Skip to content

Commit

Permalink
Merge Translations
Browse files Browse the repository at this point in the history
  • Loading branch information
exo-swf committed Oct 11, 2023
1 parent 9d88dab commit aac500e
Show file tree
Hide file tree
Showing 9 changed files with 177 additions and 0 deletions.
Original file line number Diff line number Diff line change
@@ -0,0 +1 @@
webconferencing.admin.page=Konferensi web
Original file line number Diff line number Diff line change
@@ -0,0 +1 @@
analytics.navigation.webconferencing=Web conferencing
Original file line number Diff line number Diff line change
@@ -0,0 +1,49 @@
<?xml version="1.0" encoding="UTF-8"?>
<bundle>
<webconferencing>
<admin>
<error>
<ProviderNotRegistered>The editor provider is not registered.</ProviderNotRegistered>
<EmptyRequest>The request should contain at least 1 form param (active/permissions).</EmptyRequest>
<CannotGetProviders>Can't get providers list.</CannotGetProviders>
<PermissionNotValid>The specified permission is not valid. Maybe the user or group doesn't exist.</PermissionNotValid>
<CannotLoadIdentities>Can't load identities by specified criteria</CannotLoadIdentities>
<CannotSavePreferredEditor>Can't save selected editor as preferred one</CannotSavePreferredEditor>
<UnableGetData>An error occurred while reading the data.</UnableGetData>
<UnablePostData>An error occurred while saving the data.</UnablePostData>
</error>
<page>Konferensi web</page>
<title>Administrasi Konferensi Web</title>
<info>Konferensi Web dapat ditangani oleh penyedia yang berbeda. Anda dapat mengaktifkan penyedia yang Anda butuhkan dan mengkonfigurasi pengaturan.</info>
<table>
<Provider>Provider</Provider>
<Description>Description</Description>
<Active>Active</Active>
<Permissions>Actions</Permissions>
</table>
<WebRTC>
<name>WebRTC</name>
<description>Default provider based on <![CDATA[<a href="https://webrtc.org/" target="_blank">WebRTC</a>]]> standard. Let users to place peer to peer call. No browser plugin needed.</description>
</WebRTC>
<Skype>
<name>Skype</name>
<description>Provider for Skype. Let users place a group and 1:1 calls. No plugin required but the Skype application must be installed on the client side.</description>
</Skype>
<Jitsi>
<name>Jitsi</name>
<description>Jitsi provider for WebConferencing</description>
</Jitsi>
<modal>
<title>Provider settings</title>
<SearchLabel>URL:</SearchLabel>
<WithPermissions>STUN / TURN server</WithPermissions>
<None>None</None>
<Everybody>Everybody</Everybody>
</modal>
<buttons>
<Save>Save</Save>
<Cancel>Cancel</Cancel>
</buttons>
</admin>
</webconferencing>
</bundle>
Original file line number Diff line number Diff line change
@@ -0,0 +1,11 @@
<?xml version="1.0" encoding="UTF-8"?>
<bundle>
<!-- Messages and UI texts for client parts -->
<webconferencing>
<ok>Oke</ok>
<cancel>Membatalkan</cancel>
<callHeader>Start Call</callHeader>
<notConfigured>tidak di konfigurasi</notConfigured>
<errorReference>Referensi kesalahan:</errorReference>
</webconferencing>
</bundle>
Original file line number Diff line number Diff line change
@@ -0,0 +1,15 @@
<?xml version="1.0" encoding="UTF-8"?>
<bundle>
<!-- Messages for Platform menu and view settings -->
<UIBasicProfile>
<label>
<myconnector>Panggil Saya</myconnector>
</label>
</UIBasicProfile>
<UIEditUserProfileForm>
<label>
<myconnector>Panggil Saya</myconnector>
</label>
</UIEditUserProfileForm>
<!-- Messages for Platform add-on -->
</bundle>
Original file line number Diff line number Diff line change
@@ -0,0 +1,7 @@
analytics.web-conferencing=Web conferencing
analytics.totalCalls=Total Calls
analytics.recordedCall=Recorded Calls
analytics.startedCallRepartition=Started Call Distribution
analytics.startedCallRepartitionByDay=Started Call Distribution By Day
analytics.recordUploadStatus=Recording Upload Status by Day
analytics.recordUploadSuccessRate=Recording Upload Success Rate
Original file line number Diff line number Diff line change
@@ -0,0 +1,15 @@
UINotification.label.group.webconferencing=Web conferencing
UINotification.label.CallRecordingPlugin=When a call recording is available

#setting
UINotification.title.CallRecordingPlugin=When a call recording is available

#web
Notification.webconferencing.callrecording.success=Your web conference has been recorded successfully
Notification.webconferencing.callrecording.failed=Your web conference recording has failed

#mail
Notification.title.CallRecordingPlugin=Call Recording
Notification.subject.CallRecordingPlugin=Call Recording
Notification.label.openFile=Open File
Notification.label.SayHello=Hai
Original file line number Diff line number Diff line change
@@ -0,0 +1,32 @@
<?xml version="1.0" encoding="UTF-8"?>
<bundle>
<webrtc>
<admin>
<description><![CDATA[Default provider based on <a href="https://webrtc.org/" target="_blank">WebRTC</a> standard. Let users to place peer to peer call. No browser plugin needed.]]></description>
<title>Pengaturan WebRTC</title>
<servers>SETRUM / MENGUBAH server</servers>
<serversTip>SETRUM/MENGHIDUPKAN server di haruskan untuk memastikan bahwa panggilan web conferencing akan bekerja dengan baik di seluruh router jaringan. Kami sangat menyarankan untuk menyiapkan server MEBGHIDUPKAN pribadi Anda sendiri untuk tetap mengontrol keamanan komunikasi Anda dan memastikan ketersediaan maksimal.</serversTip>
<url>URL</url>
<serverUrl>URL Server...</serverUrl>
<credentials>Kredensial</credentials>
<username>Nama Pemilik</username>
<credential>Kredensial</credential>
<save>Menyimpan</save>
<cancel>Membatalkan</cancel>

