282 lines
16 KiB
XML
282 lines
16 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 configuration 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>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>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 of a call, try to increase Media volume.</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"><a href="https://github.com/juha-h/baresip-studio">GitHub</a></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="authentication_username">Authentication Username</string>
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<string name="authentication_username_help">Authentication username if required by outbound proxy.</string>
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<string name="authentication_password">Authentication Password</string>
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<string name="authentication_password_help">Authentication password if required by outbound proxy.</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="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="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 Server</string>
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<string name="stun_server_help">A STUN Server of form host[:port].
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Factory default value is \'stun.l.google.com:19302\', pointing to public Google STUN server.
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Username and password are currently not supported.
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</string>
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<string name="stun_server_default" translatable="false">stun.l.google.com:19302</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="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="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="user_domain">user@domain</string>
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<string name="invalid_aor">Invalid user@domain \'%1$s\'</string>
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<string name="account_exists">Account \'%1$s\' already exists.</string>
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<string name="account_allocation_failure">"Failed to allocate new account.</string>
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<string name="encrypt_password">Encrypt Password</string>
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<string name="decrypt_password">Decrypt Password</string>
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<string name="export_accounts">Export</string>
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<string name="exported_accounts">Exported accounts to Download folder file \'accounts.bs\'.</string>
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<string name="export_accounts_error">"Failed to export accounts to Download folder.
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Check Apps → baresip → Permissions → Storage."
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</string>
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<string name="import_accounts">Import</string>
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<string name="imported_accounts">Imported accounts from Download folder. You need to restart
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baresip.</string>
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<string name="import_accounts_error">Failed to import accounts from Download folder file.
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Check Apps → baresip → Permissions → Storage and that file \'accounts.bs\' exists in
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the folder and, if so, you gave correct Decrypt Password."
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</string>
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<string name="delete_account">Do you want to delete account \'%1$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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<!-- 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$s from call history?
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</string>
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<!-- Chat Activity -->
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<string name="chat">Chat Messages</string>
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<string name="chat_with">Chat with %1$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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<!-- Config Activity -->
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<string name="configuration">Configuration</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="_0_0_0_0_5060" translatable="false">0.0.0.0:5060</string>
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<string name="prefer_ipv6">Prefer IPv6</string>
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<string name="prefer_ipv6_help">Prefer IPv6 if both IPv4 and IPv6 are available.</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="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="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="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="ice_lite_mode">ICE Lite Mode</string>
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<string name="ice_lite_mode_help">If checked, ICE Lite Mode is used.</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, configuration is reset
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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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<!-- 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="export_contacts">Export</string>
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<string name="import_contacts">Import</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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<string name="exported_contacts">Exported contacts to Download folder file \'contacts.bs\'.</string>
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<string name="export_error">"Failed to export contacts to Download folder.
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Check Apps → baresip → Permissions → Storage."</string>
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<string name="imported_contacts">Imported contacts from Download folder.</string>
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<string name="import_error">Failed to import contacts from Download folder. Check Apps →
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baresip → Permissions → Storage and that file \'contacts.bs\' exists in the folder.</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="dots" translatable="false">…</string>
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<!-- Main Activity -->
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<string name="about">About</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</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="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 invalid Listen Address
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or TLS file. They have been reset. Restart baresip.</string>
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<string name="registering_failed">Registering of \`%1$s\` failed.</string>
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<string name="no_microphone_permission">You have not granted microphone permission.</string>
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<string name="verify">Verify</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_query">Do you accept to transfer 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 closed</string>
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<string name="restart_needed">You need to restart baresip in order to activate the new config.</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?</string>
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<string name="unverify">Unverify</string>
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</resources>
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