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<?xml version="1.0" encoding="utf-8"?>
<resources xmlns:tools="http://schemas.android.com/tools">
<string name="app_name" translatable="false">baresip</string>
<string name="app_name_plus" translatable="false">baresip+</string>
<!-- About Activity -->
<string name="about_title">About baresip</string>
<string name="about_title_plus">About baresip+</string>
<string name="about_text">
<![CDATA[
<h1>SIP and mobile calling User Agent based on
<a href="https://github.com/baresip/baresip">baresip</a> library</h1>
<br>
<p>Juha Heinanen &lt;jh@tutpro.com&gt;</p>
<p>Version %1$s</p>
<br>
<h2>Usage Hints</h2>
<ul>
<li>Check that default values in baresip\'s Settings meet your needs
(touch item titles for help).</li>
<li>If you want use baresip for mobile calling, turn on Default Phone App setting. It will
create Mobile account for mobile calling.</li>
<li>Then in Accounts, create one or more SIP 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 status dot leads directly to account configuration.</li>
<li>Long touch on baresip bar icons shows information about the icons.</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 on contact icon can be used to install/remove image avatar.</li>
<li>Long touch on an audio codec can be used to enable/disable the codec.</li>
<li>See <a href="https://github.com/juha-h/baresip-studio/wiki">Wiki</a> for more information.</li>
</ul>
<h2>Privacy Policy</h2>
<p>Privacy policy is available <a href="https://raw.githubusercontent.com/juha-h/baresip-studio/master/PrivacyPolicy.txt">here</a>.</p>
<br>
<h2>Source Code</h2>
<p>Source code is available at <a href="https://github.com/juha-h/baresip-studio">GitHub</a>,
where also issues can be reported.</p>
<br>
<h2>Language Translations</h2>
<p>Language translations are managed via baresip
<a href="https://hosted.weblate.org/projects/baresip/">Weblate</a> project.</p>
<br>
<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> Codec2 codec</li>
<li><b>Free</b> G.722 codec</li>
<li><b>GNU GPLv3</b> G.729 codec</li>
</ul>
]]>
</string>
<string name="about_text_plus">
<![CDATA[
<h1>SIP User Agent with video calls based on <a href="https://github.com/baresip/baresip">baresip</a> library</h1>
<br>
<p>Juha Heinanen &lt;jh@tutpro.com&gt;</p>
<p>Version %1$s</p>
<br>
<h2>Usage Hints</h2>
<ul>
<li>Check that default values in baresip+\'s Settings meet your needs
(touch item titles for help).</li>
<li>If you want use baresip+ for mobile calling, turn on Default Phone App setting. It will
create Mobile account for mobile calling.</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 status dot leads directly to account configuration.</li>
<li>Long touch on baresip bar icons shows information about the icons.</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>Long touch on an audio or video codec can be used to enable/disable the codec.</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>Selfview is not properly shown when video stream is sendonly.</li>
</ul>
<h2>Privacy Policy</h2>
<p>Privacy policy is available
<a href="https://raw.githubusercontent.com/juha-h/baresip-studio/video/PrivacyPolicy.txt">here</a>.</p>
<br>
<h2>Source code</h2>
<p>Source code is available at <a href="https://github.com/juha-h/baresip-studio">GitHub</a>,
where also issues can be reported.</p>
<br>
<h2>Language translations</h2>
<p>Language translations are managed via baresip
<a href="https://hosted.weblate.org/projects/baresip/">Weblate</a> project.</p>
<br>
<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> Codec2 codec</li>
<li><b>Free</b> G.722 codec</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="sip_uri" translatable="false">SIP URI</string>
<string name="tel_uri" translatable="false">TEL URI</string>
<string name="tel_uri_calls">TEL URI for Calls</string>
<string name="tel_uri_messages">TEL URI for Messages</string>
<string name="not_available">Not available</string>
<string name="account_nickname_help">Nickname (if any) used to identify this account within
baresip 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="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 baresip 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) baresip sends REGISTER requests.