<noServer>No server available, please add your first one</noServer>
<addNewServer>Add a new server</addNewServer>
<removeServer>Remove server</removeServer>

<!-- Confirmation dialog -->
<confirmServerRemoval>Hapus ES Server?</confirmServerRemoval>
<serverRemoveText>Apakah anda yakin ingin menghapus konfigurasi server ini?</serverRemoveText>
<remove>Hapus</remove>

<!-- Error Diagnostics -->
<errorDiagnostic>Kesalahan diagnosis</errorDiagnostic>
<enableLogCollection>Aktifkan sisi klien log koleksi</enableLogCollection>
<logCollectionInfo>Bendera ini akan mencatat informasi dari klien dilog server untuk memecahkan masalah konektivitas WebRTC. Gunakan dengan hati-hati, mode ini bersifat verbose dan dapat menyebabkan overhead jaringan yang signifikan.</logCollectionInfo>
</admin>
</webrtc>
</bundle>
Original file line number Diff line number Diff line change
@@ -0,0 +1,46 @@
<?xml version="1.0" encoding="UTF-8"?>
<bundle>
<webrtc>
<call>Panggilan</call>
<callingYou>menelepon anda...</callingYou>
<callWith>Panggilan dengan</callWith>
<callTo>Panggilan untuk</callTo>
<incomingCall>Panggilan masuk</incomingCall>
<answer>Jawab</answer>
<decline>Menurun</decline>
<callStartTip>Tekan untuk memulai obrolan video</callStartTip>
<callRunningTip>Panggilan ini sudah berlangsung. Tekan untuk menampilkan jendela panggilan.</callRunningTip>

<providerError>Kesalahan pada penyedia WebRTC</providerError>
<providerNotAvailable>Penyedia WebRTC tidak ada</providerNotAvailable>
<refreshTryAgainContactAdmin>Silahkan refresh halaman ini dan coba lagi. Jika terdapat kesalahan yang sama, mohon hubungi administrator anda dengan mereferensikan kesalahan ini.</refreshTryAgainContactAdmin>
<callStopped>Panggilan telah dihentikan</callStopped>
<errorAddCall>Panggilan tidak dapat di mulai karena kesalahan pada server</errorAddCall>
<errorOpeningCall>Terjadi kesalahan saat membuka jendela panggilan. Harap segarkan halaman dan coba lagi. Jika kesalahan berlanjut, hubungi administrator Anda.</errorOpeningCall>
<callWindowNotOpen>Jendela panggilan tidak bisa di buka. Harap periksa apakah itu diblokir oleh browser Anda.</callWindowNotOpen>
<errorStartingCall>Gagal mulai panggilan</errorStartingCall>
<errorIncomingCall>Kesalahan saat panggilan masuk</errorIncomingCall>
<errorReadUserStatus>Kesalahan saat membaca informasi status pengguna dari server</errorReadUserStatus>
<errorReadCall>Kesalahan saat membaca informasi status panggilan dari server</errorReadCall>
<errorSubscribeUser>Gagal membuat saluran server-ke-browser untuk mengkomunikasikan pembaharuan pengguna</errorSubscribeUser>
<errorSubscribeCall>Gagal membuat saluran server-ke-browser untuk pembaharuan panggilan</errorSubscribeCall>
<errorStartingConnection>Kesalahan saat mulai koneksi</errorStartingConnection>
<notSupportedPlatform>Platform yang tidak di dukung</notSupportedPlatform>
<yourBrowserNotSupportWebrtc>Browser Anda tidak bisa mendukung panggilan WebRTC</yourBrowserNotSupportWebrtc>
<notInitialized>Konektor panggilan video WebRTC tidak diinisialisasi dengan benar. Silakan tutup jendela ini dan coba lagi.</notInitialized>
<connectionFailed>Gagal konek</connectionFailed>
<audioVideoRequired>Mikrofon dan/atau kamera di haruskan melakukan panggilan</audioVideoRequired>
<mediaDevicesError>Kesalahan pada saat mengakses kamera atau mikrofon</mediaDevicesError>
<accessDenied>Tidak dapat membuka perangkat media karena adanya kesalahan akses atau izin. Pastikan perangkat Anda terhubung dan tidak digunakan oleh aplikasi lain.</accessDenied>
<noAudioFound>Tidak ada mikrofon yang di temukan</noAudioFound>
<butVideoFound>tapi ada kamera yang di temukan</butVideoFound>

<admin>
<wrongSettings>Kesalahan pada pengaturan</wrongSettings>
<serverUrlMandatory>URL server tidak boleh di kosongkan</serverUrlMandatory>
<usernameMandatory>Nama pengguna tidak boleh kosong</usernameMandatory>
<credentialMandatory>Kata sandi tidak boleh kosong</credentialMandatory>
<errorSavingSettings>Kesalahan menyimpan pengaturan</errorSavingSettings>
</admin>
</webrtc>
</bundle>

0 comments on commit aac500e

Please sign in to comment.