<?xml version="1.0" encoding="utf-8"?>
<resources>
    <string name="app_name" translatable="false">ritosip</string>
    <string name="app_name_plus" translatable="false">ritosip</string>
    <!-- About Activity -->
    <string name="about_title">About ritosip</string>
    <string name="about_title_plus">About ritosip</string>
    <string name="about_text">
        <![CDATA[
        <h1>Ritosip 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 baresip\'s 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>Long touch on currently shown account enables or disables account\'s registration.</li>
            <li>Swipe left/right gesture toggles between the accounts.</li>
            <li>Previous call party can be reselect by touching the call icon when Callee is empty.</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>See <a href="https://github.com/juha-h/baresip-studio/wiki">Wiki</a> for more
                information.</li>
        </ul>
        <h2>Privacy Policy</h2>
            Privacy policy is available <a href="https://raw.githubusercontent.com/juha-h/baresip-studio/master/PrivacyPolicy.txt">here</a>.
        <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 codecs and TLS security</li>
            <li><b>AGPLv4</b> ZRTP media encryption</li>
            <li><b>GNU LGPL 2.1</b> G.722, G.726, and Codec2 codecs</li>
            <li><b>GNU GPLv3</b> G.729 codec</li>
        </ul>
        ]]>
    </string>
    <string name="about_text_plus">
        <![CDATA[
        <h1>Ritosip 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 ritosip\'s 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>Long touch on currently shown account enables or disables account\'s registration.</li>
            <li>Swipe left/right gesture toggles between the accounts.</li>
            <li>Previous call party can be reselect by touching the call icon when Callee is empty.</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>See <a href="https://github.com/juha-h/baresip-studio/wiki">Wiki</a> for more
                information.</li>
        </ul>
        <h2>Known Issues</h2>
        <ul>
            <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>Privacy Policy</h2>
            Privacy policy is available <a href="https://raw.githubusercontent.com/juha-h/baresip-studio/video/PrivacyPolicy.txt">here</a>.
        <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 codecs and TLS security</li>
            <li><b>AGPLv4</b> ZRTP media encryption</li>
            <li><b>GNU LGPL 2.1</b> G.722, G.726, and Codec2 codecs</li>
            <li><b>GNU GPLv3</b> G.729 codec</li>
            <li><b>GNU GPLv2</b> H.264 and H.265 codecs</li>
            <li><b>AOMedia</b> AV1 codec</li>
        </ul>
        ]]>
    </string>
    <!-- Account Activity -->
    <string name="account">Account</string>
    <string name="account_nickname">Account Nickname</string>
    <string name="account_nickname_help">Nickname (if any) used to identify this account within
        ritosip app.</string>
    <string name="nickname">Nickname</string>
    <string name="invalid_account_nickname">Invalid Account Nickname \'%1$s\'</string>
    <string name="non_unique_account_nickname">Nickname \'%1$s\' already exists</string>
    <string name="display_name">Display Name</string>
    <string name="your_name">Your Name</string>
    <string name="display_name_help">Name (if any) used in From URI of outbound requests.</string>
    <string name="invalid_display_name">Invalid Display Name \'%1$s\'</string>
    <string name="authentication_username">Authentication Username</string>
    <string name="authentication_username_help">Authentication username
    if authentication of SIP requests is required.  Default value is account\'s username.
    </string>
    <string name="invalid_authentication_username">Invalid Authentication Username \'%1$s\'</string>
    <string name="authentication_password">Authentication Password</string>
    <string name="authentication_password_help">Authentication
    Password up to 64 characters. If Authentication Username is given, but Password is not
    given, it will be asked when ritosip is started.
    </string>
    <string name="invalid_authentication_password">Invalid Authentication Password \'%1$s\'</string>
    <string name="outbound_proxies">Outbound Proxies</string>
    <string name="outbound_proxies_help">SIP URI of one or two proxies that must be used when sending requests.
        If two is given, REGISTER requests are sent to both and other requests are sent to
        one that responds.  If no outbound proxy is given, requests are sent based on
        DNS NAPTR/SRV/A record lookup of callee URI hostpart. If hostpart of SIP URI is an IPv6
        address, the address must be written inside brackets [].
