330 lines
20 KiB
XML
330 lines
20 KiB
XML
<?xml version="1.0" encoding="utf-8"?>
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<resources>
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<string name="app_name" translatable="false">baresip</string>
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<!-- About Activity -->
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<string name="about_title">About baresip</string>
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<string name="about_text">
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<![CDATA[
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<h1>Baresip library based SIP User Agent</h1>
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<p>Juha Heinanen <jh@tutpro.com></p>
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<p>Version %1$s</p>
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<h2>Usage Hints</h2>
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<ul>
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<li>Check that Settings default values meet your needs (touch item titles for help).</li>
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<li>Then create one or more accounts (again touch item titles for help).</li>
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<li>Registration status of an account is shown with a colored dot: green (registration
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succeeded, yellow (registration is in progress), red (registration failed), white (registration
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has not been activated).</li>
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<li>Peers of calls and messages can be added to contacts by long touches.</li>
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<li>Long touches can also be used to remove calls, chats, messages, and contacts.</li>
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<li>Touch/long touch of contact icon can be used to install/remove image avatar.</li>
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<li>You can re-reselect the previous call party by touching the call icon when Callee is empty.</li>
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<li>If you cannot hear the other party during call, try to increase Media volume of
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the device or set Default Call Volume in Settings.</li>
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</ul>
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<h2>Known Issues</h2>
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<ul>
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<li>Due to limitations in underlying libraries, baresip does not currently support multiple,
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concurrently active network interfaces.</li>
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</ul>
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<h2>Source code</h2>
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Source code is available at <a href="https://github.com/juha-h/baresip-studio">GitHub</a>,
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where also issues can be reported.
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]]>
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</string>
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<string name="baresip_link" translatable="false">
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<a href="https://github.com/juha-h/baresip-studio">GitHub</a>
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</string>
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<!-- Account Activity -->
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<string name="account">Account</string>
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<string name="display_name">Display Name</string>
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<string name="your_name">Your Name</string>
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<string name="display_name_help">Name (if any) used in From URI of outbound requests.</string>
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<string name="invalid_display_name">Invalid Display Name \'%1$s\'</string>
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<string name="authentication_username">Authentication Username</string>
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<string name="authentication_username_help">Authentication username
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if authentication of SIP requests is required. Default value is account\'s username.
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</string>
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<string name="invalid_authentication_username">Invalid Authentication Username \'%1$s\'</string>
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<string name="authentication_password">Authentication Password</string>
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<string name="authentication_password_help">Authentication
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Password up to 64 characters. If Authentication Username is given, but Password is not
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given, it will be asked when baresip is started.
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</string>
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<string name="invalid_authentication_password">Invalid Authentication Password \'%1$s\'</string>
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<string name="outbound_proxies">Outbound Proxies</string>
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<string name="outbound_proxies_help">SIP URI of one or two proxies that must be used when sending requests.
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If two is given, REGISTER requests are sent to both and other requests are sent to
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one that responds. If no outbound proxy is given, requests are sent based on
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DNS NAPTR/SRV/A record lookup of callee URI hostpart. If hostpart of SIP URI is an IPv6
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address, the address must be written inside brackets [].
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\nExamples:
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\n • sip:foo.com:5060;transport=tls
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\n • sip:[2001:67c:223:777::10]:5060;transport=tcp
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</string>
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<string name="sip_uri_of_proxy_server">SIP URI of Proxy Server</string>
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<string name="sip_uri_of_another_proxy_server">SIP URI of another Proxy Server</string>
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<string name="invalid_proxy_server_uri">Invalid Proxy Server URI \'%1$s\'</string>
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<string name="register">Register</string>
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<string name="register_help">If checked, registration is enabled and REGISTER requests are sent at
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12 minute intervals.
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</string>
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<string name="audio_codecs">Audio Codecs</string>
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<string name="audio_codecs_help">List of supported audio codecs in priority order</string>
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<string name="video_codecs">Video Codecs</string>
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<string name="video_codecs_help">List of supported video codecs in priority order</string>
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<string name="media_nat">Media NAT Traversal</string>
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<string name="media_nat_help">Selects media NAT traversal protocol (if any). Possible choices are STUN
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(Session Traversal Utilities for NAT, RFC 5389) and ICE (Interactive Connectivity
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Establishment, RFC 5245).
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</string>
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<string name="stun_server">STUN/TURN Server</string>
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<string name="stun_server_help">A STUN/TURN Server URI of form scheme:host[:port], where scheme
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is \'stun\', \'stuns\', \'turn\', or \'turns\'. Factory default STUN server is
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\'stun:stun.l.google.com:19302\', pointing to public Google STUN server. There is no
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factory default TURN server.
