1194 lines
549 KiB
XML
1194 lines
549 KiB
XML
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<?xml version="1.0" encoding="utf-8"?>
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<merger version="3" xmlns:ns1="http://schemas.android.com/tools"><dataSet aapt-namespace="http://schemas.android.com/apk/res-auto" config="main$Generated" generated="true" ignore_pattern="!.svn:!.git:!.ds_store:!*.scc:.*:<dir>_*:!CVS:!thumbs.db:!picasa.ini:!*~"><source path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res"/></dataSet><dataSet aapt-namespace="http://schemas.android.com/apk/res-auto" config="main" generated-set="main$Generated" ignore_pattern="!.svn:!.git:!.ds_store:!*.scc:.*:<dir>_*:!CVS:!thumbs.db:!picasa.ini:!*~"><source path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res"><file name="backward" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\backward.png" qualifiers="" type="drawable"/><file name="bg_button_selector" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\bg_button_selector.xml" qualifiers="" type="drawable"/><file name="bg_button_selector_2" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\bg_button_selector_2.xml" qualifiers="" type="drawable"/><file name="call" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\call.xml" qualifiers="" type="drawable"/><file name="callbutton" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\callbutton.png" qualifiers="" type="drawable"/><file name="callbutton2" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\callbutton2.png" qualifiers="" type="drawable"/><file name="calls" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\calls.xml" qualifiers="" type="drawable"/><file name="calls_large" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\calls_large.xml" qualifiers="" type="drawable"/><file name="calls_missed" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\calls_missed.xml" qualifiers="" type="drawable"/><file name="call_down_green" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\call_down_green.xml" qualifiers="" type="drawable"/><file name="call_down_red" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\call_down_red.xml" qualifiers="" type="drawable"/><file name="call_hold" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\call_hold.xml" qualifiers="" type="drawable"/><file name="call_missed_in" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\call_missed_in.xml" qualifiers="" type="drawable"/><file name="call_missed_out" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\call_missed_out.xml" qualifiers="" type="drawable"/><file name="call_small" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\call_small.xml" qualifiers="" type="drawable"/><file name="call_transfer" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\call_transfer.xml" qualifiers="" type="drawable"/><file name="call_transfer_execute" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\call_transfer_execute.xml" qualifiers="" type="drawable"/><file name="call_up_green" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\call_up_green.xml" qualifiers="" type="drawable"/><file name="call_up_red" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\call_up_red.xml" qualifiers="" type="drawable"/><file name="call_video" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\call_video.xml" qualifiers="" type="drawable"/><file name="call_video_new" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\call_video_new.xml" qualifiers="" type="drawable"/><file name="camera_front" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\camera_front.xml" qualifiers="" type="drawable"/><file name="camera_rear" path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\drawable\camera_rear.xml" qualifiers="" type="drawable"/><file name="circle" path="D:\Wor
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<![CDATA[
|
|||
|
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<h1>Ritosip library based SIP User Agent</h1>
|
|||
|
|
<p>Juha Heinanen <jh@tutpro.com></p>
|
|||
|
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<p>Version %1$s</p>
|
|||
|
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<h2>Usage Hints</h2>
|
|||
|
|
<ul>
|
|||
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<li>Check that default values in baresip\'s Settings meet your needs
|
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(touch item titles for help).</li>
|
|||
|
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<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>
|
|||
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<li>Touch on the dot leads directly to account configuration.</li>
|
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|
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<li>Swipe down gesture causes re-registration of the currently shown account.</li>
|
|||
|
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<li>Long touch on currently shown account enables or disables account\'s registration.</li>
|
|||
|
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<li>Swipe left/right gesture toggles between the accounts.</li>
|
|||
|
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<li>Previous call party can be reselect by touching the call icon when Callee is empty.</li>
|
|||
|
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<li>Peers of calls and messages can be added to contacts by long touches.</li>
|
|||
|
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<li>Long touches can also be used to remove calls, chats, messages, and contacts.</li>
|
|||
|
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<li>Touch/long touch of contact icon can be used to install/remove image avatar.</li>
|
|||
|
|
<li>See <a href="https://github.com/juha-h/baresip-studio/wiki">Wiki</a> for more
|
|||
|
|
information.</li>
|
|||
|
|
</ul>
|
|||
|
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<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.
|
|||
|
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<h2>Licenses</h2>
|
|||
|
|
<ul>
|
|||
|
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<li><b>BSD-3-Clause</b> except the following:</li>
|
|||
|
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<li><b>Apache 2.0</b> AMR codecs and TLS security</li>
|
|||
|
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<li><b>AGPLv4</b> ZRTP media encryption</li>
|
|||
|
|
<li><b>GNU LGPL 2.1</b> G.722, G.726, and Codec2 codecs</li>
|
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<li><b>GNU GPLv3</b> G.729 codec</li>
|
|||
|
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</ul>
|
|||
|
|
]]>
|
|||
|
|
</string><string name="about_text_plus">
|
|||
|
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<![CDATA[
|
|||
|
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<h1>Ritosip library based SIP User Agent with video calls</h1>
|
|||
|
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<p>Juha Heinanen <jh@tutpro.com></p>
|
|||
|
|
<p>Version %1$s</p>
|
|||
|
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<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
|
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|
|
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>
|
|||
|
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]]>
|
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|
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</string><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
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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$s\'</string><string name="authentication_username">Authentication Username</string><string name="authentication_username_help">Authentication username
|
|||
|
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if authentication of SIP requests is required. Default value is account\'s username.
|
|||
|
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</string><string name="invalid_authentication_username">Invalid Authentication Username \'%1$s\'</string><string name="authentication_password">Authentication Password</string><string name="authentication_password_help">Authentication
|
|||
|
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Password up to 64 characters. If Authentication Username is given, but Password is not
|
|||
|
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given, it will be asked when ritosip is started.
|
|||
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</string><string name="invalid_authentication_password">Invalid Authentication Password \'%1$s\'</string><string name="outbound_proxies">Outbound Proxies</string><string name="outbound_proxies_help">SIP URI of one or two proxies that must be used when sending requests.
|
|||
|
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If two is given, REGISTER requests are sent to both and other requests are sent to
|
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|
|
one that responds. If no outbound proxy is given, requests are sent based on
|
|||
|
|
DNS NAPTR/SRV/A record lookup of callee URI hostpart. If hostpart of SIP URI is an IPv6
|
|||
|
|
address, the address must be written inside brackets [].
|
|||
|
|
\nExamples:
|
|||
|
|
\n • sip:example.com:5061;transport=tls
|
|||
|
|
\n • sip:[2001:67c:223:777::10];transport=tcp
|
|||
|
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\n • sip:192.168.43.50:443;transport=wss
|
|||
|
|
</string><string name="sip_uri_of_proxy_server">SIP URI of Proxy Server</string><string name="sip_uri_of_another_proxy_server">SIP URI of another Proxy Server</string><string name="invalid_proxy_server_uri">Invalid Proxy Server URI \'%1$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$s\'</string><string name="media_nat">Media NAT Traversal</string><string name="media_nat_help">Selects media NAT traversal protocol (if any). Possible choices are STUN
|
|||
|
|
(Session Traversal Utilities for NAT, RFC 5389) and ICE (Interactive Connectivity
|
|||
|
|
Establishment, RFC 5245).
|
|||
|
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</string><string name="stun_server">STUN/TURN Server</string><string name="stun_server_help">A STUN/TURN Server URI of form scheme:host[:port][\?transport=udp|tcp],
|
|||
|
|
where scheme is \'stun\', \'stuns\', \'turn\', or \'turns\'. Factory default STUN Server
|
|||
|
|
for STUN and ICE protocols is \'stun:stun.l.google.com:19302\' pointing to public Google
|
|||
|
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STUN server. There is no factory default TURN server.