\nExamples:
\n • sip:example.com:5061;transport=tls
\n • sip:[2001:67c:223:777::10];transport=tcp
\n • sip:192.168.43.50:443;transport=wss
    </string>
    <string name="sip_uri_of_proxy_server">SIP URI of Proxy Server</string>
    <string name="sip_uri_of_another_proxy_server">SIP URI of another Proxy Server</string>
    <string name="invalid_proxy_server_uri">Invalid Proxy Server URI \'%1$s\'</string>
    <string name="register">Register</string>
    <string name="register_help">If checked, registration is enabled and REGISTER requests are sent
        at the interval specified by Registration Interval.</string>
    <string name="reg_int">Registration Interval</string>
    <string name="reg_int_help">Tells how often (in seconds) ritosip sends REGISTER requests.
        Valid values are from 60 to 3600.</string>
    <string name="invalid_reg_int">Invalid Registration Interval\'%1$s\'</string>
    <string name="media_nat">Media NAT Traversal</string>
    <string name="media_nat_help">Selects media NAT traversal protocol (if any).  Possible choices are STUN
        (Session Traversal Utilities for NAT, RFC 5389) and ICE (Interactive Connectivity
        Establishment, RFC 5245).
    </string>
    <string name="stun_server">STUN/TURN Server</string>
    <string name="stun_server_help">A STUN/TURN Server URI of form scheme:host[:port][\?transport=udp|tcp],
        where scheme is \'stun\', \'stuns\', \'turn\', or \'turns\'. Factory default STUN Server
        for STUN and ICE protocols is \'stun:stun.l.google.com:19302\' pointing to public Google
        STUN server. There is no factory default TURN server.
    </string>
    <string name="stun_server_uri">STUN/TURN Server URI</string>
    <string name="invalid_stun_server">Invalid STUN/TURN Server URI \'%1$s\'</string>
    <string name="stun_server_default" translatable="false">stun:stun.l.google.com:19302</string>
    <string name="stun_username">STUN/TURN Username</string>
    <string name="stun_username_help">Username if required by STUN/TURN server</string>
    <string name="invalid_stun_username">Invalid Username \'%1$s\'</string>
    <string name="stun_password">STUN/TURN Password</string>
    <string name="stun_password_help">Password if required by STUN/TURN server</string>
    <string name="invalid_stun_password">Invalid Password \'%1$s\'</string>
    <string name="media_encryption">Media Encryption</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="prefer_ipv6_media">Prefer IPv6 Media</string>
    <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>
    <string name="rtcp_mux">RTCP Multiplexing</string>
    <string name="rtcp_mux_help">If checked, RTP and RTCP packets are multiplexed on a single port (RFC 5761).</string>
    <string name="rel_100">Reliable Provisional Responses</string>
    <string name="rel_100_help">If checked, indicate support for reliable provisional responses (RFC 3262).</string>
    <string name="dtmf_mode">DTMF Mode</string>
    <string name="dtmf_mode_help">Selects how DTMF tones 0–9, #, *, and A-D are sent.</string>
    <string name="dtmf_inband">In-band RTP Events</string>
    <string name="dtmf_info">SIP INFO Requests</string>
    <string name="dtmf_auto">In-band RTP or SIP INFO</string>
    <string name="answer_mode">Answer Mode</string>
    <string name="answer_mode_help">Selects how incoming calls are answered.</string>
    <string name="redirect_mode">Redirect Mode</string>
    <string name="redirect_mode_help">Selects if call redirect request is followed automatically or
        if confirmation is requested.</string>
    <string name="manual">Manual</string>
    <string name="auto">Automatic</string>
    <string name="voicemail_uri">Voicemail URI</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="invalid_voicemail_uri">Invalid Voicemail URI \'%1$s\'</string>
    <string name="country_code">Country Code</string>
    <string name="country_code_help">E.164 country code of this account.  If From URI userpart of
        incoming call or message contains a telephone number that does not start with \'+\' sign and if contact
        lookup fails, the number is prefixed with this country code and contact lookup is
        tried again. If the telephone number starts with a single digit \'0\', digit \'0\' is removed
        before the number is prefixed.