Valid values are from 60 to 3600.</string>
<string name="invalid_reg_int">Invalid Registration Interval \'%1$s\'</string>
<string name="check_origin">Check Origin</string>
<string name="check_origin_help">If checked, inbound requests are allowed only from IP addresses
where register requests were sent.
</string>
<string name="media_nat">Media NAT Traversal</string>
<string name="media_nat_help">Selects media NAT traversal protocol (if any). Possible choices are STUN
(Session Traversal Utilities for NAT, RFC 5389) and ICE (Interactive Connectivity
Establishment, RFC 5245).
</string>
<string name="stun_server">STUN/TURN Server</string>
<string name="stun_server_help">A STUN/TURN Server URI of form scheme:host[:port][\?transport=udp|tcp],
where scheme is \'stun\', \'stuns\', \'turn\', or \'turns\'. Factory default STUN Server
for STUN and ICE protocols is \'stun:stun.l.google.com:19302\' pointing to public Google
STUN server. There is no factory default TURN server.
</string>
<string name="invalid_stun_server">Invalid STUN/TURN Server URI \'%1$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="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 09, #, *, 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="block_unknown">Block Unknown</string>
<string name="block_unknown_help">Block calls and messages from peers that are not found in contacts.</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="voicemain_tel_uri_help">TEL URI for checking of voicemail messages.</string>
<string name="country_code">Country Code</string>
<string name="country_code_help">E.164 country code of this account. If peer URI userpart of
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 number starts with a single \'0\' digit, it is removed
before the number is prefixed.
</string>
<string name="invalid_country_code">Invalid Country Code \'%1$s\'</string>
<string name="telephony_provider">Telephony Provider</string>
<string name="telephony_provider_help">SIP URI host part used in calls to telephone numbers.
Factory default is account\'s domain. If not given, this account cannot be used to call
telephone numbers.
</string>
<string name="numeric_keypad">Numeric Keypad</string>"
<string name="numeric_keypad_help">If checked, numeric keypad is shown when \"Call to …\"
field is focused.</string>"
<string name="invalid_sip_uri_hostpart">Invalid SIP URI host part \'%1$s\'</string>
<string name="default_account">Default Account</string>
<string name="default_account_help">If checked, this account is selected when baresip is started.
</string>
<string name="custom_parameters">Custom Parameters</string>
<string name="custom_parameters_help">Semicolor separeted list of custom account parameters</string>
<!-- Accounts Activity -->
<string name="accounts">Accounts</string>
<string name="new_account">SIP URI of New Account</string>
<string name="accounts_help">SIP URI of new account of form
&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="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$s\'\?</string>
<!-- Baresip Service -->
<string name="is_calling">is calling</string>
<string name="missed_call_from">Missed call from</string>
<string name="missed_calls">Missed calls</string>
<string name="missed_calls_count" tools:ignore="PluralsCandidate">%1$d missed calls</string>
<string name="transfer_request_to">Call transfer request to</string>
<string name="call_auto_rejected">Auto-rejected call from \`%1$s\`</string>
<string name="call_blocked">Auto-rejected blocked call from \`%1$s\`</string>
<string name="message_blocked">Auto-rejected blocked message from \`%1$s\`</string>
<!-- Blocked Activity -->
<string name="blocked">Blocked</string>
<string name="blocked_calls">Blocked Calls</string>
<string name="blocked_messages">Blocked Messages</string>
<string name="blocked_delete_alert">Do you want to delete blocked requests of \'%1$s\'\?</string>
<string name="blocked_contact_question">Do you want to add peer \'%1$s\' to contacts\?</string>
<!-- Calls Activity -->
<string name="call_history">Call History</string>
<string name="call_details">Call Details</string>
<string name="call">Call</string>
<string name="calls_calls">calls</string>
<string name="calls_call">call</string>
<string name="peer">Peer</string>
<string name="direction">Direction</string>
<string name="time">Time</string>
<string name="calls_duration">Duration</string>
<string name="calls_add_delete_question">Do you want to add \'%1$s\' to contacts, copy its URI,
or delete this %2$s from call history\?</string>
<string name="calls_delete_question">Do you want to copy URI of \'%1$s\' or delete this %2$s