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</string>
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<string name="invalid_stun_server">Invalid STUN/TURN Server URI \'%1$s\'</string>
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<string name="stun_server_default" translatable="false">stun:stun.l.google.com:19302</string>
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<string name="stun_username">STUN/TURN Username</string>
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<string name="stun_username_help">Username if required by STUN/TURN server</string>
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<string name="invalid_stun_username">Invalid Username \'%1$s\'</string>
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<string name="stun_password">STUN/TURN Password</string>
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<string name="stun_password_help">Password if required by STUN/TURN server</string>
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<string name="invalid_stun_password">Invalid Password \'%1$s\'</string>
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<string name="media_encryption">Media Encryption</string>
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<string name="media_encryption_help">Selects media transport encryption protocol (if any).
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\n • ZRTP (recommended) means that ZRTP end-to-end media encryption negotiation is tried after
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the call has been established.
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\n • DTLS-SRTPF means that UDP/TLS/RTP/SAVPF is offered in outgoing call and that RTP/SAVP,
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RTP/SAVPF, UDP/TLS/RTP/SAVP, or UDP/TLS/RTP/SAVPF is used if offered in incoming call.
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\n • SRTP-MANDF means that RTP/SAVPF is offered in outgoing call and required in incoming call.
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\n • SRTP-MAND means that RTP/SAVP is offered in outgoing call and required in incoming call.
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\n • SRTP means that RTP/AVP is offered in outgoing call and that RTP/SAVP or RTP/SAVPF is used
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if offered in incoming call.
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</string>
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<string name="prefer_ipv6_media">Prefer IPv6 Media</string>
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<string name="prefer_ipv6_media_help">If checked, offer to use IPv6 media protocol (if available) when media protocol of peer cannot be automatically determined.</string>
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<string name="answer_mode">Answer Mode</string>
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<string name="answer_mode_help">Selects how incoming calls are answered.</string>
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<string name="manual">Manual</string>
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<string name="auto">Automatic</string>
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<string name="voicemail_uri">Voicemail URI</string>
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<string name="voicemain_uri_help">SIP URI for checking of voicemail messages. If left empty, voicemail
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messages (Message Waiting Indications) are not subscribed to.
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</string>
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<string name="invalid_voicemail_uri">Invalid Voicemail URI \'%1$s\'</string>
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<string name="default_account">Default Account</string>
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<string name="default_account_help">If checked, this account is selected when baresip is started.
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</string>
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<!-- Accounts Activity -->
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<string name="accounts">Accounts</string>
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<string name="new_account">New Account</string>
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<string name="accounts_help">Account\'s port number and transport protocol may be optionally given when a new account is created: username@domain[:port][;transport=udp|tcp|tls]. If port is given and transport protocol is not given, transport protocol defaults to udp. If port is not given and transport protocol is given, port defaults to 5060 or 5061 (TLS). If neither is given and no outbound proxy is specified, account\'s registrar (if any) is determined solely based on domain\'s DNS information.</string>
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<string name="user_domain">user@domain</string>
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<string name="invalid_aor">Invalid user@domain[:port][;transport=udp|tcp|tls] \'%1$s\'</string>
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<string name="account_exists">Account \'%1$s\' already exists.</string>
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<string name="account_allocation_failure">"Failed to allocate new account.</string>
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<string name="encrypt_password">Encrypt Password</string>
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<string name="decrypt_password">Decrypt Password</string>
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<string name="delete_account">Do you want to delete account \'%1$s\'?</string>
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<!-- Baresip Service -->
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<string name="answer">Answer</string>
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<string name="reject">Reject</string>
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<string name="incoming_call_from">Call from</string>
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<string name="transfer_request">Call transfer request to</string>
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<string name="message_from">Message from</string>
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<!-- Boot Completed Receiver -->
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<string name="tap_to_start">Tap to start application</string>
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<!-- Calls Activity -->
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<string name="call_history">Call History</string>
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<string name="call">Call</string>
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<string name="calls_calls">calls</string>
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<string name="calls_call">call</string>
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<string name="calls_call_message_question">Do you want to call or send message to \'%1$s\'?</string>
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<string name="calls_add_delete_question">Do you want to add \'%1$s\' to contacts or delete
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%2$s from call history?
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</string>
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<string name="calls_delete_question">Do you want to delete \'%1$s\' %2$s from call history?