    </string>
    <string name="country_code_hint">+code</string>
    <string name="invalid_country_code">Invalid Country Code \'%1$s\'</string>
    <string name="telephony_provider">Telephony Provider</string>
    <string name="telephony_provider_help">SIP URI host part used in calls to telephone numbers.
        Factory default is account\'s domain. If not given, this account cannot be used to call
        telephone numbers.
    </string>
    <string name="telephony_provider_hint">SIP URI host part</string>
    <string name="invalid_sip_uri_hostpart">Invalid SIP URI host part \'%1$s\'</string>
    <string name="default_account">Default Account</string>
    <string name="default_account_help">If checked, this account is selected when ritosip is started.
    </string>
    <!-- Accounts Activity -->
    <string name="accounts">Accounts</string>
    <string name="new_account">New Account</string>
    <string name="accounts_help">When a new account is created, account\'s port number and transport
        protocol may be optionally given: &lt;user&gt;@&lt;domain&gt;[:&lt;port&gt;][;transport=udp|tcp|tls].
        If &lt;port&gt; is given and transport protocol is not given, transport protocol defaults to udp.
        If &lt;port&gt; is not given and transport protocol is given, &lt;port&gt; defaults to 5060 or 5061 (TLS).
        If neither is given and no outbound proxy is specified, account\'s registrar (if any) is
        determined solely based on domain\'s DNS information.
    </string>
    <string name="user_domain">user@domain</string>
    <string name="invalid_aor">Invalid user@domain[:port][;transport=udp|tcp|tls] \'%1$s\'</string>
    <string name="account_exists">Account \'%1$s\' already exists.</string>
    <string name="account_allocation_failure">"Failed to allocate new account.</string>
    <string name="encrypt_password">Encrypt Password</string>
    <string name="decrypt_password">Decrypt Password</string>
    <string name="delete_account">Do you want to delete account \'%1$s\'\?</string>
    <!-- Ritosip Service -->
    <string name="answer">Answer</string>
    <string name="reject">Reject</string>
    <string name="incoming_call_from">Incoming call from</string>
    <string name="missed_call_from">Missed call from</string>
    <string name="missed_calls">Missed calls</string>
    <string name="missed_calls_count">%1$d missed calls</string>
    <string name="transfer_request_to">Call transfer request to</string>
    <string name="message_from">Message from</string>
    <string name="call_auto_rejected">Auto-rejected call from \`%1$s\`</string>
    <!-- Calls Activity -->
    <string name="call_history">Call History</string>
    <string name="call_details">Call Details</string>
    <string name="call">Call</string>
    <string name="calls_calls">calls</string>
    <string name="calls_call">call</string>
    <string name="peer">Peer</string>
    <string name="direction">Direction</string>
    <string name="time">Time</string>
    <string name="calls_duration">Duration</string>
    <string name="calls_call_message_question">Do you want to call or send message to \'%1$s\'\?</string>
    <string name="calls_add_delete_question">Do you want to add \'%1$s\' to contacts or delete
        %2$s from call history\?
    </string>
    <string name="calls_delete_question">Do you want to delete \'%1$s\' %2$s from call history\?