from call history\?</string>
<string name="disable_history">Disable</string>
<string name="enable_history">Enable</string>
<string name="delete_history_alert">Do you want to delete call history of account \'%1$s\'\?</string>
<string name="call_answered">Call answered</string>
<string name="call_answered_elsewhere">Call answered elsewhere</string>
<string name="call_missed">Call missed</string>
<string name="call_rejected">Call rejected</string>
<!-- Call Details Activity -->
<string name="playing_recording">Playing recording …</string>
<string name="save_recording">Save Recording</string>
<string name="save_recording_question">Do you want to save this recording?</string>
<string name="recording_saved">Recording saved</string>
<string name="delete_call_alert">Do you want to delete this call from history?</string>
<!-- Chat Activity -->
<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="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_alert">Do you want to delete chat history of account \'%1$s\'\?</string>
<!-- Codecs Activity -->
<string name="audio_codecs">Audio Codecs</string>
<string name="video_codecs">Video Codecs</string>
<!-- Settings Activity -->
<string name="configuration">Settings</string>
<string name="start_automatically">Start Automatically</string>
<string name="start_automatically_help">If checked, baresip starts automatically after device boots
or after a new version of baresip is installed. Start is delayed until device is unlocked.
</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 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">Default phone app role is not available</string>
<string name="default_phone_app_help">If checked, baresip is the default phone app.</string>
<string name="default_messaging_app">Default Messaging App</string>
<string name="default_messaging_app_help">If checked, baresip is the default messaging app for SMS/MMS.</string>
<string name="messaging_role_not_available">Default messaging app role is not available</string>
<string name="enable_default_messaging">Please enable Default Messaging App in Settings</string>
<string name="listen_address">Listen Address</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 an arbitrary port at
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 baresip is
using. If IPv4 or IPv6 is chosen, baresip uses only IPv4 or IPv6 addresses. If neither is
chosen, baresip uses both IPv4 and IPv6 addresses.
</string>
<string name="transport_protocols">Transport Protocols</string>
<string name="transport_protocols_help">Comma separated list of supported SIP request/response
transport protocols. If left empty, default value is \'udp,tcp,tls,ws,wss\' that includes
all supported transport protocols. List only the ones you need. Use of UDP is not
recommended for security and other reasons.
</string>
<string name="invalid_transport_protocols">Invalid Transport Protocols list</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 baresip 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, baresip 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="no_read_permission">No External Storage read permission</string>
<string name="unique_contact_uri">Unique Contact URI</string>
<string name="unique_contact_uri_help">If checked, Contact URI is guaranteed to be unique.
Needs to be checked if there is more than one account with the same SIP URI userpart,
but also improves protection of the accounts from attacks.</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="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="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="opus_packet_loss">Expected Opus packet-loss</string>
<string name="opus_packet_loss_help">Expected Opus audio stream packet loss percentage,
from 0100. Factory default value is 1. Value 0 also turns off Opus Forward Error
Correction (FEC).</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 110.</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="dynamic_colors">Dynamic Colors</string>
<string name="dynamic_colors_help">Use dynamic colors if enabled in Android Settings</string>
<string name="colorblind">Colorblind</string>
<string name="colorblind_help">Use colorblind friendly registration status icons</string>
<string name="proximity_sensing">Proximity Sensing</string>
<string name="proximity_sensing_help">If checked, proximity sensing is active during calls.</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="ringtone">Ringtone</string>
<string name="select_ringtone">Select Ringtone</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 baresip contacts, Android contacts, or both are used.
