<?xml version="1.0" encoding="utf-8"?>
<resources>
    <string name="about_text">
        <![CDATA[
        <h1>Baresip library based SIP User Agent</h1>
        <p>Juha Heinanen &lt;jh@tutpro.com&gt;</p>
        <p>Version %1$s</p>
        <h2>Usage Hints</h2>
        <ul>
            <li>Check that default values in Settings meet your needs (touch item titles for help).</li>
            <li>Then in Accounts, create one or more accounts (again touch item titles for help).</li>
            <li>Registration status of an account is shown with a colored dot: green (registration
            succeeded, yellow (registration is in progress), red (registration failed), white (registration
            has not been activated).</li>
            <li>Touch on the dot leads directly to account configuration.</li>
            <li>Swipe down gesture causes re-registration of the currently shown account.</li>
            <li>Peers of calls and messages can be added to contacts by long touches.</li>
            <li>Long touches can also be used to remove calls, chats, messages, and contacts.</li>
            <li>Touch/long touch of contact icon can be used to install/remove image avatar.</li>
            <li>You can re-reselect the previous call party by touching the call icon when Callee is empty.</li>
        </ul>
        <h2>Known Issues</h2>
        <ul>
            <li>Due to limitations in underlying libraries, baresip does not currently support multiple,
            concurrently active network interfaces. Active network interface preference order is VPN,
            Internet, other.</li>
        </ul>
        <h2>Source code</h2>
        Source code is available at <a href="https://github.com/juha-h/baresip-studio">GitHub</a>,
        where also issues can be reported.
        <h2>Licenses</h2>
        <ul>
            <li><b>BSD-3-Clause</b> except the following:</li>
            <li><b>Apache 2.0</b> AMR codec and TLS security</li>
            <li><b>LGPL 2.1</b> G.722 and G.726 codecs</li>
            <li><b>AGPLv4</b> ZRTP media encryption</li>
            <li><b>GNU GPLv3</b> G.729 codec</li>
        </ul>
        ]]>
    </string>
    <string name="about_text_plus">
        <![CDATA[
        <h1>Baresip library based SIP User Agent with video calls</h1>
        <p>Juha Heinanen &lt;jh@tutpro.com&gt;</p>
        <p>Version %1$s</p>
        <h2>Usage Hints</h2>
        <ul>
            <li>Check that default values in Settings meet your needs (touch item titles for help).</li>
            <li>Then in Accounts, create one or more accounts (again touch item titles for help).</li>
            <li>Registration status of an account is shown with a colored dot:  green (registration
            succeeded, yellow (registration is in progress), red (registration failed), white (registration
            has not been activated).</li>
            <li>Touch on the dot leads directly to account configuration.</li>
            <li>Swipe down gesture causes re-registration of the currently shown account.</li>
            <li>Peers of calls and messages can be added to contacts by long touches.</li>
            <li>Long touches can also be used to remove calls, chats, messages, and contacts.</li>
            <li>Touch/long touch of contact icon can be used to install/remove image avatar.</li>
            <li>You can re-reselect the previous call party by touching the call icon when Callee is empty.</li>
        </ul>
        <h2>Known Issues</h2>
        <ul>
            <li>Due to limitations in underlying libraries, multiple concurrently active network
            interfaces are not supported. Active network interface preference order is VPN,
            Internet, other.</li>
            <li>In video calls, the device needs to be held in landscape
            mode rotated 90 degrees left from portrait orientation.</li>
            <li>Selfview is not properly shown when video stream is sendonly.</li>
        </ul>
        <h2>Source code</h2>
        Source code is available at <a href="https://github.com/juha-h/baresip-studio">GitHub</a>,
        where also issues can be reported.