    </string>
    <string name="delete_history">Delete</string>
    <string name="disable_history">Disable</string>
    <string name="enable_history">Enable</string>
    <string name="delete_history_alert">Do you want to delete call history of account \'%1$s\'\?</string>
    <!-- Chat Activity -->
    <string name="chat">Chat Messages</string>
    <string name="chat_with">Chat with %1$s</string>
    <string name="new_message">New message</string>
    <string name="long_message_question">Do you want to delete message or add peer \'%1$s\' to contacts\?</string>
    <string name="short_message_question">Do you want to delete message\?</string>
    <string name="add_contact">Add Contact</string>
    <string name="sending_failed">Sending of message failed</string>
    <string name="message_failed">Failed</string>
    <!-- Chats Activity -->
    <string name="chats">Chat History</string>
    <string name="today">Today</string>
    <string name="you">You</string>
    <string name="new_chat_peer">New Chat Peer</string>
    <string name="invalid_chat_peer_uri">Invalid chat peer URI</string>
    <string name="long_chat_question">Do you want to delete chat with peer \'%1$s\' or
        add peer to contacts\?</string>
    <string name="short_chat_question">Do you want to delete chat with \'%1$s\'\?</string>
    <string name="delete_chats">Delete</string>
    <string name="delete_chats_alert">Do you want to delete chat history of account \'%1$s\'\?</string>
    <!-- Codecs Activity -->
    <string name="audio_codecs">Audio Codecs</string>
    <string name="audio_codecs_help">List of audio codecs in priority order. Drag to reorder,
        swipe right to enable or disable.</string>
    <string name="video_codecs">Video Codecs</string>
    <string name="video_codecs_help">List of video codecs in priority order. Drag to reorder,
        swipe right to enable or disable.</string>
    <string name="codec_action">Reorder</string>
    <!-- Config Activity -->
    <string name="configuration">Settings</string>
    <string name="start_automatically">Start Automatically</string>
    <string name="start_automatically_help">If checked, ritosip starts automatically after device (re)start.</string>
    <string name="appear_on_top_permission">Automatic start needs Appear on Top permission.</string>
    <string name="battery_optimizations">Battery Optimizations</string>
    <string name="battery_optimizations_help">Disable battery optimizations (recommended) if you want
        to reduce likelihood that Android restricts baresip\'s access to network or enters baresip
        to standby state.</string>
    <string name="default_phone_app">Default Phone App</string>
    <string name="dialer_role_not_available">Dialer role is not available</string>
    <string name="default_phone_app_help">If checked, ritosip is the default phone app.  Do not check
        if your device may need to handle also other than SIP calls or messages.</string>
    <string name="listen_address">Listen Address</string>
    <string name="listen_address_help">IP address and port of form \'address:port\' at which ritosip 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 ritosip listen at all
        available addresses. If left empty (factory default), ritosip listens at port 5060 of
        all available addresses.
    </string>
    <string name="invalid_listen_address">Invalid Listen Address</string>
    <string name="_0_0_0_0_5060" translatable="false">0.0.0.0:5060</string>
    <string name="address_family">Address Family</string>
    <string name="address_family_help">Chooses which IP addresses ritosip is
        using.  If IPv4 or IPv6 is chosen, ritosip uses only IPv4 or IPv6 addresses.  If neither is
        chosen, ritosip uses both IPv4 and IPv6 addresses.
    </string>
    <string name="dns_servers">DNS Servers</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="invalid_dns_servers">Invalid DNS Servers</string>
    <string name="failed_to_set_dns_servers">Failed to set DNS servers</string>
    <string name="tls_certificate_file">TLS Certificate File</string>
    <string name="tls_certificate_file_help">If checked, a file containing TLS certificate and
        private key of this ritosip instance has been or will be loaded. In Android version 9,
        a file called \'cert.pem\' is loaded from Download folder. For security reasons,
        delete the file after loading.</string>
    <string name="verify_server">Verify Server Certificates</string>
    <string name="verify_server_help">If checked, ritosip verifies TLS certificates of SIP User
        Agent and SIP Proxy Servers when TLS transport is used.</string>
    <string name="tls_ca_file">TLS CA File</string>
    <string name="tls_ca_file_help">If checked, a file has been or will be loaded that contains
        TLS certificates of such Certificate Authorities that are not included in Android OS.
        In Android version 9, a file called \'ca_certs.crt\' is loaded from Download folder.</string>
    <string name="audio_settings">Audio Settings</string>
    <string name="speaker_phone">Speaker Phone</string>
    <string name="speaker_phone_help">If checked, speaker phone is turned automatically on
        when call starts.</string>
    <string name="audio_modules_title">Audio Modules</string>
    <string name="audio_modules_help">Audio codecs provided by the checked modules are
        available for use by the accounts.</string>
    <string name="failed_to_load_module">Failed to load module.</string>
    <string name="aec">Acoustic Echo Cancellation</string>
    <string name="aec_help">If checked, software echo cancellation is attempted on call audio.</string>
    <string name="aec_extended_filter">AEC Extended Filter</string>
    <string name="aec_extended_filter_help">If checked, echo cancellation is using extended filter.</string>
    <string name="microphone_gain">Microphone Gain</string>
    <string name="microphone_gain_help">Multiply microphone volume by this decimal number.  Minimum
        value is 1.0 (factory default) that disables microphone gain. Larger values may negatively
        affect audio quality.</string>
    <string name="invalid_microphone_gain">Invalid Microphone Gain value</string>
    <string name="_1.0" translatable="false">1.0</string>
    <string name="opus_bit_rate">Opus Bit Rate</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="_28000" translatable="false">28000</string>
    <string name="opus_packet_loss">Expected Opus packet-loss</string>
    <string name="opus_packet_loss_help">Expected Opus audio stream packet loss percentage,
        from 0–100. Factory default value is 1. Value 0 also turns off Opus Forward Error
        Correction (FEC).</string>
    <string name="_0" translatable="false">0</string>
    <string name="invalid_opus_bitrate">Invalid Opus bitrate</string>
    <string name="invalid_opus_packet_loss">Invalid Opus Packet Loss Percentage</string>
    <string name="audio_delay">Audio Delay</string>
    <string name="audio_delay_help">Time (in milliseconds) to wait audio from callee when call is established.