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</string>
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<string name="delete_history">Delete</string>
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<string name="disable_history">Disable</string>
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<string name="enable_history">Enable</string>
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<string name="delete_history_alert">Do you want to delete call history of account \'%1$s\'?</string>
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<!-- Chat Activity -->
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<string name="chat">Chat Messages</string>
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<string name="chat_with">Chat with %1$s</string>
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<string name="new_message">New message</string>
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<string name="long_message_question">Do you want to delete message or add peer \'%1$s\' to contacts?</string>
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<string name="short_message_question">Do you want to delete message?</string>
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<string name="add_contact">Add Contact</string>
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<string name="sending_failed">Sending of message failed</string>
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<string name="message_failed">Failed</string>
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<!-- Chats Activity -->
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<string name="chats">Chat History</string>
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<string name="today">Today</string>
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<string name="you">You</string>
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<string name="new_chat_peer">New Chat Peer</string>
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<string name="invalid_chat_peer_uri">Invalid SIP URI</string>
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<string name="long_chat_question">Do you want to delete chat with peer \'%1$s\' or
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add peer to contacts?</string>
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<string name="short_chat_question">Do you want to delete chat with \'%1$s\'?</string>
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<string name="delete_chats">Delete</string>
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<string name="delete_chats_alert">Do you want to delete chat history of account \'%1$s\'?</string>
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<!-- Config Activity -->
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<string name="configuration">Settings</string>
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<string name="start_automatically">Start Automatically</string>
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<string name="start_automatically_help">If checked, baresip starts automatically after device (re)start.</string>
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<string name="listen_address">Listen Address</string>
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<string name="listen_address_help">IP address and port of form \'address:port\' at which baresip listens
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for incoming SIP requests. If IP address is an IPv6 address, it must be written inside
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brackets []. IPv4 address 0.0.0.0 or IPv6 address [::] makes baresip listen at all
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available addresses. If left empty (factory default), baresip listens at port 5060 of
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all available addresses.
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</string>
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<string name="invalid_listen_address">Invalid Listen Address</string>
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<string name="_0_0_0_0_5060" translatable="false">0.0.0.0:5060</string>
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<string name="dns_servers">DNS Servers</string>
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<string name="dns_servers_help">Comma separated list of addresses of DNS servers. If not given,
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DNS server addresses are obtained dynamically from the system. Each DNS address is of form
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\'ip:port\' or \'ip\'. If port is omitted, it defaults to 53. If ip is an IPv6 address and
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also port is given, ip must
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be written inside brackets []. As an example, list \'8.8.8.8:53,[2001:4860:4860::8888]:53\'
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points to IPv4 and IPv6 addresses of public Google DNS servers.</string>
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<string name="invalid_dns_servers">Invalid DNS Servers</string>
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<string name="failed_to_set_dns_servers">Failed to set DNS servers</string>
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<string name="tls_certificate_file">TLS Certificate File</string>
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<string name="tls_certificate_file_help">If checked, file \'cert.pem\' containing
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TLS certificate and private key of this baresip instance has been or will be loaded from
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Download directory. For security reasons, delete the file after loading.</string>
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<string name="tls_ca_file">TLS CA File</string>
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<string name="tls_ca_file_help">If checked, file \'ca_certs.crt\' containing TLS certificates
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of Certificate Authorities has been or will be loaded from Download directory.</string>
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<string name="audio">Audio</string>
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<string name="audio_modules_title">Audio Modules</string>
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<string name="audio_modules_help">Audio codecs provided by the checked modules are
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available for use by the accounts.</string>
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<string name="failed_to_load_module">Failed to load module.</string>
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<string name="aec">Acoustic Echo Cancellation</string>
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<string name="aec_help">If checked, echo cancellation is attempted on call audio.</string>
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<string name="aec_extented_filter">AEC Extented Filter</string>
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<string name="aec_extented_filter_help">If checked, echo cancellation is using extended filter.</string>
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<string name="opus_bit_rate">Opus Bit Rate</string>
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<string name="opus_bit_rate_help">Average maximum bit rate used by Opus audio stream.
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Valid values are 6000-510000. Factory default is 28000.</string>
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<string name="_28000" translatable="false">28000</string>
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<string name="opus_packet_loss">Expected Opus packet-loss</string>
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<string name="opus_packet_loss_help">Expected Opus audio stream packet loss percentage,
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from 0–100. By default 0, turning off Opus Forward Error Correction (FEC).</string>
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<string name="_0" translatable="false">0</string>
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<string name="invalid_opus_bitrate">Invalid Opus bitrate</string>
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<string name="invalid_opus_packet_loss">Invalid Opus Packet Loss Percentage</string>
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<string name="default_call_volume">Default Call Volume</string>
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<string name="default_call_volume_help">If set, default call audio volume at scale 1–10.</string>
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<string name="debug">Debug</string>
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<string name="debug_help">Provides debug and info level log message availability in Logcat.</string>
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<string name="reset_config">Reset to Factory Defaults</string>
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<string name="reset_config_help">If checked, settings are reset to factory default values.</string>
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<string name="read_cert_error">Failed to read file \'cert.pem\' from Download directory.</string>
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<string name="read_ca_certs_error">Failed to read file \'ca_certs.crt\' from Download directory.</string>
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<string name="config_restart">You need to restart baresip in order to activate the new
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settings. Restart now?