|
|||
|
|
</string><string name="stun_server_uri">STUN/TURN Server URI</string><string name="invalid_stun_server">Invalid STUN/TURN Server URI \'%1$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.
|
|||
|
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</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$s\'</string><string name="default_account">Default Account</string><string name="default_account_help">If checked, this account is selected when ritosip is started.
|
|||
|
|
</string><string name="accounts">Accounts</string><string name="new_account">New Account</string><string name="accounts_help">When a new account is created, account\'s port number and transport
|
|||
|
|
protocol may be optionally given: <user>@<domain>[:<port>][;transport=udp|tcp|tls].
|
|||
|
|
If <port> is given and transport protocol is not given, transport protocol defaults to udp.
|
|||
|
|
If <port> is not given and transport protocol is given, <port> defaults to 5060 or 5061 (TLS).
|
|||
|
|
If neither is given and no outbound proxy is specified, account\'s registrar (if any) is
|
|||
|
|
determined solely based on domain\'s DNS information.
|
|||
|
|
</string><string name="user_domain">user@domain</string><string name="invalid_aor">Invalid user@domain[:port][;transport=udp|tcp|tls] \'%1$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><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><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><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><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><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><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><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$s\' already exists.</string><string name="invalid_contact_uri">Invalid SIP URI</string><string name="android_contact_help">If checked, this contact is added to Android contacts.</string><string name="avatar_image">Profile image</string><string name="contacts">Contacts</string><string name="contact_action_question">Do you want to call or send message to \'%1$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><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><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 <%1$s>\?</string><string name="transfer_request">Transfer Request</string><string name="transfer_request_query">Do you accept to transfer this call to \'%1$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></file><file path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\values\styles.xml" qualifiers=""><style name="AppTheme" parent="Theme.MaterialComponents.DayNight">
|
|||
|
|
<item name="colorPrimary">@color/colorPrimary</item>
|
|||
|
|
<item name="colorPrimaryDark">@color/colorPrimaryDark</item>
|
|||
|
|
<item name="colorAccent">@color/colorAccent</item>
|
|||
|
|
<item name="actionBarTheme">@style/ActionBar</item>
|
|||
|
|
<item name="actionOverflowMenuStyle">@style/Spinner</item>
|
|||
|
|
<item name="android:dropDownListViewStyle">@style/Spinner</item>
|
|||
|
|
<item name="android:colorBackground">@color/colorBackground</item>
|
|||
|
|
<item name="android:windowBackground">@color/colorBackground</item>
|
|||
|
|
<item name="android:statusBarColor">@color/colorBackground</item>
|
|||
|
|
</style><style name="AppTheme.Main" parent="AppTheme">
|
|||
|
|
<item name="windowActionBar">false</item>
|
|||
|
|
<item name="windowNoTitle">true</item>
|
|||
|
|
</style><style name="ActionBar" parent="ThemeOverlay.MaterialComponents.Dark.ActionBar">
|
|||
|
|
<item name="android:colorBackground">@color/colorSecondaryDark</item>
|
|||
|
|
<item name="android:textColor">@color/colorLight</item>
|
|||
|
|
<item name="android:textColorPrimary">@color/colorLight</item>
|
|||
|
|
<item name="android:textColorSecondary">@color/colorLight</item>
|
|||
|
|
</style><style name="Spinner" parent="Widget.AppCompat.ListView.DropDown">
|
|||
|
|
<item name="android:popupBackground">@color/colorPopupBackground</item>
|
|||
|
|
<item name="android:textColor">@color/colorLight</item>
|
|||
|
|
<item name="android:dividerHeight">1dp</item>
|
|||
|
|
<item name="android:divider">@color/colorSpinnerDivider</item>
|
|||
|
|
</style><style name="AlertDialogTheme">
|
|||
|
|
<item name="buttonBarPositiveButtonStyle">@style/Alert.Button.Positive</item>
|
|||
|
|
<item name="buttonBarNegativeButtonStyle">@style/Alert.Button.Positive</item>
|
|||
|
|
<item name="buttonBarNeutralButtonStyle">@style/Alert.Button.Neutral</item>
|
|||
|
|
<item name="materialAlertDialogBodyTextStyle">@style/Alert.Message</item>
|
|||
|
|
<item name="materialAlertDialogTitleTextStyle">@style/Alert.Title</item>
|
|||
|
|
<item name="shapeAppearanceOverlay">@style/DialogCorners</item>
|
|||
|
|
<item name="android:textSize">14sp</item>
|
|||
|
|
</style><style name="DialogCorners">
|
|||
|
|
<item name="cornerFamily">rounded</item>
|
|||
|
|
<item name="cornerSize">16dp</item>
|
|||
|
|
</style><style name="Alert.Button.Positive" parent="Widget.Material3.Button.TextButton">
|
|||
|
|
<item name="android:textColor">@color/colorAlert</item>
|
|||
|
|
<item name="rippleColor">@color/colorAccent</item>
|
|||
|
|
<item name="android:textSize">14sp</item>
|
|||
|
|
<item name="android:textAllCaps">true</item>
|
|||
|
|
</style><style name="Alert.Button.Neutral" parent="Widget.Material3.Button.TextButton">
|
|||
|
|
<item name="backgroundTint">@android:color/transparent</item>
|
|||
|
|
<item name="rippleColor">@color/colorAccent</item>
|
|||
|
|
<item name="android:textColor">@color/colorGray</item>
|
|||
|
|
<item name="android:textSize">14sp</item>
|
|||
|
|
</style><style name="Alert.Title" parent="@style/MaterialAlertDialog.Material3.Title.Text">
|
|||
|
|
<item name="android:textColor">@color/colorAlert</item>
|
|||
|
|
<item name="android:textSize">20sp</item>
|
|||
|
|
</style><style name="Alert.Message" parent="@style/MaterialAlertDialog.Material3.Body.Text">
|
|||
|
|
<item name="android:textColor">@color/colorStrong</item>
|
|||
|
|
<item name="android:textSize">16sp</item>
|
|||
|
|
</style></file><file path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\values-bg\strings.xml" qualifiers="bg"><string name="about_title">Относно baresip</string><string name="about_text">
|
|||
|
|
<![CDATA[
|
|||
|
|
<h1>SES потребителско приложение, базирано на библиотеката Baresip</h1>
|
|||
|
|
<p>Juha Heinanen <jh@tutpro.com></p>
|
|||
|
|
<p>версия %1$s</p>
|
|||
|
|
<h2>Съвети за употреба</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li>Проверете дали стойностите по подразбиране на конфигурацията отговарят на вашите нужди (заглавия на елементите с докосване).</li>
|
|||
|
|
<li>След това създайте един или повече акаунти (отново докоснете заглавия на артикули за помощ).</li>
|
|||
|
|
<li>Участниците в обажданията и съобщенията могат да се добавят към контактите с по-продължителни докосвания.</li>
|
|||
|
|
<li>По същия начин може да се премахват повиквания, чатове, съобщения и контакти.</li>
|
|||
|
|
<li>Имате възможност да преизберете повторно последния абонат, като кликнете на иконата за повикване, когато полето е празно.</li>
|
|||
|
|
<li>Ако не можете да чуете другата страна по време на разговор, опитайте се да увеличите силата на звука на Media устройството си или задайте силата на звука на разговора по подразбиране в конфигурация.</li>
|
|||
|
|
</ul>
|
|||
|
|
<h2>Програмен код</h2>
|
|||
|
|
Изходният код е достъпен на адрес <a href="https://github.com/juha-h/baresip-studio">GitHub</a>,
|
|||
|
|
където също могат да се докладват проблеми.