        <h2>Licenses</h2>
        <ul>
            <li><b>BSD-3-Clause</b> except the following:</li>
            <li><b>Apache 2.0</b> AMR codec and TLS security</li>
            <li><b>LGPL 2.1</b> G.722 and G.726 codecs</li>
            <li><b>AGPLv4</b> ZRTP media encryption</li>
            <li><b>GNU GPLv2</b> H.264 codec</li>
            <li><b>GNU GPLv3</b> G.729 codec</li>
        </ul>
        ]]>
    </string>
    <string name="authentication_username_help">SIPリクエストの認証が必要な場合は、認証ユーザー名を入力して下さい。
        デフォルト値はアカウントのユーザー名です。
    </string>
    <string name="authentication_password_help">認証パスワードは64文字までです。
        もし、ユーザー名を入力したのにパスワードが入力されていない場合は、baresipの起動時に問い合わせます。
    </string>
    <string name="outbound_proxies_help">リクエストを送るときに、1つか2つSIP URIを使う必要がある。
        2つとも入力された場合、両方にREGISTERリクエストが送られ、他のリクエストは応答する方に送られます。
        outboundプロキシが与えられない場合、リクエストはcalllee URI hostpartのDNS NAPTR/SRV/Aレコード検索に基づいて送信される。
        SIP URIのhostpartがIPv6アドレスの場合、アドレスは括弧[]内に記載しなければなりません。
        \n記入例:
        \n • sip:example.com:5061;transport=tls
        \n • sip:[2001:67c:223:777::10];transport=tcp
        \n • sip:192.168.43.50:443;transport=wss
    </string>
    <string name="register_help">チェックを入れると、登録が有効になり、12 分間隔で REGISTER 要求が送信されます。</string>
    <string name="audio_codecs_help">対応しているオーディオコーデックの優先順位一覧</string>
    <string name="video_codecs_help">対応しているビデオコーデックの優先順位一覧</string>
    <string name="media_nat_help">必要であればメディアのNAT探索プロトコルを選択してください。
        選択肢としては、STUN（Session Traversal Utilities for NAT、RFC 5389）
        とICE（Interactive Connectivity Establishment、RFC 5245）があります。
    </string>
    <string name="stun_server_help">A STUN/TURN Server URI of form scheme:host[:port], where scheme
        is \'stun\' or \'turn\'. Factory default STUN Server for STUN and
        ICE protocols is \'stun:stun.l.google.com:19302\' pointing to public Google STUN server.
        There is no factory default TURN server.
    </string>
    <string name="media_encryption_help">Selects media transport encryption protocol (if any).
        \n • ZRTP (recommended) means that ZRTP end-to-end media encryption negotiation is tried after
        the call has been established.
        \n • DTLS-SRTPF means that UDP/TLS/RTP/SAVPF is offered in outgoing call and that RTP/SAVP,
        RTP/SAVPF, UDP/TLS/RTP/SAVP, or UDP/TLS/RTP/SAVPF is used if offered in incoming call.
        \n • SRTP-MANDF means that RTP/SAVPF is offered in outgoing call and required in incoming call.
        \n • SRTP-MAND means that RTP/SAVP is offered in outgoing call and required in incoming call.
        \n • SRTP means that RTP/AVP is offered in outgoing call and that RTP/SAVP or RTP/SAVPF is used
        if offered in incoming call.
    </string>
    <string name="voicemain_uri_help">SIP URI for checking of voicemail messages. If left empty, voicemail
        messages (Message Waiting Indications) are not subscribed to.
    </string>
    <string name="default_account_help">If checked, this account is selected when baresip is started.</string>
    <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>
    <string name="calls_add_delete_question">Do you want to add \'%1$s\' to contacts or delete %2$s from call history?
    </string>
    <string name="long_chat_question">Do you want to delete chat with peer \'%1$s\' or add peer to contacts?</string>
    <string name="listen_address_help">IP address and port of form \'address:port\' at which baresip listens
        for incoming SIP requests. If IP address is an IPv6 address, it must be written inside
        brackets []. IPv4 address 0.0.0.0 or IPv6 address [::] makes baresip listen at all
        available addresses. If left empty (factory default), baresip listens at port 5060 of
        all available addresses.
    </string>
    <string name="dns_servers_help">Comma separated list of addresses of DNS servers. If not given,
        DNS server addresses are obtained dynamically from the system. Each DNS address is of form
        \'ip:port\' or \'ip\'. If port is omitted, it defaults to 53. If ip is an IPv6 address and
        also port is given, ip must
        be written inside brackets []. As an example, list \'8.8.8.8:53,[2001:4860:4860::8888]:53\'
        points to IPv4 and IPv6 addresses of public Google DNS servers.
    </string>
    <string name="tls_certificate_file_help">If checked, file \'cert.pem\' containing TLS certificate and private key of this baresip instance has been or will be loaded from Download directory. For security reasons, delete the file after loading.</string>
    <string name="tls_ca_file_help">If checked, file \'ca_certs.crt\' containing TLS certificates of Certificate Authorities has been or will be loaded from Download directory.</string>
    <string name="audio_modules_help">Audio codecs provided by the checked modules are
        available for use by the accounts.
    </string>
    <string name="opus_bit_rate_help">Average maximum bit rate used by Opus audio stream.
        Valid values are 6000-510000. Factory default is 28000.
