })\n    },\n    checkOutgoingUpdate: (payload: CurrentCallTypes, rootState) => {\n      // Check call type and outgoing confirmation source\n      if (\n        (rootState.currentUser.default_device?.type === 'webrtc' && payload.outgoingWebRTC) ||\n        (rootState.currentUser.default_device?.type === 'physical' && payload.outgoingSocket)\n      ) {\n        payload.outgoing = true\n        // Dispatch an event for outgoing call\n        dispatchOutgoingCallStarted(payload.displayName, payload.number)\n      }\n      // Update the current call values and set outgoing\n      dispatch.currentCall.updateCurrentCall({\n        ...payload,\n      })\n    },\n    checkAcceptedUpdate: (payload: CurrentCallTypes, rootState) => {\n      // Check call type and accepted confirmation source\n      if (\n        (rootState.currentUser.default_device?.type === 'webrtc' && payload.acceptedWebRTC) ||\n        (rootState.currentUser.default_device?.type === 'physical' && payload.acceptedSocket)\n      ) {\n        payload.accepted = true\n        // TODO - dispatch accepted event\n      }\n      // Update the current call values\n      dispatch.currentCall.updateCurrentCall({\n        ...payload,\n      })\n    },\n  }),\n})\n\nexport interface CurrentCallTypes {\n  displayName?: string\n  username?: string\n  number?: string\n  incomingSocket?: boolean\n  incomingWebRTC?: boolean\n  incoming?: boolean\n  acceptedSocket?: boolean\n  acceptedWebRTC?: boolean\n  accepted?: boolean\n  outgoingSocket?: boolean\n  outgoingWebRTC?: boolean\n  outgoing?: boolean\n  startTime?: string\n  muted?: boolean\n  paused?: boolean\n  conversationId?: string\n  transferring?: boolean\n  transferringName?: string\n  transferringNumber?: string\n  transferringStartTime?: string\n  transferSwitching?: boolean\n}\n"],"names":["defaultState","displayName","username","number","incomingSocket","incomingWebRTC","incoming","acceptedSocket","acceptedWebRTC","accepted","outgoingSocket","outgoingWebRTC","outgoing","startTime","muted","paused","keypadValue","conversationId","transferring","transferringName","transferringNumber","transferringStartTime","transferSwitching","currentCall","createModel","state","reducers","updateCurrentCall","payload","__assign","updateKeypadValue","updateTransferring","updateTransferSwitching","reset","effects","dispatch","checkIncomingUpdatePlay","rootState","_a","currentUser","default_device","type","_b","player","updateStartAudioPlayer","src","incomingRingtone","loop","checkOutgoingUpdate","dispatchOutgoingCallStarted","checkAcceptedUpdate"],"mappings":"sWAQA,IAAMA,EAAe,CACnBC,YAAa,GACbC,SAAU,GACVC,OAAQ,GACRC,gBAAgB,EAChBC,gBAAgB,EAChBC,UAAU,EACVC,gBAAgB,EAChBC,gBAAgB,EAChBC,UAAU,EACVC,gBAAgB,EAChBC,gBAAgB,EAChBC,UAAU,EACVC,UAAW,GACXC,OAAO,EACPC,QAAQ,EACRC,YAAa,GACbC,eAAgB,GAChBC,cAAc,EACdC,iBAAkB,GAClBC,mBAAoB,GACpBC,sBAAuB,GACvBC,mBAAmB,GAGRC,EAAcC,EAAWA,aAAXA,CAAyB,CAClDC,MAAOzB,EACP0B,SAAU,CACRC,kBAAmB,SAACF,EAAOG,GACzB,OACKC,EAAAA,SAAAA,EAAAA,SAAA,CAAA,EAAAJ,GACAG,EAEN,EACDE,kBAAmB,SAACL,EAAOG,GACzB,OAAAC,EAAAA,SAAAA,EAAAA,SAAA,GACKJ,GAAK,CACRT,YAAaY,GAEhB,EACDG,mBAAoB,SAACN,EAAOG,GAC1B,OAAAC,EAAAA,SAAAA,EAAAA,SAAA,GACKJ,GAAK,CACRP,aAAcU,GAEjB,EACDI,wBAAyB,SAACP,EAAOG,GAC/B,OAAAC,EAAAA,SAAAA,EAAAA,SAAA,GACKJ,GAAK,CACRH,kBAAmBM,GAEtB,EACDK,MAAO,WACL,OAAOjC,CACR,GAEHkC,QAAS,SAACC,GAAa,MAAC,CACtBC,wBAAyB,SAACR,EAA2BS,YAGD,YAAT,QAAtCC,EAAAD,EAAUE,YAAYC,sBAAgB,IAAAF,OAAA,EAAAA,EAAAG,OAAqBb,EAAQvB,gBACpB,cAAX,QAApCqC,EAAAL,EAAUE,YAAYC,sBAAc,IAAAE,OAAA,EAAAA,EAAED,OAAuBb,EAAQxB,kBAEtEwB,EAAQtB,UAAW,EAGnB6B,EAASQ,OAAOC,uBAAuB,CAAEC,IAAKC,EAAAA,QAAkBC,MAAM,KAGxEZ,EAASZ,YAAYI,kBAChBE,EAAAA,SAAA,CAAA,EAAAD,GAEN,EACDoB,oBAAqB,SAACpB,EAA2BS,YAGG,YAAT,QAAtCC,EAAAD,EAAUE,YAAYC,sBAAgB,IAAAF,OAAA,EAAAA,EAAAG,OAAqBb,EAAQjB,gBACpB,cAAX,QAApC+B,EAAAL,EAAUE,YAAYC,sBAAc,IAAAE,OAAA,EAAAA,EAAED,OAAuBb,EAAQlB,kBAEtEkB,EAAQhB,UAAW,EAEnBqC,EAAAA,4BAA4BrB,EAAQ3B,YAAa2B,EAAQzB,SAG3DgC,EAASZ,YAAYI,kBAChBE,EAAAA,SAAA,CAAA,EAAAD,GAEN,EACDsB,oBAAqB,SAACtB,EAA2BS,YAGG,YAAT,QAAtCC,EAAAD,EAAUE,YAAYC,sBAAgB,IAAAF,OAAA,EAAAA,EAAAG,OAAqBb,EAAQpB,gBACpB,cAAX,QAApCkC,EAAAL,EAAUE,YAAYC,sBAAc,IAAAE,OAAA,EAAAA,EAAED,OAAuBb,EAAQrB,kBAEtEqB,EAAQnB,UAAW,GAIrB0B,EAASZ,YAAYI,kBAChBE,EAAAA,SAAA,CAAA,EAAAD,GAEN,EACD"}
+{"version":3,"file":"currentCall.js","sources":["../../src/models/currentCall.ts"],"sourcesContent":["// Copyright (C) 2022 Nethesis S.r.l.\n// SPDX-License-Identifier: AGPL-3.0-or-later\n\nimport { createModel } from '@rematch/core'\nimport type { RootModel } from '.'\nimport incomingRingtone from '../static/incoming_ringtone'\nimport { dispatchOutgoingCallStarted } from '../events/index'\n\nconst defaultState = {\n  displayName: '',\n  username: '',\n  number: '',\n  incomingSocket: false,\n  incomingWebRTC: false,\n  incoming: false,\n  acceptedSocket: false,\n  acceptedWebRTC: false,\n  accepted: false,\n  outgoingSocket: false,\n  outgoingWebRTC: false,\n  outgoing: false,\n  startTime: '',\n  muted: false,\n  paused: false,\n  keypadValue: '',\n  conversationId: '',\n  transferring: false,\n  transferringName: '',\n  transferringNumber: '',\n  transferringStartTime: '',\n  transferSwitching: false,\n  transferCalls: new Array(),\n}\n\nexport const currentCall = createModel<RootModel>()({\n  state: defaultState,\n  reducers: {\n    updateCurrentCall: (state, payload: CurrentCallTypes) => {\n      return {\n        ...state,\n        ...payload,\n        transferCalls: state.transferCalls,\n      }\n    },\n    updateKeypadValue: (state, payload: string) => {\n      return {\n        ...state,\n        keypadValue: payload,\n      }\n    },\n    updateTransferring: (state, payload: boolean) => {\n      return {\n        ...state,\n        transferring: payload,\n      }\n    },\n    updateTransferSwitching: (state, payload: boolean) => {\n      return {\n        ...state,\n        transferSwitching: payload,\n      }\n    },\n    addTransferCalls: (state, payload: TransferCallsTypes) => {\n      if (state.transferCalls.find((item) => item.number === payload.number)) {\n        return state\n      } else {\n        return {\n          ...state,\n          transferCalls: [...state.transferCalls, payload],\n        }\n      }\n    },\n    reset: () => {\n      return defaultState\n    },\n  },\n  effects: (dispatch) => ({\n    checkIncomingUpdatePlay: (payload: CurrentCallTypes, rootState) => {\n      // Check call type and incoming confirmation source\n      if (\n        (rootState.currentUser.default_device?.type === 'webrtc' && payload.incomingWebRTC) ||\n        (rootState.currentUser.default_device?.type === 'physical' && payload.incomingSocket)\n      ) {\n        payload.incoming = true\n\n        // Update local player and play the audio\n        