|
|||
|
|
]]>
|
|||
|
|
</string><string name="account">Акаунт</string><string name="display_name">Показване на име</string><string name="your_name">Твоето име</string><string name="display_name_help">Публично име (ако има такова), използвано в URI на изходящи заявки.</string><string name="authentication_username">Потребителско име за удостоверяване</string><string name="authentication_username_help">Потребителско име за удостоверяване, ако се изисква от изходящ прокси.</string><string name="authentication_password">Парола за удостоверяване</string><string name="authentication_password_help">Парола за удостоверяване, ако се изисква от изходящ прокси.</string><string name="outbound_proxies">Изходящи прокси</string><string name="outbound_proxies_help">SIP URI на един или два прокси сървъра, които трябва да се използват при изпращане на заявки.
|
|||
|
|
Ако са дадени две, заявките за РЕГИСТРИРАНЕ се изпращат и до двете, и до други заявки
|
|||
|
|
тази, която отговаря. Ако не е даден изходящ прокси, заявките се изпращат въз основа на
|
|||
|
|
DNS NAPTR / SRV / Търсене на запис на хоризонталната част на URI на повикващия. Ако хостпарт на SIP URI е IPv6
|
|||
|
|
адрес, адресът трябва да бъде написан в скоби [].
|
|||
|
|
\nExamples:
|
|||
|
|
\n • sip:foo.com:5060;transport=tls
|
|||
|
|
\n • sip:[2001:67c:223:777::10]:5060;transport=tcp
|
|||
|
|
</string><string name="sip_uri_of_proxy_server">SIP URI на прокси сървър</string><string name="sip_uri_of_another_proxy_server">SIP URI на друг прокси сървър</string><string name="register">Регистрирай</string><string name="register_help">Ако е отметнато, регистрацията е активирана и заявките за РЕГИСТРИРАНЕ се изпращат на
|
|||
|
|
12 минути интервали. </string><string name="audio_codecs">Аудио кодеци</string><string name="audio_codecs_help">Списък на поддържаните аудио кодеци в приоритетен ред.</string><string name="media_nat">NAT Traversal Average</string><string name="media_nat_help">Избира протокол за преминаване на носител NAT (ако има такъв). Възможният избор е STUN
|
|||
|
|
(Помощни програми за сесионно преминаване за NAT, RFC 5389) и ICE (Интерактивна свързаност
|
|||
|
|
Учредяване, RFC 5245).
|
|||
|
|
</string><string name="stun_server">СТУН Сървър</string><string name="stun_server_help">STUN сървър на хост на формуляра [: port].
|
|||
|
|
Фабричната стойност по подразбиране е \'stun.l.google.com:19302\', сочещи към обществен сървър на Google STUN.
|
|||
|
|
В момента потребителското име и паролата не се поддържат. </string><string name="media_encryption">Шифроване на медиите</string><string name="media_encryption_help">Избира протокол за криптиране на медийния транспорт (ако има такъв).
|
|||
|
|
\n • ZRTP (препоръчително) означава, че след изпробване на преговорите за криптиране на медиите от край до край ZRTP
|
|||
|
|
разговорът е установен.
|
|||
|
|
\n • DTLS-SRTPF означава, че UDP / TLS / RTP / SAVPF се предлага при изходящо повикване и RTP / SAVP,
|
|||
|
|
RTP / SAVPF, UDP / TLS / RTP / SAVP или UDP / TLS / RTP / SAVPF се използва, ако се предлага при входящо повикване.
|
|||
|
|
\n • SRTP-MANDF означава, че RTP / SAVPF се предлага при изходящо повикване и се изисква при входящо повикване.
|
|||
|
|
\n • SRTP-MAND означава, че RTP / SAVP се предлага при изходящо повикване и се изисква при входящо повикване.
|
|||
|
|
\n • SRTP означава, че RTP / AVP се предлага при изходящо повикване и че RTP / SAVP или RTP / SAVPF се използва
|
|||
|
|
ако се предлага при входящо повикване.
|
|||
|
|
</string><string name="answer_mode">Режим на отговори</string><string name="answer_mode_help">Избира как се отговаря на входящите повиквания.</string><string name="manual">Ръчно</string><string name="auto">Автоматично</string><string name="voicemail_uri">URI на гласова поща</string><string name="voicemain_uri_help">SIP URI за проверка на гласови съобщения. Ако се остави празно, съобщенията на гласовата поща
|
|||
|
|
(Индикации за чакане на съобщение) не са активирани.
|
|||
|
|
</string><string name="default_account">Профил по подразбиране</string><string name="default_account_help">Ако е отметнато, този акаунт се избира при стартиране на baresip. </string><string name="accounts">Профили</string><string name="user_domain">потребител@домейн</string><string name="invalid_aor">Невалиден потребител@домейн \'%1$s\'</string><string name="account_exists">Профила \'%1$s\' вече съществува.</string><string name="account_allocation_failure">"Неуспешно разпределяне на нов акаунт.</string><string name="encrypt_password">Шифроване на парола</string><string name="decrypt_password">Дешифриране на парола</string><string name="delete_account">Искате ли да изтриете акаунта \'%1$s\'?</string><string name="answer">Отговор</string><string name="reject">Отхвърляне</string><string name="incoming_call_from">Обадете се от</string><string name="transfer_request">Заявка за прехвърляне на обаждане до</string><string name="message_from">Съобщение от</string><string name="call_history">История на обажданията</string><string name="call">Обади се</string><string name="calls_calls">повиквания</string><string name="calls_call">обади се</string><string name="calls_call_message_question">Искате ли да се обадите или да изпратите съобщение до \'%1$s\'?</string><string name="calls_add_delete_question">Искате ли да добавите \'%1$s\' към контакти или изтриване
|
|||
|
|
%2$s от историята на обажданията?
|
|||
|
|
</string><string name="calls_delete_question">Искате ли да изтриете \'%1$s\' %2$s от историята на обажданията? </string><string name="delete_history">Изтрий</string><string name="disable_history">Забрани история на обаждания</string><string name="enable_history">Разреши история на обаждания</string><string name="delete_history_alert">Искате ли да изтриете историята на обажданията на акаунта \'%1$s\'?</string><string name="chat">Съобщения в чата</string><string name="chat_with">Чатя с %1$s</string><string name="new_message">Ново съобщение</string><string name="long_message_question">Искате ли да изтриете съобщението или да добавите потребителя \'%1$s\' в контакти?</string><string name="short_message_question">Искате ли да изтриете съобщението?</string><string name="add_contact">Добави контакт</string><string name="sending_failed">Изпращането на съобщението не бе успешно</string><string name="message_failed">Неуспешно</string><string name="chats">История на чата</string><string name="today">днес</string><string name="you">Вие</string><string name="new_chat_peer">Нов потребител за чат</string><string name="invalid_chat_peer_uri">Невалиден SIP URI</string><string name="long_chat_question">Искате ли да изтриете чата с този потребител \'%1$s\' или
|
|||
|
|
да го добавяне в контактите?</string><string name="short_chat_question">Искате ли да изтриете чата с \'%1$s\'?</string><string name="delete_chats">Изтрий</string><string name="delete_chats_alert">Искате ли да изтриете историята на чата на акаунта \'%1$s\'?</string><string name="configuration">Конфигуриране</string><string name="start_automatically">Стартирайте автоматично</string><string name="start_automatically_help">Ако е отметнато, baresip се стартира автоматично след включване на устройството.</string><string name="listen_address">Слушане на адрес</string><string name="listen_address_help">IP адрес и порт на формата \'адрес: порт\', на който baresip слуша
|
|||
|
|
за входящи SIP заявки. Ако IP адресът е IPv6 адрес, той трябва да бъде написан вътре в скоби [].