    </string>
    <string name="opus_packet_loss_help">Expected Opus audio stream packet loss percentage,
        from 0–100. By default 0, turning off Opus Forward Error Correction (FEC).
    </string>
    <string name="video_size_help">Size of transmitted video frames (width x height)
    </string>
    <string name="sip_trace_help">If checked and if Debug is checked, provides also SIP
        request and response trace to Logcat. Unchecked automatically at baresip start.
    </string>
    <string name="config_restart">You need to restart baresip in order to activate the new settings. Restart now?
    </string>
    <string name="start_failed">Baresip failed to start. This may be due to an invalid Settings value.
        Check Listen Address, TLS Certificate File, and TLS CA File. Then restart baresip.
    </string>
    <string name="call_is_secure">This call is SECURE and peer is VERIFIED! Do you want to unverify the peer?</string>
    <string name="backup_failed">Failed to back up application data to Download folder file
        \'%1$s\'. Check Apps → baresip → Permissions → Storage.
    </string>
    <string name="restore_failed">Failed to restore application data from Download folder. Check Apps → baresip → Permissions → Storage and that backup file \'%1$s\' exists in the folder and, if so, you gave correct Decrypt Password.</string>
    <string name="restored">Application data restored. baresip needs to be restarted. Restart now?</string>
    <string name="calls_delete_question"> %1$s  %2$s を通話履歴から削除してもよいですか?</string>
    <string name="long_message_question"> %1$s のメッセージと連絡先を削除してもよいですか?</string>
    <string name="invalid_authentication_username">無効な認証ユーザー名 %1$s です</string>
    <string name="transfer">転送</string>
    <string name="video_request">テレビ電話リクエスト</string>
    <string name="video_call">テレビ電話</string>
    <string name="hangup">切断</string>
    <string name="transferring_call_to_dots">転送中</string>
    <string name="incoming_call_from_dots">着信中</string>
    <string name="outgoing_call_to_dots">発信中</string>
    <string name="quit">終了</string>
    <string name="restart">再起動</string>
    <string name="about">詳細</string>
    <string name="restore">復元</string>
    <string name="backup">バックアップ</string>
    <string name="confirmation">確認</string>
    <string name="help">ヘルプ</string>
    <string name="error">エラー</string>
    <string name="status">状態</string>
    <string name="send">送信</string>
    <string name="edit">編集</string>
    <string name="delete">削除</string>
    <string name="add">追加</string>
    <string name="password">パスワード</string>
    <string name="deny">拒否</string>
    <string name="accept">承認</string>
    <string name="no">いいえ</string>
    <string name="ok">OK</string>
    <string name="cancel">キャンセル</string>
    <string name="info">情報</string>
    <string name="send_message">メッセージ送信</string>
    <string name="contacts">連絡先</string>
    <string name="contact_already_exists">連絡先名 %1$s は既に存在します</string>
    <string name="invalid_contact">無効な連絡先名 %1$s です</string>
    <string name="contact_name">名前</string>
    <string name="new_contact">新規連絡先</string>
    <string name="contact">連絡先</string>
    <string name="debug">デバック</string>
    <string name="dns_servers">DNSサーバー</string>
    <string name="start_automatically">自動的に開始</string>
    <string name="configuration">設定</string>
    <string name="delete_chats">削除</string>
    <string name="you">あなた</string>
    <string name="today">今日</string>
    <string name="sending_failed">メッセージの送信に失敗</string>
    <string name="add_contact">連絡先の追加</string>
    <string name="new_message">新規メッセージ</string>
    <string name="delete_history_alert">アカウント %1$s の通話履歴を削除しますか?</string>
    <string name="enable_history">有効化</string>
    <string name="disable_history">無効化</string>
    <string name="delete_history">削除</string>
    <string name="calls_call_message_question"> %1$s に電話をかけるか、メッセージを送信しますか?</string>
    <string name="calls_call">通話</string>
    <string name="calls_calls">通話</string>
    <string name="call">着信中</string>
    <string name="call_history">通話履歴</string>