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</string>
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<!-- Contact Activity -->
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<string name="contact">Contact</string>
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<string name="new_contact">New Contact</string>
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<string name="contact_name">Name</string>
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<string name="invalid_contact">Invalid contact name \'%1$s\'</string>
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<string name="contact_already_exists">Contact \'%1$s\' already exists.</string>
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<string name="invalid_contact_uri">Invalid SIP URI</string>
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<!-- Contacts Activity -->
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<string name="contacts">Contacts</string>
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<string name="contact_action_question">Do you want to call or send message to \'%1$s\'?</string>
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<string name="send_message">Send Message</string>
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<string name="contact_delete_question">Do you want to delete contact \'%1$s\'?</string>
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<string name="contacts_exceeded">Your maximum number of contacts %1%d has been exceeded.</string>
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<!-- Generic -->
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<string name="alert">Alert</string>
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<string name="info">Info</string>
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<string name="notice">Notice</string>
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<string name="cancel">Cancel</string>
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<string name="ok">OK</string>
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<string name="yes">Yes</string>
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<string name="no">No</string>
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<string name="accept">Accept</string>
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<string name="deny">Deny</string>
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<string name="user_id">User ID</string>
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<string name="password">Password</string>
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<string name="sip_uri" translatable="false">SIP URI</string>
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<string name="add">Add</string>
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<string name="delete">Delete</string>
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<string name="edit">Edit</string>
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<string name="send">Send</string>
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<string name="status">Status</string>
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<string name="error">Error</string>
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<string name="help">Help</string>
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<string name="confirmation">Confirmation</string>
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<string name="dots" translatable="false">…</string>
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<!-- Main Activity -->
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<string name="backup">Backup</string>
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<string name="restore">Restore</string>
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<string name="about">About</string>
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<string name="restart">Restart</string>
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<string name="quit">Quit</string>
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<string name="outgoing_call_to_dots">Outgoing call to …</string>
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<string name="incoming_call_from_dots">Incoming call from …</string>
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<string name="transferring_call_to_dots">Transferring call to …</string>
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<string name="invalid_sip_uri">Invalid SIP URI \'%1$s\'</string>
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<string name="callee">Callee</string>
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<string name="hangup">Hangup</string>
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<string name="hold">Hold</string>
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<string name="dtmf">DTMF</string>
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<string name="call_info">Call Info</string>
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<string name="call_info_not_available">No info available</string>
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<string name="duration">Duration</string>
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<string name="codecs">Codecs</string>
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<string name="rate">Rate</string>
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<string name="voicemail">Voicemail</string>
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<string name="voicemail_messages">Voicemail Messages</string>
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<string name="you_have">You have</string>
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<string name="one_new_message">one new message</string>
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<string name="new_messages">new messages</string>
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<string name="one_old_message">one old message</string>
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<string name="old_messages">old messages</string>
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<string name="and">and</string>
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<string name="no_messages">You have no messages</string>
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<string name="listen">Listen</string>
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<string name="messages">Messages</string>
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<string name="dialpad">Dialpad</string>
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<string name="call_already_active">You already have an active call.</string>
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<string name="start_failed">Baresip failed to start. This may be due to an invalid Settings value.
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Check Listen Address, TLS Certificate File, and TLS CA File. Then restart baresip.
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</string>
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<string name="registering_failed">Registering of \`%1$s\` failed.</string>
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<string name="verify">Verify Request</string>
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<string name="verify_sas">Do you want to verify SAS <%1$s> <%2$s>?</string>
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<string name="transfer">Transfer Request</string>
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<string name="transfer_query">Do you accept to transfer this call to \'%1$s\'?</string>
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<string name="call_failed">Call failed.</string>
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<string name="call_closed">Call is closed.</string>
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<string name="call_not_secure">This call is NOT secure!</string>
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<string name="peer_not_verified">This call is SECURE, but peer is NOT verified!</string>
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<string name="call_is_secure">This call is SECURE and peer is VERIFIED!
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Do you want to unverify the peer?
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</string>
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<string name="unverify">Unverify</string>
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<string name="backed_up">Application data backed up to Download folder file \'%1$s\'.</string>
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<string name="backup_failed">Failed to back up application data to Download folder file
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\'%1$s\'. Check Apps → baresip → Permissions → Storage.</string>
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<string name="restart_request">Restart Request</string>
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<string name="restored">Application data restored. baresip needs to be restarted.
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Restart now?
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</string>
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<string name="restore_failed">Failed to restore application data from Download folder. Check Apps →
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baresip → Permissions → Storage and that backup file \'%1$s\' exists in the folder
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and, if so, you gave correct Decrypt Password.
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</string>
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<string name="no_calls">You are not able to place or answer calls without Microphone permission.</string>
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<string name="show_password">Show Password</string>
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</resources>
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