|
|||
|
|
IPv4 адрес 0.0.0.0 или IPv6 адрес [::] позволява на baresip да слуша всички налични адреси.
|
|||
|
|
Ако се остави празно (фабрично по подразбиране), baresip ще слуша на порт 5060 от всички налични адреси.
|
|||
|
|
</string><string name="invalid_listen_address">Невалиден адрес за слушане</string><string name="dns_servers">DNS сървъри</string><string name="dns_servers_help">Списък разделен със запетая на адреси на DNS сървъри. Ако не е зададено,
|
|||
|
|
адресите на DNS сървъра се получават динамично от системата. Всеки DNS адрес е във форма \'ip:port\'
|
|||
|
|
или \'ip\'. Ако портът е пропуснат, той по подразбиране е 53. Ако ip е IPv6 адрес и също е зададен порт,
|
|||
|
|
ip трябва да бъде написани вътре в скоби []. Като пример, посочете \`8.8.8.8:53,[2001:4860:4860::8888]:53 \'
|
|||
|
|
сочи към IPv4 и IPv6 адреси на обществени DNS сървъри на Google.
|
|||
|
|
</string><string name="invalid_dns_servers">Невалидни DNS сървъри</string><string name="failed_to_set_dns_servers">Неуспешно задаване на DNS сървъри</string><string name="tls_certificate_file">Файл за сертификати TLS</string><string name="tls_certificate_file_help">Ако е поставена отметка, файл \'cert.pem\' съдържащ TLS сертификатът и личният ключ на този екземпляр baresip е бил или ще бъде зареден от директорията за Изтегляне. От съображения за сигурност изтрийте файла след зареждането му.</string><string name="tls_ca_file">TLS CA File</string><string name="tls_ca_file_help">Ако е поставено отметка, файл \'ca_certs.crt\' съдържащи TLS сертификати на Сертифициращите органи е бил или ще бъде зареден от директорията за Изтегляне.</string><string name="aec">Акустично отменяне на ехото</string><string name="aec_help">Ако е поставена отметка, се прави опит за отмяна на ехо при аудио повикване.</string><string name="opus_bit_rate">Opus Bit Rate</string><string name="opus_bit_rate_help">Средна максимална битова скорост, използвана от аудио поток на Opus.
|
|||
|
|
Валидните стойности са 6000-510000. Фабрично по подразбиране е 28000.
|
|||
|
|
</string><string name="opus_packet_loss">Очаквана загуба на пакети Opus</string><string name="opus_packet_loss_help">Очакван процент загуба на пакет от аудио поток на Opus, от 0–100.
|
|||
|
|
По подразбиране 0 изключване на Opus Forward Error Correction (FEC).
|
|||
|
|
</string><string name="invalid_opus_bitrate">Невалиден битрейт на Opus</string><string name="invalid_opus_packet_loss">Невалиден процент загуба на пакет Opus</string><string name="default_call_volume">Усилване на повикването по подразбиране</string><string name="default_call_volume_help">Ако се задава, трябва да се избира сила на звука при
|
|||
|
|
повикване по подразбиране в мащаб 1–10.
|
|||
|
|
</string><string name="debug">Debug</string><string name="debug_help">Осигурява наличие на съобщения за грешки и информация на ниво информация в Logcat.</string><string name="reset_config">Възстановяване на фабричните настройки</string><string name="reset_config_help">Ако е отметнато, конфигурацията се нулира до фабрични стойности по подразбиране</string><string name="read_cert_error">Неуспешно четене на файл \'cert.pem\' от директорията за изтегляне.</string><string name="read_ca_certs_error">Файлът не беше прочетен \'ca_certs.crt\' от директорията за изтегляне.</string><string name="config_restart">Трябва да рестартирате baresip, за да активирате новата конфигурация. Рестартирай сега?</string><string name="contact">контакт</string><string name="new_contact">Нов контакт</string><string name="contact_name">име</string><string name="invalid_contact">Невалидно име за контакт \'%1$s\'</string><string name="contact_already_exists">Този контакт \'%1$s\' вече съществува.</string><string name="invalid_contact_uri">Невалиден SIP URI</string><string name="contacts">Контакти</string><string name="contact_action_question">Искате ли да се обадите или да изпратите съобщение до \'%1$s\'?</string><string name="send_message">Изпрати съобщение</string><string name="contact_delete_question">Искате ли да изтриете този контакт \'%1$s\'?</string><string name="contacts_exceeded">Вашият максимален брой контакти %1$d е надвишен.</string><string name="alert">Внимание</string><string name="info">Информация</string><string name="notice">известие</string><string name="cancel">Отказ</string><string name="ok">Добре</string><string name="yes">да</string><string name="no">не</string><string name="accept">приемам</string><string name="deny">отказвам</string><string name="user_id">Потребителско Име</string><string name="password">Парола</string><string name="add">Добави</string><string name="delete">Изтрий</string><string name="edit">Редактиране</string><string name="send">Изпращам</string><string name="status">Статус</string><string name="error">Грешка</string><string name="backup">Създай Резервно копие</string><string name="restore">Възстанови от Резервно копие</string><string name="about">Относно</string><string name="restart">Рестартирай</string><string name="quit">Отписване</string><string name="outgoing_call_to_dots">Изходящо повикване до…</string><string name="incoming_call_from_dots">Входящо обаждане от…</string><string name="transferring_call_to_dots">Прехвърляне на обаждане до…</string><string name="invalid_sip_uri">Невалиден SIP URI \'%1$s\'</string><string name="callee">Набиране</string><string name="hangup">Затвори</string><string name="hold">Задържане</string><string name="dtmf">DTMF</string><string name="call_info">Информация за обаждане</string><string name="duration">Продължителност %1$d (s)</string><string name="codecs">Кодеци</string><string name="rate">Скорост: %1$s</string><string name="voicemail">Гласова поща</string><string name="voicemail_messages">Съобщения за гласова поща</string><string name="you_have">Ти имаш</string><string name="one_new_message">едно ново съобщение</string><string name="new_messages">Нови съ<D181>
|
|||
|
|
Те са нулирани. Рестартирайте baresip.