        Set to a higher value if you miss audio from callee at the beginning of the call.</string>
    <string name="invalid_audio_delay">Invalid Audio Delay \'%1$s\'. Valid values are from 100 to 3000.</string>
    <string name="default_call_volume">Default Call Volume</string>
    <string name="default_call_volume_help">If set, default call audio volume at scale 1–10.</string>
    <string name="tone_country">Tone Country</string>
    <string name="tone_country_help">Country of call ringing, waiting, and callee busy tones</string>
    <string name="dark_theme">Dark Theme</string>
    <string name="dark_theme_help">Force dark display theme</string>
    <string name="video_size">Video Frame Size</string>
    <string name="video_size_help">Size of transmitted video frames (width x height)</string>
    <string name="video_fps">Video Frames Per Second</string>
    <string name="video_fps_help">Video frame rate that will be offered during the SDP handshake.
        Valid values are from 10 to 30.</string>
    <string name="invalid_fps">Invalid Frames Per Second \'%1$d\'</string>
    <string name="user_agent">User Agent</string>
    <string name="user_agent_help">Custom SIP request/response User-Agent header field value</string>
    <string name="invalid_user_agent">Invalid User-Agent header field value</string>
    <string name="contacts_help">Chooses if ritosip contacts, Android contacts, or both are used.
        If both are used and a contact with the same name exists in both contacts,
        the ritosip contact will be chosen.</string>
    <string name="both">Both</string>
    <string name="debug">Debug</string>
    <string name="debug_help">If checked, provides debug and info level log messages to Logcat.</string>
    <string name="sip_trace">SIP Trace</string>
    <string name="sip_trace_help">If checked and if Debug is checked, Logcat messages include also SIP
        request and response trace. Unchecked automatically at ritosip start.</string>
    <string name="reset_config">Reset to Factory Defaults</string>
    <string name="reset_config_help">If checked, settings are reset to factory default values.</string>
    <string name="reset_config_alert">Are you sure you want to reset settings to factory
        default values\?</string>
    <string name="reset">Reset</string>
    <string name="read_cert_error">Failed to read file \'cert.pem\'.</string>
    <string name="read_ca_certs_error">Failed to read file \'ca_certs.crt\'.</string>
    <string name="config_restart">You need to restart ritosip in order to activate the new
        settings. Restart now\?
    </string>
    <string name="consent_request">Consent Request</string>
    <string name="contacts_consent">If Android contacts is chosen, they can be used
        in calling and messaging as references to SIP and tel URIs. ritosip app does not store Android
        contacts nor share them with anyone.  In order to make Android contacts available in baresip,
        Google requires that you accept their use as described here and in app\'s
        <a href="https://raw.githubusercontent.com/juha-h/baresip-studio/master/PrivacyPolicy.txt">Privacy Policy</a>.