dispatch.player.updateStartAudioPlayer({ src: incomingRingtone, loop: true })\n      }\n      // Update the current call values and set incoming\n      dispatch.currentCall.updateCurrentCall({\n        ...payload,\n      })\n    },\n    checkOutgoingUpdate: (payload: CurrentCallTypes, rootState) => {\n      // Check call type and outgoing confirmation source\n      if (\n        (rootState.currentUser.default_device?.type === 'webrtc' && payload.outgoingWebRTC) ||\n        (rootState.currentUser.default_device?.type === 'physical' && payload.outgoingSocket)\n      ) {\n        payload.outgoing = true\n        // Dispatch an event for outgoing call\n        dispatchOutgoingCallStarted(payload.displayName, payload.number)\n      }\n      // Update the current call values and set outgoing\n      dispatch.currentCall.updateCurrentCall({\n        ...payload,\n      })\n    },\n    checkAcceptedUpdate: (payload: CurrentCallTypes, rootState) => {\n      // Check call type and accepted confirmation source\n      if (\n        (rootState.currentUser.default_device?.type === 'webrtc' && payload.acceptedWebRTC) ||\n        (rootState.currentUser.default_device?.type === 'physical' && payload.acceptedSocket)\n      ) {\n        payload.accepted = true\n        // TODO - dispatch accepted event\n      }\n      // Update the current call values\n      dispatch.currentCall.updateCurrentCall({\n        ...payload,\n      })\n    },\n  }),\n})\n\nexport type TransferCallsTypes = {\n  type: 'transferred' | 'destination'\n  displayName: string\n  number: string\n  startTime: string\n}\n\nexport interface CurrentCallTypes {\n  displayName?: string\n  username?: string\n  number?: string\n  incomingSocket?: boolean\n  incomingWebRTC?: boolean\n  incoming?: boolean\n  acceptedSocket?: boolean\n  acceptedWebRTC?: boolean\n  accepted?: boolean\n  outgoingSocket?: boolean\n  outgoingWebRTC?: boolean\n  outgoing?: boolean\n  startTime?: string\n  muted?: boolean\n  paused?: boolean\n  conversationId?: string\n  transferring?: boolean\n  transferringName?: string\n  transferringNumber?: string\n  transferringStartTime?: string\n  transferSwitching?: boolean\n  transferCalls?: TransferCallsTypes[]\n}\n"],"names":["defaultState","displayName","username","number","incomingSocket","incomingWebRTC","incoming","acceptedSocket","acceptedWebRTC","accepted","outgoingSocket","outgoingWebRTC","outgoing","startTime","muted","paused","keypadValue","conversationId","transferring","transferringName","transferringNumber","transferringStartTime","transferSwitching","transferCalls","Array","currentCall","createModel","state","reducers","updateCurrentCall","payload","__assign","updateKeypadValue","updateTransferring","updateTransferSwitching","addTransferCalls","find","item","__spreadArray","reset","effects","dispatch","checkIncomingUpdatePlay","rootState","_a","currentUser","default_device","type","_b","player","updateStartAudioPlayer","src","incomingRingtone","loop","checkOutgoingUpdate","dispatchOutgoingCallStarted","checkAcceptedUpdate"],"mappings":"sWAQA,IAAMA,EAAe,CACnBC,YAAa,GACbC,SAAU,GACVC,OAAQ,GACRC,gBAAgB,EAChBC,gBAAgB,EAChBC,UAAU,EACVC,gBAAgB,EAChBC,gBAAgB,EAChBC,UAAU,EACVC,gBAAgB,EAChBC,gBAAgB,EAChBC,UAAU,EACVC,UAAW,GACXC,OAAO,EACPC,QAAQ,EACRC,YAAa,GACbC,eAAgB,GAChBC,cAAc,EACdC,iBAAkB,GAClBC,mBAAoB,GACpBC,sBAAuB,GACvBC,mBAAmB,EACnBC,cAAe,IAAIC,OAGRC,EAAcC,EAAWA,aAAXA,CAAyB,CAClDC,MAAO3B,EACP4B,SAAU,CACRC,kBAAmB,SAACF,EAAOG,GACzB,OACKC,WAAAA,EAAAA,SAAAA,EAAAA,SAAA,CAAA,EAAAJ,GACAG,GAAO,CACVP,cAAeI,EAAMJ,eAExB,EACDS,kBAAmB,SAACL,EAAOG,GACzB,OAAAC,EAAAA,SAAAA,EAAAA,SAAA,GACKJ,GAAK,CACRX,YAAac,GAEhB,EACDG,mBAAoB,SAACN,EAAOG,GAC1B,OAAAC,EAAAA,SAAAA,EAAAA,SAAA,GACKJ,GAAK,CACRT,aAAcY,GAEjB,EACDI,wBAAyB,SAACP,EAAOG,GAC/B,OAAAC,EAAAA,SAAAA,EAAAA,SAAA,GACKJ,GAAK,CACRL,kBAAmBQ,GAEtB,EACDK,iBAAkB,SAACR,EAAOG,GACxB,OAAIH,EAAMJ,cAAca,MAAK,SAACC,GAAS,OAAAA,EAAKlC,SAAW2B,EAAQ3B,MAAM,IAC5DwB,EAGFI,EAAAA,SAAAA,EAAAA,SAAA,CAAA,EAAAJ,IACHJ,cAAae,EAAAA,cAAAA,EAAAA,cAAA,GAAMX,EAAMJ,eAAe,GAAA,CAAAO,IACzC,IAEJ,EACDS,MAAO,WACL,OAAOvC,CACR,GAEHwC,QAAS,SAACC,GAAa,MAAC,CACtBC,wBAAyB,SAACZ,EAA2Ba,YAGD,YAAT,QAAtCC,EAAAD,EAAUE,YAAYC,sBAAgB,IAAAF,OAAA,EAAAA,EAAAG,OAAqBjB,EAAQzB,gBACpB,cAAX,QAApC2C,EAAAL,EAAUE,YAAYC,sBAAc,IAAAE,OAAA,EAAAA,EAAED,OAAuBjB,EAAQ1B,kBAEtE0B,EAAQxB,UAAW,EAGnBmC,EAASQ,OAAOC,uBAAuB,CAAEC,IAAKC,EAAAA,QAAkBC,MAAM,KAGxEZ,EAAShB,YAAYI,kBAChBE,EAAAA,SAAA,CAAA,EAAAD,GAEN,EACDwB,oBAAqB,SAACxB,EAA2Ba,YAGG,YAAT,QAAtCC,EAAAD,EAAUE,YAAYC,sBAAgB,IAAAF,OAAA,EAAAA,EAAAG,OAAqBjB,EAAQnB,gBACpB,cAAX,QAApCqC,EAAAL,EAAUE,YAAYC,sBAAc,IAAAE,OAAA,EAAAA,EAAED,OAAuBjB,EAAQpB,kBAEtEoB,EAAQlB,UAAW,EAEnB2C,EAAAA,4BAA4BzB,EAAQ7B,YAAa6B,EAAQ3B,SAG3DsC,EAAShB,YAAYI,kBAChBE,EAAAA,SAAA,CAAA,EAAAD,GAEN,EACD0B,oBAAqB,SAAC1B,EAA2Ba,YAGG,YAAT,QAAtCC,EAAAD,EAAUE,YAAYC,sBAAgB,IAAAF,OAAA,EAAAA,EAAAG,OAAqBjB,EAAQtB,gBACpB,cAAX,QAApCwC,EAAAL,EAAUE,YAAYC,sBAAc,IAAAE,OAAA,EAAAA,EAAED,OAAuBjB,EAAQvB,kBAEtEuB,EAAQrB,UAAW,GAIrBgC,EAAShB,YAAYI,kBAChBE,EAAAA,SAAA,CAAA,EAAAD,GAEN,EACD"}

package/dist/models/motions.js

@@ -1,2 +1,2 @@
-"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("../node_modules/@rematch/core/dist/core.esm.js").createModel()({state:{padding_expanded:24,padding_x_collapsed:8,padding_y_collapsed:16,border_radius_expanded:20,border_radius_collapsed:99,variants:{call:{expanded:{incoming:{width:418,height:96},outgoing:{width:418,height:96},accepted:{width:348,height:236},transfer:{width:348,height:304}},collapsed:{width:168,height:40}},keypad:{expanded:{width:338,height:462},collapsed:{width:168,height:40}},transfer:{expanded:{width:408,height:472},collapsed:{width:168,height:40}},player:{expanded:{width:374,height:236},collapsed:{width:168,height:40}},alerts:{width:418,height:88}}}});exports.motions=e;
+"use strict";Object.defineProperty(exports,"__esModule",{value:!0});var e=require("../node_modules/@rematch/core/dist/core.esm.js").createModel()({state:{padding_expanded:24,padding_x_collapsed:8,padding_y_collapsed:16,border_radius_expanded:20,border_radius_collapsed:99,variants:{call:{expanded:{incoming:{width:418,height:96},outgoing:{width:418,height:96},accepted:{width:348,height:236},transfer:{width:348,height:304}},collapsed:{width:168,height:40}},keypad:{expanded:{width:338,height:400},collapsed:{width:168,height:40}},transfer:{expanded:{width:408,height:410},collapsed:{width:168,height:40}},player:{expanded:{width:374,height:236},collapsed:{width:168,height:40}},alerts:{width:418,height:88}}}});exports.motions=e;
 //# sourceMappingURL=motions.js.map

package/dist/models/motions.js.map

@@ -1 +1 @@
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package/dist/node_modules/@emotion/stylis/dist/stylis.browser.esm.js

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