|
|||
|
|
</string><string name="registering_failed">Регистрация на \`%1$s\` се провали.</string><string name="verify">Потвърди</string><string name="verify_sas">Искате ли да потвърдите <%1$s>\?</string><string name="transfer_request_query">Приемате ли да прехвърлите обаждане до \'%1$s\'?</string><string name="call_failed">Обаждането не бе успешно</string><string name="call_closed">Обаждането е затворено</string><string name="call_not_secure">Това обаждане НЕ е сигурно!</string><string name="peer_not_verified">Това обаждане е СИГУРНО, но другият абонат НЕ е потвърдил!</string><string name="call_is_secure">Този разговор е СИГУРЕН и другият абонат е ПРОВЕРЕН! Искате ли да го потвърдите? </string><string name="unverify">Отмяна на потвърждението</string><string name="backed_up">Данните от приложението са архивирани във файла за изтегляне на папки \'%1$s\'.</string><string name="backup_failed">Неуспешно архивиране на данни от приложението за изтегляне на файла с папки \'%1$s\'. Проверете Приложения → baresip → Разрешения → Съхранение</string><string name="restored">Данните за приложението са възстановени. baresip трябва да се рестартира. Рестартирай сега?</string><string name="restore_failed">Възстановяването на данните на приложението от папката за изтегляне не бе успешно. Проверете Приложения → baresip → Разрешения → Съхранение и този архивен файл \'%1$s\' съществува в папката и ако е така, вие сте дали правилна парола за дешифриране.</string><string name="audio_modules_title">Аудио модули</string><string name="audio_modules_help">Аудио кодеци предоставени от проверените модули могат да бъдат използвани от акаунтите.</string><string name="failed_to_load_module">Не могат да бъдат заредени модулите.</string><string name="no_calls">Разговори не могат да бъдат осъществени без разрешен достъп до микрофона.</string><string name="accounts_help">Порта и транспортния протокол на акаунта могат да бъдат посочени по избор, когато се създаде нов акаунт: потребителско име@домейн[:port] [;transport=udp|tcp|tls]. Ако е даден порт и транспортният протокол не е даден, транспортният протокол по подразбиране е udp. Ако порт не е даден и транспортният протокол е даден, портът по подразбиране е 5060 или 5061 (tls). Ако нито едното не е посочено и не е посочен изходящ прокси, регистраторът на акаунта (ако има такъв) се определя единствено въз основа на DNS информация от домейна.</string><string name="new_account">Нов Акаунт</string></file><file path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\values-cs\strings.xml" qualifiers="cs"><string name="about_title">O aplikaci baresip</string><string name="about_title_plus">O aplikaci baresip+</string><string name="authentication_username">Uživatelské jméno pro ověřování</string><string name="invalid_stun_username">Neplatné u<>
|
|||
|
|
<h1>SIP klient založený na knihovně Baresip s podporou video hovorů</h1>
|
|||
|
|
<p>Juha Heinanen <jh@tutpro.com></p>
|
|||
|
|
<p>Verze %1$s</p>
|
|||
|
|
<h2>Nápověda k používání</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li>Zkontrolujte, zda výchozí hodnoty v nastavení baresipu vyhovují vašim potřebám
|
|||
|
|
(nápovědu získáte ťuknutím na názvy položek).</li>
|
|||
|
|
<li>Poté v sekci Účty vytvořte jeden nebo více účtů (pro nápovědu opět ťukněte na názvy položek).</li>
|
|||
|
|
<li>Stav registrace účtu je indikován barevnou tečkou: zelená (registrace
|
|||
|
|
proběhla úspěšně), žlutá (registrace probíhá), červená (registrace se nezdařila), bílá (registrace
|
|||
|
|
nezačala).</li>
|
|||
|
|
<li>Ťuknutí na tečku spustí konfiguraci účtu.</li>
|
|||
|
|
<li>Gesto přejetí prstem dolů zahájí opětovnou registraci aktuálně zobrazeného účtu.</li>
|
|||
|
|
<li>Dlouhým dotykem na aktuálně zobrazený účet povolíte nebo zakážete jeho registraci.</li>
|
|||
|
|
<li>Mezi účty se přepíná gestem přejetí vlevo/vpravo.</li>
|
|||
|
|
<li>Naposledy volaný kontakt lze znovu vybrat dotykem na ikonu hovoru, pokud je pole Číslo nebo adresa prázdné.</li>
|
|||
|
|
<li>Účastníky hovorů a zpráv lze přidávat do kontaktů dlouhými dotyky.</li>
|
|||
|
|
<li>Dlouhými dotyky lze také odstraňovat hovory, chaty, zprávy a kontakty.</li>
|
|||
|
|
<li>Dotyk/dlouhý dotyk na ikonu kontaktu lze použít k přidání/odstranění obrázkového avatara.</li>
|
|||
|
|
<li>Navštivte <a href=https://github.com/juha-h/baresip-studio/wiki>Wiki</a> pro více
|
|||
|
|
informací.</li>
|
|||
|
|
</ul>
|
|||
|
|
<h2>Známé problémy</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li>Při videohovorech je třeba držet zařízení v režimu na šířku
|
|||
|
|
otočeném o 90 stupňů doleva oproti orientaci na výšku.</li>
|
|||
|
|
<li>Vlastní náhled volajícího se nezobrazuje správně, když je videoproud pouze v režimu odesílání (sendonly).</li>
|
|||
|
|
</ul>
|
|||
|
|
<h2>Zásady ochrany osobních údajů</h2>
|
|||
|
|
Zásady ochrany osobních údajů jsou k dispozici <a href=https://raw.githubusercontent.com/juha-h/baresip-studio/video/PrivacyPolicy.txt>zde</a>.
|
|||
|
|
<h2>Zdrojový kód</h2>
|
|||
|
|
Zdrojový kód je k dispozici na <a href=https://github.com/juha-h/baresip-studio>GitHubu</a>,
|
|||
|
|
kde lze také hlásit problémy.
|
|||
|
|
<h2>Licence</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li><b>BSD-3-Clause</b> kromě následujících:</li>
|
|||
|
|
<li><b>Apache 2.0</b> kodeky AMR a zabezpečení TLS</li>
|
|||
|
|
<li><b>AGPLv4</b> šifrování médií ZRTP</li>
|
|||
|
|
<li><b>GNU LGP 2.1</b> G.722, G.726, a Codec2 kodeky</li>
|
|||
|
|
<li><b>GNU GPLv3</b> kodek G.729</li>
|
|||
|
|
<li><b>GNU GPLv2</b> H.264 a H.265 kodeky</li>
|
|||
|
|
<li><b>AOMedia</b> AV1 kodek</li>
|
|||
|
|
</ul>
|
|||
|
|
]]></string><string name="about_text"><![CDATA[
|
|||
|
|
<h1>SIP klient založený na knihovně Baresip</h1>
|
|||
|
|
<p>Juha Heinanen <jh@tutpro.com></p>
|
|||
|
|
<p>Verze %1$s</p>
|
|||
|
|
<h2>Nápověda</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li>Zkontrolujte, zda výchozí hodnoty v nastavení baresipu vyhovují vašim potřebám
|
|||
|
|
(nápovědu získáte ťuknutím na názvy položek).</li>
|
|||
|
|
<li>Poté v sekci Účty vytvořte jeden nebo více účtů (pro nápovědu opět ťukněte na názvy položek).</li>
|
|||
|
|
<li>Stav registrace účtu je indikován barevnou tečkou: zelená (registrace
|
|||
|
|
proběhla úspěšně), žlutá (registrace probíhá), červená (registrace se nezdařila), bílá (registrace
|
|||
|
|
nezačala).</li>
|
|||
|
|
<li>Ťuknutí na tečku spustí konfiguraci účtu.</li>
|
|||
|
|
<li>Gesto přejetí prstem dolů zahájí opětovnou registraci aktuálně zobrazeného účtu.</li>
|
|||
|
|
<li>Dlouhým dotykem na aktuálně zobrazený účet povolíte nebo zakážete jeho registraci.</li>
|
|||
|
|
<li>Mezi účty se přepíná gestem přejetí vlevo/vpravo.</li>
|
|||
|
|
<li>Naposledy volaný kontakt lze znovu vybrat dotykem na ikonu hovoru, pokud je pole Číslo nebo adresa prázdné.</li>
|
|||
|
|
<li>Účastníky hovorů a zpráv lze přidávat do kontaktů dlouhými dotyky.</li>
|
|||
|
|
<li>Dlouhými dotyky lze také odstraňovat hovory, chaty, zprávy a kontakty.</li>
|
|||
|
|
<li>Dotyk/dlouhý dotyk na ikonu kontaktu lze použít k přidání/odstranění obrázkového avatara.</li>
|
|||
|
|
<li>Navštivte <a href=https://github.com/juha-h/baresip-studio/wiki>Wiki</a>
|
|||
|
|
pro více informací.</li>
|
|||
|
|
</ul>
|
|||
|
|
<h2>Zásady ochrany osobních údajů</h2>
|
|||
|
|
Zásady ochrany osobních údajů jsou k dispozici <a href=https://raw.githubusercontent.com/juha-h/baresip-studio/master/PrivacyPolicy.txt>zde</a>.