    <string name="message_from">メッセージ着信</string>
    <string name="transfer_request_to">転送リクエスト</string>
    <string name="incoming_call_from">着信</string>
    <string name="reject">拒否</string>
    <string name="answer">応答</string>
    <string name="delete_account">アカウント %1$s を削除しますか?</string>
    <string name="decrypt_password">復号化パスワード</string>
    <string name="encrypt_password">暗号化パスワード</string>
    <string name="new_account">新規アカウント</string>
    <string name="accounts">アカウント</string>
    <string name="media_encryption">メディアの暗号化</string>
    <string name="stun_password">STUN/TURNパスワード</string>
    <string name="invalid_stun_username">無効な表示名 %1$s です</string>
    <string name="stun_username">STUN/TURNユーザー名</string>
    <string name="voicemail_messages">Voicemailメッセージ</string>
    <string name="voicemail">Voicemail</string>
    <string name="voicemail_uri">VoicemailのURI</string>
    <string name="authentication_username">認証ユーザー名</string>
    <string name="display_name_help">必要であればFrom URIで使用される名前を入力</string>
    <string name="your_name">Votre nom</string>
    <string name="display_name">表示名</string>
    <string name="account">アカウント</string>
    <string name="about_title_plus">baresip+について</string>
    <string name="about_title">baresipについて</string>
    <string name="invalid_display_name">無効な表示名 %1$s です</string>
    <string name="invalid_stun_password">無効な表示名 %1$s です</string>
    <string name="missed_call_from">着信失敗</string>
    <string name="yes">はい</string>
    <string name="show_password">パスワード表示</string>
    <string name="authentication_password">認証パスワード</string>
    <string name="invalid_authentication_password">無効な認証パスワード %1$s です</string>
    <string name="outbound_proxies">発信プレフィクス</string>
    <string name="sip_uri_of_proxy_server">SIP URIのプロキシサーバー</string>
    <string name="sip_uri_of_another_proxy_server">SIP URIの認証プロキシサーバー</string>
    <string name="invalid_proxy_server_uri">無効なプロキシサーバー %1$s です</string>
    <string name="register">登録</string>
    <string name="audio_codecs">オーディオコーデック</string>
    <string name="video_codecs">ビデオコーデック</string>
    <string name="media_nat">メディアのNAT探索</string>
    <string name="stun_server">STUN/TURNサーバー</string>
    <string name="stun_server_uri">STUN/TURNサーバーのURI</string>
    <string name="invalid_stun_server">無効なSTUN/TURNサーバーのURI %1$s です</string>
    <string name="invalid_voicemail_uri">無効なVoicemail URI %1$s です</string>
    <string name="invalid_aor">無効なuser@domain[:port][;transport=udp|tcp|tls] %1$s です</string>
    <string name="prefer_ipv6_media">IPv6メディアを優先</string>
    <string name="answer_mode">応答モード</string>
    <string name="manual">マニュアル</string>
    <string name="auto">自動</string>
    <string name="default_account">既定のアカウント</string>
    <string name="user_domain">user@domain</string>
    <string name="account_exists">アカウント %1$s は既に存在します</string>
    <string name="account_allocation_failure">新しいアカウントの割り当てに失敗しました。</string>
    <string name="chat">チャットメッセージ</string>
    <string name="chat_with"> %1$s とチャット</string>
    <string name="short_message_question">メッセージを削除しますか?</string>
    <string name="message_failed">失敗</string>
    <string name="chats">チャット履歴</string>
    <string name="new_chat_peer">新しいチャット相手</string>
    <string name="invalid_chat_peer_uri">SIP URIが無効です</string>
    <string name="short_chat_question"> %1$s とのチャットを削除しますか?</string>
    <string name="delete_chats_alert">アカウント %1$s とのチャット履歴を削除しますか?</string>
    <string name="listen_address">受信対象アドレス</string>
    <string name="invalid_listen_address">受信対象アドレスが無効です</string>
    <string name="invalid_dns_servers">DNSサーバーが無効です</string>
    <string name="failed_to_set_dns_servers">DNSサーバーの設定に失敗しました</string>
    <string name="tls_certificate_file">TLS証明書ファイル</string>
    <string name="tls_ca_file">TLS CA証明書ファイル</string>
    <string name="audio_modules_title">オーディオモジュール</string>
    <string name="failed_to_load_module">モジュールの読み込みに失敗しました</string>
    <string name="aec_help">チェックを入れると、通話音声のエコーキャンセルを試みます</string>
    <string name="aec_extended_filter">音響エコーキャンセル拡張フィルタ</string>
    <string name="opus_bit_rate">Opusのビットレート</string>