If both are used and a contact with the same name exists in both contacts,
the baresip 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 baresip 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 baresip 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. baresip 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="new_contact">New Contact</string>
<string name="contact_name">Name</string>
<string name="label">Label</string>
<string name="email">Email</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="favorite_help">If checked, contact is shown among other favorites
on top of contacts list.</string>
<string name="invalid_contact">Invalid contact name \'%1$s\'</string>
<string name="contact_already_exists">Contact \'%1$s\' already exists.</string>
<string name="contact_no_sip_or_tel_uri">Contact \'%1$s\' has no SIP or TEL URI</string>
<string name="contact_no_tel_uri">Contact \'%1$s\' has no TEL URI</string>
<string name="android_contact_help">If checked, this contact is added to Android contacts.</string>
<string name="avatar_image">Profile image</string>
<!-- Contacts Activity -->
<string name="contacts">Contacts</string>
<string name="contact_action_question">Do you want to call or send message to \'%1$s\'\?</string>
<string name="contact_email_action_question">Do you want to call, send message, or email to \'%1$s\'\?</string>
<string name="send_message">Send Message</string>
<string name="send_email">Send Email</string>
<string name="contact_delete_question">Do you want to delete contact \'%1$s\'\?</string>
<string name="contacts_delete_question">Do you want to delete all contacts?</string>
<string name="search">Search</string>
<string name="import_contacts">Import</string>
<string name="export_contacts">Export</string>
<string name="contact_import_success">Contacts imported successfully</string>
<string name="contact_import_failure">Failed to import contacts</string>
<string name="contact_export_success">Contacts exported successfully</string>
<string name="contact_export_failure">Failed to export contacts</string>
<!-- Generic -->
<string name="alert">Alert</string>
<string name="info">Info</string>
<string name="notice">Notice</string>
<string name="cancel">Cancel</string>
<string name="stop">Stop</string>
<string name="reply">Reply</string>
<string name="save">Save</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="add">Add</string>
<string name="delete">Delete</string>
<string name="edit">Edit</string>
<string name="copy_uri">Copy URI</string>
<string name="copied_to_clipboard">Copied to clipboard</string>
<string name="status">Status</string>
<string name="error">Error</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="diverted_by">diverted by</string>
<string name="no_telephony_provider">Account \'%1$s\' has no Telephony Provider</string>
<string name="no_telephone_number">Mobile call only be made to telephone number</string>
<string name="video_call">Video call</string>
<string name="video_request">Video Request</string>
<string name="allow_video">Accept sending and receiving video with \'%1$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="call_is_ringing">Call is ringing</string>
<string name="call_is_on_hold">Call is on hold</string>
<string name="call_is_connected">Call is connected</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="replaces_not_supported">Peer does not support REPLACES feature</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_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="call_already_active">You already have an active call.</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="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 → baresip → Permissions → Storage.</string>
<string name="restart_request">Restart Request</string>
<string name="restored">Application data restored. baresip 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 → baresip → 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="airplane_mode">Airplane mode is ON. Turn it OFF to use mobile account.</string>
<string name="no_calls">baresip needs \"Microphone\" permission for voice calls.</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">No supported video cameras!</string>
<string name="no_network">No network connection!</string>
<string name="no_aec">No hardware Acoustic Echo Cancellation!</string>
<string name="audio_focus_denied">Audio focus denied!</string>
<string name="permissions_rationale">Permissions rationale</string>
<string name="audio_permissions">baresip needs \"Microphone\" permission for voice calls,
\"Nearby devices\" permission for Bluetooth microphone/speaker detection,
\"Notifications\" permission for posting notifications, and in Android 9
\"Storage\" permission for Backup/Restore operations.</string>
<string name="audio_and_video_permissions">baresip+ needs \"Microphone\" permission
for voice calls, \"Camera\" permission for video calls, \"Nearby devices\" permission
for Bluetooth microphone/speaker detection, \"Notifications\" permission for posting
notifications, and in Android 9 \"Storage\" permissions for Backup/Restore operations.</string>
<string name="call_recording_title">Call Recording</string>
<string name="call_recording_tip">If activated, new incoming and outgoing calls will be recorded.
Recordings can be played on Call Details page.</string>
<string name="microphone_title">Microphone</string>
<string name="microphone_tip">If activated during call, microphone is muted.</string>
<string name="speakerphone_title">Speakerphone</string>
<string name="speakerphone_tip">If activated, audio is played via device speakerphone.</string>
</resources>