|
|||
|
|
<h2>Zdrojový kód</h2>
|
|||
|
|
Zdrojový kód je k dispozici na <a href=https://github.com/juha-h/baresip-studio>GitHubu</a>,
|
|||
|
|
kde lze také hlásit problémy.
|
|||
|
|
<h2>Licence</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li><b>BSD-3-Clause</b> kromě následujících:</li>
|
|||
|
|
<li><b>Apache 2.0</b> AMR codecs a TLS security</li>
|
|||
|
|
<li><b>AGPLv4</b> šifrování médií ZRTP</li>
|
|||
|
|
<li><b>GNU LGPL 2.1</b> kodeky G.722, G.726 a Codec2</li>
|
|||
|
|
<li><b>GNU GPLv3</b> kodek G.729 </li>
|
|||
|
|
</ul>
|
|||
|
|
]]></string><string name="account_nickname">Přezdívka účtu</string><string name="account_nickname_help">Přezdívka (pokud existuje) používaná k identifikaci tohoto účtu v aplikaci baresip.</string><string name="nickname">Přezdívka</string><string name="authentication_password">Heslo pro ověření</string><string name="outbound_proxies">Odchozí proxy servery</string><string name="register">Registrace</string><string name="invalid_proxy_server_uri">Neplatná URI proxy serveru \'%1$s\'</string><string name="sip_uri_of_proxy_server">SIP URI proxy serveru</string><string name="outbound_proxies_help">SIP URI jednoho nebo dvou proxy serverů, které je třeba použít při odesílání požadavků. Pokud jsou zadány dva, jsou požadavky REGISTER zasílány oběma a ostatní požadavky jsou zasílány tomu, který odpoví. Pokud není zadán žádný odchozí proxy server, jsou požadavky odesílány na základě vyhledávání záznamů DNS NAPTR/SRV/A z hostitelského jména URI volaného. Pokud je hostitelské jméno URI SIP adresa IPv6, musí být adresa zapsána v závorkách [].
|
|||
|
|
\nPříklady:
|
|||
|
|
\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_another_proxy_server">SIP URI dalšího proxy serveru</string><string name="reg_int_help">Udává, jak často (v sekundách) baresip odesílá požadavky REGISTER. Platné hodnoty jsou od 60 do 3600.</string><string name="invalid_reg_int">Neplatný interval registrace\'%1$s\'</string><string name="register_help">Pokud je zatrženo, registrace je povolena a požadavky REGISTER jsou odesílány v intervalu určeném parametrem Interval registrace.</string><string name="reg_int">Interval registrace</string><string name="answer_mode_help">Vybírá způsob přijímání příchozích hovorů.</string><string name="media_nat">průchod médií skrze NAT</string><string name="invalid_stun_server">Neplatná URI serveru STUN/TURN \'%1$s\'</string><string name="media_nat_help">Vybírá protokol pro průchod médií skrze NAT (pokud existuje). Možné volby jsou STUN (Session Traversal Utilities for NAT, RFC 5389) a ICE (Interactive Connectivity Establishment, RFC 5245).</string><string name="stun_server">Server STUN/TURN</string><string name="stun_server_uri">URI serveru STUN/TURN</string><string name="stun_username_help">Uživatelské jméno, pokud ho server STUN/TURN vyžaduje</string><string name="stun_server_help">URI serveru STUN/TURN ve tvaru scheme:host[:port][\?transport=udp|tcp], kde scheme je \"stun\", \"stuns\", \"turn\" nebo \"turns\". Výchozí server STUN pro protokoly STUN a ICE je \'stun:stun.l.google.com:19302\', který ukazuje na veřejný server STUN společnosti Google. Výchozí server TURN neexistuje.</string><string name="stun_password">Heslo STUN/TURN</string><string name="stun_password_help">Heslo, pokud ho server STUN/TURN vyžaduje</string><string name="invalid_stun_password">Neplatné heslo \'%1$s\'</string><string name="media_encryption">Šifrování médií</string><string name="rtcp_mux_help">Pokud je zatrženo, pakety RTP a RTCP jsou multiplexovány na jednom portu (RFC 5761).</string><string name="rtcp_mux">Multiplexování RTCP</string><string name="authentication_username_help">Uživatelské jméno pro ověřování, pokud je vyžadováno ověřování požadavků SIP. Výchozí hodnotou je uživatelské jméno účtu.</string><string name="telephony_provider">Poskytovatel telefonních služeb</string><string name="time">Čas</string><string name="delete_chats">Odstranit</string><string name="display_name">Zobrazované jméno</string><string name="display_name_help">Název (pokud existuje) použitý v URI odchozích požadavků.</string><string name="invalid_authentication_username">Neplatné uživatelské jméno pro ověřování \'%1$s\'</string><string name="authentication_password_help">Ověřovací heslo až 64 znaků. Pokud je zadáno Uživatelské jméno pro ověřování, ale není zadáno Heslo, bude po spuštění baresipu požadováno.</string><string name="peer">Účastník</string><string name="direction">Směr</string><string name="calls_call_message_question">Chcete zavolat nebo odeslat zprávu \'%1$s\'\?</string><string name="calls_add_delete_question">Chcete přidat \'%1$s\' do kontaktů nebo odstranit %2$s z historie hovorů\?</string><string name="delete_history_alert">Chcete odstranit historii volání účtu \"%1$s\"\?</string><string name="calls_duration">Trvání</string><string name="long_chat_question">Chcete smazat chat s účastníkem \'%1$s\' nebo přidat účastníka do kontaktů\?</string><string name="long_message_question">Chcete odstranit zprávu nebo přidat účastníka \'%1$s\' do kontaktů\?</string><string name="disable_history">Zakázat</string><string name="chat">Zprávy chatu</string><string name="new_message">Nová zpráva</string><string name="short_message_question">Chcete zprávu odstranit\?</string><string name="calls_delete_question">Chcete odstranit \'%1$s\' %2$s z historie hovorů\?</string><string name="delete_history">Smazat</string><string name="chat_with">Chatovat s %1$s</string><string name="sending_failed">Odeslání zprávy se nezdařilo</string><string name="you">Vy
|
|||
|
|
\n • ZRTP (doporučeno) znamená, že se po navázání hovoru vyzkouší vyjednávání o koncové šifrování médií ZRTP.
|
|||
|
|
\n • DTLS-SRTPF znamená, že v odchozím hovoru je nabízen UDP/TLS/RTP/SAVPF a že v případě příchozího hovoru je použit RTP/SAVP, RTP/SAVPF, UDP/TLS/RTP/SAVP nebo UDP/TLS/RTP/SAVPF.
|
|||
|
|
\n • SRTP-MANDF znamená, že RTP/SAVPF je nabízen v odchozím hovoru a vyžadován v příchozím hovoru.
|
|||
|
|
\n • SRTP-MAND znamená, že RTP/SAVP je nabízen v odchozím hovoru a vyžadován v příchozím hovoru.