    <string name="opus_packet_loss">期待されるOpusのパケットロス</string>
    <string name="invalid_opus_bitrate">Opusのビットレートが無効です</string>
    <string name="invalid_opus_packet_loss">Opusパケットロス率が無効です</string>
    <string name="default_call_volume">既定の通話ボリューム</string>
    <string name="dark_theme">ダークテーマ</string>
    <string name="video_size">ビデオフレームサイズ</string>
    <string name="sip_trace">SIP追跡</string>
    <string name="reset_config">工場出荷時設定にリセット</string>
    <string name="read_cert_error">ダウンロード・ディレクトリから \'cert.pem\' の読み込みに失敗しました。</string>
    <string name="read_ca_certs_error">ダウンロード・ディレクトリから \'ca_certs.crt\' の読み込みに失敗しました。</string>
    <string name="invalid_contact_uri">無効なSIP URI %1$s です</string>
    <string name="alert">警告</string>
    <string name="notice">通知</string>
    <string name="user_id">ユーザーID</string>
    <string name="invalid_sip_uri">無効なSIP URI %1$s です</string>
    <string name="callee">呼び出し先</string>
    <string name="hold">保留</string>
    <string name="call_transfer">通話転送</string>
    <string name="transfer_destination">転送先</string>
    <string name="transfer_failed">転送失敗</string>
    <string name="dtmf">DTMF</string>
    <string name="call_info_not_available">情報はありません</string>
    <string name="call_info">電話情報</string>
    <string name="duration">通話時間 %1$d</string>
    <string name="codecs">コーデック</string>
    <string name="rate">レート: %1$s</string>
    <string name="you_have">あなたは</string>
    <string name="one_new_message">1件のメッセージ</string>
    <string name="new_messages">新規メッセージ</string>
    <string name="one_old_message">1件の古いメッセージ</string>
    <string name="old_messages">古いメッセージ</string>
    <string name="and">と</string>
    <string name="no_messages">メッセージはありません</string>
    <string name="messages">メッセージ</string>
    <string name="dialpad">ダイヤルパット</string>
    <string name="call_already_active">すでに通話中です</string>
    <string name="verify">確認要求</string>
    <string name="transfer_request">転送要求</string>
    <string name="call_failed">呼び出し失敗</string>
    <string name="call_closed">通話終了</string>
    <string name="call_not_secure">この通話は安全ではありません!</string>
    <string name="peer_not_verified">この通話は安全ですが、相手は確認されていません!</string>
    <string name="unverify">未検証</string>
    <string name="restart_request">再起動要求</string>
    <string name="no_video_calls">カメラにビデオ通話の許可を与えます</string>
    <string name="no_cameras">対応しているビデオカメラがありません</string>
    <string name="stun_username_help">STUN/TURN サーバーで必要な場合のユーザー名</string>
    <string name="stun_password_help">STUN/TURN サーバーで必要な場合のパスワード</string>
    <string name="prefer_ipv6_media_help">チェックを入れると、ピアのメディアプロトコルが自動的に決定できない場合に、IPv6メディアプロトコルの使用をオファーします(使用可能な場合)
    </string>
    <string name="answer_mode_help">着信した電話にどのように応答するかを選択します。</string>
    <string name="start_automatically_help">チェックを入れると、デバイスの再起動後に自動的にbaresipが起動します</string>
    <string name="aec">音響エコーキャンセル</string>
    <string name="aec_extended_filter_help">チェックを入れると、エコーキャンセルは拡張フィルタを使用しています。</string>
    <string name="default_call_volume_help">設定されている場合、デフォルトの通話音声の音量は1～10段階です</string>
    <string name="dark_theme_help">ダークテーマを強制する</string>
    <string name="debug_help">チェックを入れると、デバッグおよび情報レベルのログメッセージをLogcatに提供します。</string>
    <string name="reset_config_help">チェックを入れると、設定は工場出荷時のデフォルト値にリセットされます。</string>
    <string name="contact_action_question"> %1$s に電話をかけるか、メッセージを送信しますか？</string>
    <string name="contact_delete_question">連絡先 %1$s を削除しますか?</string>
    <string name="contacts_exceeded">連絡先の最大数 %1$d を超えました</string>
    <string name="allow_video"> %1$s でテレビ通話の送受信を許可しますか?</string>
    <string name="allow_video_send"> %1$s へのテレビ通話を許可しますか?</string>
    <string name="allow_video_recv"> %1$s からのテレビ通話を許可しますか?</string>
    <string name="listen">受信</string>
    <string name="registering_failed"> %1$s への登録失敗</string>
    <string name="verify_sas">SAS &lt;%1$s&gt; を検証しますか\?</string>
    <string name="transfer_request_query"> %1$s への通話転送を許可しますか?</string>
    <string name="backed_up">アプリケーションデータがダウンロードフォルダ %1$s にバックアップされました</string>
    <string name="no_calls">マイクへの権限付与がなく電話をかけたり、応答したりすることはできません。</string>
</resources>