    </string>
    <!-- Contact Activity -->
    <string name="contact">Contact</string>
    <string name="new_contact">New Contact</string>
    <string name="contact_name">Name</string>
    <string name="sip_or_tel_uri">SIP or tel URI</string>
    <string name="user_domain_or_number">user@domain or telephone number</string>
    <string name="favorite">Favorite</string>
    <string name="invalid_contact">Invalid contact name \'%1$s\'</string>
    <string name="contact_already_exists">Contact \'%1$s\' already exists.</string>
    <string name="invalid_contact_uri">Invalid SIP URI</string>
    <string name="android_contact_help">If checked, this contact is added to Android contacts.</string>
    <string name="avatar_image">Profile image</string>
    <!-- Contacts Activity -->
    <string name="contacts">Contacts</string>
    <string name="contact_action_question">Do you want to call or send message to \'%1$s\'\?</string>
    <string name="send_message">Send Message</string>
    <string name="contact_delete_question">Do you want to delete contact \'%1$s\'\?</string>
    <string name="contacts_exceeded">Your maximum number of contacts %1$d has been exceeded.</string>
    <!-- Generic -->
    <string name="alert">Alert</string>
    <string name="info">Info</string>
    <string name="notice">Notice</string>
    <string name="cancel">Cancel</string>
    <string name="ok">OK</string>
    <string name="yes">Yes</string>
    <string name="no">No</string>
    <string name="accept">Accept</string>
    <string name="deny">Deny</string>
    <string name="user_id">User ID</string>
    <string name="password">Password</string>
    <string name="sip_uri" translatable="false">SIP URI</string>
    <string name="add">Add</string>
    <string name="delete">Delete</string>
    <string name="edit">Edit</string>
    <string name="send">Send</string>
    <string name="status">Status</string>
    <string name="error">Error</string>
    <string name="help">Help</string>
    <string name="confirmation">Confirmation</string>
    <string name="anonymous">Anonymous</string>
    <string name="unknown">Unknown</string>
    <string name="dots" translatable="false">…</string>
    <string name="bullet_item" translatable="false">\u2022 %1$s</string>
    <string name="invalid_sip_or_tel_uri">Invalid SIP or tel URI \'%1$s\'</string>
    <string name="baresip" translatable="false">baresip</string>
    <string name="android" translatable="false">Android</string>
    <!-- Main Activity -->
    <string name="backup">Backup</string>
    <string name="restore">Restore</string>
    <string name="logcat" translatable="false">Logcat</string>
    <string name="about">About</string>
    <string name="restart">Restart</string>
    <string name="quit">Quit</string>
    <string name="outgoing_call_to_dots">Call to …</string>
    <string name="incoming_call_from_dots">Call from …</string>
    <string name="diverted_by_dots">Diverted by …</string>
    <string name="transferring_call_to_dots">Transferring call to …</string>
    <string name="invalid_sip_uri">Invalid SIP URI \'%1$s\'</string>
    <string name="no_telephony_provider">Account \'%1$s\' has no Telephony Provider</string>
    <string name="callee">Callee</string>
    <string name="hangup">Hangup</string>
    <string name="video_call">Video call</string>
    <string name="video_request">Video Request</string>
    <string name="allow_video">Accept sending and receiving video with \'%1$s\'\?</string>
    <string name="allow_video_send">Accept sending of video to \'%1$s\'\?</string>
    <string name="allow_video_recv">Accept receiving video from \'%1$s\'\?</string>
    <string name="hold">Call Hold/Unhold</string>
    <string name="call_is_on_hold">Call is on hold</string>
    <string name="mic">Microphone On/Off</string>
    <string name="rec_in_call">Recording can be turned on or off only when call is not
        connected</string>
    <string name="call_transfer">Call Transfer</string>
    <string name="blind">Blind</string>
    <string name="attended">Attended</string>
    <string name="transfer_destination">Transfer destination</string>
    <string name="choose_destination_uri">Choose destination URI</string>
    <string name="transfer">Transfer</string>
    <string name="transfer_failed">Transfer failed</string>
    <string name="dtmf">DTMF</string>
    <string name="call_info">Call Info</string>
    <string name="call_info_not_available">No info available</string>
    <string name="duration">Duration: %1$d (secs)</string>
    <string name="codecs">Codecs</string>
    <string name="rate">Current Rate: %1$s (Kbits/s)</string>
    <string name="average_rate">Average Rate: %1$s (Kbits/s)</string>
    <string name="packets">Packets</string>
    <string name="lost">Lost</string>
    <string name="jitter">Jitter: %1$s (ms)</string>
    <string name="voicemail">Voicemail</string>
    <string name="voicemail_messages">Voicemail Messages</string>
    <string name="you_have">You have</string>
    <string name="one_new_message">one new message</string>
    <string name="new_messages">new messages</string>
    <string name="one_old_message">one old message</string>
    <string name="old_messages">old messages</string>
    <string name="and">and</string>
    <string name="no_messages">You have no messages</string>
    <string name="listen">Listen</string>
    <string name="messages">Messages</string>
    <string name="dialpad">Dialpad</string>
    <string name="call_already_active">You already have an active call.</string>
    <string name="start_failed">Ritosip failed to start. This may be due to an invalid Settings value.