|
|||
|
|
\n • SRTP znamená, že RTP/AVP je nabízen v odchozím hovoru a že RTP/SAVP nebo RTP/SAVPF je použit, pokud je nabízen v příchozím hovoru.</string><string name="prefer_ipv6_media">Preferovat média IPv6</string><string name="dtmf_mode_help">Vybírá způsob odesílání tónů DTMF 0-9, #, * a A-D.</string><string name="prefer_ipv6_media_help">Pokud je zatrženo, nabídne použití protokolu medií IPv6 (je-li k dispozici), pokud protokol medií partnera nelze určit automaticky.</string><string name="voicemail_uri">URI hlasové schránky</string><string name="dtmf_mode">Režim DTMF</string><string name="answer_mode">Režim odpovědi</string><string name="manual">Ručně</string><string name="auto">Automaticky</string><string name="country_code">Kód země</string><string name="voicemain_uri_help">SIP URI pro kontrolu zpráv hlasové pošty. Pokud je ponecháno prázdné, zprávy hlasové pošty (indikace čekající zprávy) nejsou přihlášeny k odběru.</string><string name="invalid_voicemail_uri">Neplatné URI hlasové schránky \'%1$s\'</string><string name="country_code_hint">+kód</string><string name="country_code_help">Kód země E.164 tohoto účtu. Pokud uživatelská část \"Od\" v URI příchozího hovoru nebo zprávy obsahuje telefonní číslo, které nezačíná znakem \"+\", a pokud se vyhledání kontaktu nezdaří, je číslo předřazeno tomuto kódu země a vyhledání kontaktu je zopakováno. Pokud telefonní číslo začíná jedinou číslicí \"0\", číslice \"0\" se před prefixací čísla odstraní.</string><string name="default_account">Výchozí účet</string><string name="invalid_country_code">Neplatný kód země \'%1$s\'</string><string name="telephony_provider_help">Hostitelská část SIP URI používaná při volání na telefonní čísla. Výchozí tovární nastavení je doména účtu. Pokud není zadána, nelze tento účet použít pro volání na telefonní čísla.</string><string name="default_account_help">Pokud je zatrženo, je tento účet vybrán při spuštění baresipu.</string><string name="telephony_provider_hint">Hostitelská část SIP URI</string><string name="invalid_sip_uri_hostpart">Neplatná část hostitelského URI SIP \'%1$s\'</string><string name="new_account">Nový účet</string><string name="accounts">Účty</string><string name="account_exists">Účet \'%1$s\' již existuje.</string><string name="decrypt_password">Dešifrovat heslo</string><string name="accounts_help">Při vytváření nového účtu lze volitelně zadat číslo portu účtu a přenosový protokol: <uživatel>@<doména>[:<port>][;transport=udp|tcp|tls]. Pokud je zadán <port> a transportní protokol není zadán, je výchozí transportní protokol udp. Pokud není zadán <port> a je zadán transportní protokol, je výchozí hodnota <port> 5060 nebo 5061 (TLS). Pokud není zadán ani jeden z těchto parametrů a není zadán žádný odchozí proxy server, je registrátor účtu (pokud existuje) určen pouze na základě DNS informací o doméně.</string><string name="answer">Odpovědět</string><string name="reject">Odmítnout</string><string name="account_allocation_failure">Nepodařilo se přiřadit nový účet.</string><string name="delete_account">Chcete odstranit účet \'%1$s\'\?</string><string name="incoming_call_from">Příchozí hovor od</string><string name="missed_call_from">Zmeškaný hovor od</string><string name="missed_calls">Zmeškané hovory</string><string name="call">Hovor</string><string name="transfer_request_to">Žádost o přepojení hovoru na</string><string name="enable_history">Povolit</string><string name="message_from">Zpráva od</string><string name="call_history">Historie volání</string><string name="calls_calls">volání</string><string name="call_details">Podrobnosti o volání</string><string name="calls_call">volání</string><string name="tls_ca_file_help">Pokud je zaškrtnuto, byl nebo bude načten soubor obsahující certifikáty TLS takových certifikačních autorit, které nejsou součástí operačního syst<73>
|
|||
|
|
\nBeispiele:
|
|||
|
|
\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="stun_server_help">Ein STUN/TURN Server URI in der Form: schema:host[:port][?transport=udp|tcp], wobei schema \'stun\', \'stuns\', \'turn\', oder \'turns\' ist. Voreingestellter STUN Server für STUN und ICE Protokoll ist \'stun:stun.l.google.com:19302\' der zum öffentlichen Google STUN Server verweist. Es gibt keinen voreingestellten TURN Server.</string><string name="media_encryption_help">Wählt eine (oder keine) Medientransportverschlüsselung.
|
|||
|
|
\n • ZRTP (recommended) means that ZRTP end-to-end media encryption negotiation is tried afterthe 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 usedif offered in incoming call.</string><string name="start_failed">Baresip konnte nicht gestartet werden. Das kann wegen eines ungültigen Einstellungswertes sein. Überprüfen Sie die Hören (Listen) Adresse, TLS Zertifikat, TLS CA File. Starten Sie dann baresip neu.</string><string name="audio_permissions">baresip benötigt \"Mikrofon\"-Berechtigung für Sprachanrufe, \"Geräte in der Nähe\"-Berechtigung für Bluetooth-Mikrofon/Hörer-Erkennung und \"Benachrichtigungen\"-Berechtigung für die Benachrichtigungen.</string><string name="about_text"><h1>SIP Endgerät, basierend auf der Baresip Bibliothek </h1>
|
|||
|
|
\n <p>Juha Heinanen &lt;jh@tutpro.com&gt;</p> <p>Version %1$s</p>
|
|||
|
|
\n<h2>Benutzerhinweise</h2>
|
|||
|
|
\n <ul><li>Überprüfen Sie, dass die Einstellungen in baresip Ihren Bedürfnissen entsprechen(tippen Sie auf die Überschriften für Hilfe).</li>
|
|||
|
|
\n <li>In Konten, erstellen sie mindestens eines (tippen Sie auf die einzelnen Punkte für Hilfe).</li>
|
|||
|
|
\n <li>Der Registrierungsstatus eines Kontos wird mit einem farbigen Punkt markiert: grün (Registrierung erfolgreich), gelb (Registrierung im Gange), rot (Registrierung fehlgeschlagen), weiß (Konto ist nicht aktiviert).</li>
|
|||
|
|
\n<li>Tippen auf den Punkt öffnet direkt die Kontokonfiguration.</li>
|
|||
|
|
\n <li>Nach-unten-wischen registriert das aktuelle Konto erneut.</li>
|
|||
|
|
\n<li>Langes Drücken auf das gegenwärtige Konto aktiviert oder deaktiviert die Registrierung.</li>
|
|||
|
|
\n <li>Links/rechts-Wischen schaltet durch die verschiedenen Konten.</li>
|
|||
|
|
\n <li>Der letzte Anrufer kann erneut gewählt werden durch Berühren des Anruf-Symbols, wenn das Anruffeld leer ist.</li>
|
|||
|
|
\n<li>Anrufer und Absender von Nachrichten können zu den Kontakten hinzugefügt werden durch langes Drücken.</li>
|
|||
|
|
\n<li>Langes Drücken kann ebenfalls verwendet werden um Anrufe, Chats, Nachrichten und Kontakte zu löschen.</li>
|
|||
|
|
\n<li>Drücken/langes Drücken des Kontakte-Bildes kann benutzt werden um das Bild zu installieren/entfernen.</li>
|
|||
|
|
\n<li>See <a href=https://github.com/juha-h/baresip-studio/wiki>Wiki</a> for more information (in english).</li></ul>
|
|||
|
|
\n<h2>Privacy Policy</h2>
|
|||
|
|
\n Privacy policy is available <a href=https://raw.githubusercontent.com/juha-h/baresip-studio/master/PrivacyPolicy.txt>here</a>.
|
|||
|
|
\n<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.