        Check Listen Address, TLS Certificate File, and TLS CA File.  Then restart baresip.
    </string>
    <string name="registering_failed">Registering of \`%1$s\` failed.</string>
    <string name="verify">Verify Request</string>
    <string name="verify_sas">Do you want to verify SAS &lt;%1$s&gt;\?</string>
    <string name="transfer_request">Transfer Request</string>
    <string name="transfer_request_query">Do you accept to transfer this call to \'%1$s\'\?</string>
    <string name="call_request">Call Request</string>
    <string name="call_request_query">Do you accept request to call \'%1$s\'\?</string>
    <string name="redirect_notice">Automatic redirection to \'%1$s\'\</string>
    <string name="redirect_request">Redirect Request</string>
    <string name="redirect_request_query">Do you accept call redirection to \'%1$s\'\?</string>
    <string name="call_failed">Call failed</string>
    <string name="call_closed">Call is closed</string>
    <string name="call_not_secure">This call is NOT secure!</string>
    <string name="peer_not_verified">This call is SECURE, but peer is NOT verified!</string>
    <string name="call_is_secure">This call is SECURE and peer is VERIFIED!
        Do you want to unverify the peer\?
    </string>
    <string name="unverify">Unverify</string>
    <string name="backed_up">Application data (excluding recordings) backed up
        to file \'%1$s\'. In Android version 9, the file is in Download folder.</string>
    <string name="backup_failed">Failed to back up application data to file
        \'%1$s\'. Check Apps → ritosip → Permissions → Storage.</string>
    <string name="restart_request">Restart Request</string>
    <string name="restored">Application data restored. ritosip needs to be restarted.
        Restart now\?
    </string>
    <string name="restore_failed">Failed to restore application data. Check that you gave correct
        password and that the backup file is from this application. In Android versions 9,
        also check Apps → ritosip → Permissions → Storage and that file \'%1$s\' exists
        in Download folder.
    </string>
    <string name="restore_unzip_failed">Failed to restore application data. Android version 14 and
        above does not allow restoring data that was backed up before %1$s version %2$s.
    </string>
    <string name="no_notifications">You are not able to use this application without \"Notifications\"
        permission.</string>
    <string name="no_calls">ritosip needs \"Microphone\" permission for voice calls.</string>
    <string name="no_bluetooth">ritosip is not able to detect Bluetooth connectivity without
        \"Nearby devices\" permission.</string>
    <string name="no_video_calls">Grant \"Camera\" permission to make or answer video calls.</string>
    <string name="no_backup">You are not able create backup without \"Storage\" permission.</string>
    <string name="no_restore">You are not able restore backup without \"Storage\" permission.</string>
    <string name="no_android_contacts">You are not able to access Android contacts without
        \"Contacts\" permission.</string>
    <string name="no_cameras">You don\'t have any supported video cameras.</string>
    <string name="show_password">Show Password</string>
    <string name="no_network">No network connection!</string>
    <string name="audio_focus_denied">Audio focus denied!</string>
    <string name="permissions_rationale">Permissions rationale</string>
    <string name="audio_permissions">ritosip needs \"Microphone\" permission for voice calls,
        \"Nearby devices\" permission for Bluetooth microphone/speaker detection, and
        \"Notifications\" permission for posting notifications.</string>
    <string name="audio_and_video_permissions">ritosip needs \"Microphone\" permission for voice calls,
        \"Camera\" permission for video calls, \"Nearby devices\" permission for Bluetooth
        microphone/speaker detection, and \"Notifications\" permission for posting notifications.</string>
</resources>