|
|||
|
|
\n<h2>Lizenzen</h2>
|
|||
|
|
\n<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>SIP Endgerät mit Videoanrufen, basierend auf der Baresip Bibliothek </h1>
|
|||
|
|
|
|||
|
|
<p>Juha Heinanen <jh@tutpro.com></p>
|
|||
|
|
|
|||
|
|
<p>Version %1$s</p>
|
|||
|
|
|
|||
|
|
<h2>Benutzerhinweise</h2>
|
|||
|
|
|
|||
|
|
<ul>
|
|||
|
|
|
|||
|
|
<li>Überprüfen Sie, dass die Einstellungen in baresip+ Ihren Bedürfnissen entsprechen
|
|||
|
|
|
|||
|
|
(tippen Sie auf die Überschriften für Hilfe).</li>
|
|||
|
|
|
|||
|
|
<li>In Konten, erstellen sie mindestens eines (tippen Sie auf die einzelnen Punkte für Hilfe).</li>
|
|||
|
|
|
|||
|
|
<li>Der Registrierungsstatus eines Kontos wird mit einem farbigen Punkt markiert: grün (Registrierung
|
|||
|
|
|
|||
|
|
erfolgreich), gelb (Registrierung im Gange), rot (Registrierung fehlgeschlagen), weiß (Konto
|
|||
|
|
|
|||
|
|
ist nicht aktiviert).</li>
|
|||
|
|
|
|||
|
|
<li>Tippen auf den Punkt öffnet direkt die Kontokonfiguration.</li>
|
|||
|
|
|
|||
|
|
<li>Nach-unten-wischen registriert das aktuelle Konto erneut.</li>
|
|||
|
|
|
|||
|
|
<li>Langes Drücken auf das gegenwärtige Konto aktiviert oder deaktiviert die Registrierung.</li>
|
|||
|
|
|
|||
|
|
<li>Links/rechts-Wischen schaltet durch die verschiedenen Konten.</li>
|
|||
|
|
|
|||
|
|
<li>Der letzte Anrufer kann erneut gewählt werden durch Berühren des Anruf-Symbols, wenn das Anruffeld leer ist.</li>
|
|||
|
|
|
|||
|
|
<li>Anrufer und Absender von Nachrichten können zu den Kontakten hinzugefügt werden durch langes Drücken.</li>
|
|||
|
|
|
|||
|
|
<li>Langes Drücken kann ebenfalls verwendet werden um Anrufe, Chats, Nachrichten und Kontakte zu löschen.</li>
|
|||
|
|
|
|||
|
|
<li>Drücken/langes Drücken des Kontakte-Bildes kann benutzt werden um das Bild zu installieren/entfernen.</li>
|
|||
|
|
|
|||
|
|
<li>Hier: <a href="https://github.com/juha-h/baresip-studio/wiki">Wiki</a> gibt es mehr
|
|||
|
|
|
|||
|
|
Informationen (auf English).</li>
|
|||
|
|
|
|||
|
|
</ul>
|
|||
|
|
|
|||
|
|
<h2>Bekannte Probleme</h2>
|
|||
|
|
|
|||
|
|
<ul>
|
|||
|
|
|
|||
|
|
<li>Bei Videoanrufen muß das Gerät quer gehalten werden,
|
|||
|
|
|
|||
|
|
um 90 Grad nach links geneigt, von der Hochkantposition betrachtet.</li>
|
|||
|
|
|
|||
|
|
<li>Das eigene Bild wird nicht richtig gezeigt, wenn Video auf Nursenden eingestellt ist.</li>
|
|||
|
|
|
|||
|
|
</ul>
|
|||
|
|
|
|||
|
|
<h2>Datenschutzerklärung</h2>
|
|||
|
|
|
|||
|
|
Die Datenschutzerklärung ist <a href="https://raw.githubusercontent.com/juha-h/baresip-studio/video/PrivacyPolicy.txt">hier</a> abrufbar.
|
|||
|
|
|
|||
|
|
<h2>Quellcode</h2>
|
|||
|
|
|
|||
|
|
Der Quellcode wird auf <a href="https://github.com/juha-h/baresip-studio">GitHub</a> bereitgestellt,
|
|||
|
|
|
|||
|
|
wo auch Fehler gemeldet werden können.
|
|||
|
|
|
|||
|
|
<h2>Lizenzen</h2>
|
|||
|
|
|
|||
|
|
<ul>
|
|||
|
|
|
|||
|
|
<li><b>BSD-3-Clause</b>, außer bei folgendem:</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><string name="country_code_help">E.164 Ländercode dieses Kontos. Wenn der Benutzerteil der Von-URI des eingehenden Anrufs oder der Nachricht eine Telefonnummer enthält, die nicht mit \'+\'-Zeichen beginnt und wenn der Kontakt nicht bekannt ist, wird der Nummer dieser Ländercode vorangestellt und die Kontaktsuche beginnt erneut. Wenn die Telefonnummer mit einer einzigen Ziffer \'0\' beginnt, wird die Zahl \'0\' entfernt, bevor der Ländercode vorangestellt wird.</string><string name="accounts_help">Wenn ein neues Konto erstellt wird, kann die Portnummer und das Transportprotokoll des Kontos optional angegeben werden: <user>@<domain>[:<port>][;transport=udp|tcp|tls]. Wird <port> angegeben und das Transportprotokoll nicht angegeben, wird das Transportprotokoll standardmäßig auf udp gesetzt. Wenn <port> nicht und das Transportprotokoll schon angegeben ist, wird <port> 5060 oder 5061 eingestellt (TLS). Wenn weder noch angegeben wird und kein Proxy für ausgehende Verbindungen angegeben wird, wird der Registrar (falls vorhanden) des Kontos ausschließlich auf Basis der DNS-Informationen der Domain ermittelt.</string><string name="listen_address_help">IP-Adresse und Port in der Form \'address:port\', an der Baresip für eingehende SIP-Anfragen lauscht. Wenn die IP-Adresse eine IPv6-Adresse ist, muss sie innerhalb von Klammern geschrieben werden []. IPv4 Adresse 0.0.0.0 oder IPv6 Adresse [:] läßt Baresip an allen verfügbaren Adressen lauschen. Wenn leer gelassen (Standard), hört Baresip auf Port 5060 aller verfügbaren Adressen.</string><string name="address_family_help">Wählt, welche IP-Adressen Baresip verwendet. Wird IPv4 oder IPv6 gewählt, verwendet Baresip nur IPv4 oder IPv6 Adressen. Wird nichts gewählt, verwendet Baresip sowohl IPv4 als auch IPv6 Adressen.</string><string name="dns_servers_help">Kommagetrennte Liste von Adressen von DNS-Servern. Wenn nicht angegeben, werden DNS-Serveradressen dynamisch aus dem System gewonnen. Jede DNS-Adresse ist in der Form \'ip:port\' oder \'ip\'. Wenn der Port weggelassen ist, wird dieser auf 53 gesetzt. Wenn ip eine IPv6-Adresse ist und auch Port angegeben ist, muss ip in Klammern geschrieben werden []. Als Beispiel zeigt die Liste \'8.8.8.8.8:53,[2001:4860:4860:::888888]:53\' auf IPv4 und IPv6 Adressen öffentlicher Google DNS-Server.</string><string name="tls_certificate_file_help">Falls angeklickt, wurde oder wird eine Datei mit TLS-Zertifikat und privatem Schlüssel dieser Baresip-Instanz geladen. In Android-Version 9 wird eine Datei namens \'cert.pem\' aus dem Download-Ordner geladen. Löschen Sie aus Sicherheitsgründen diese Datei nach dem Laden.</string><string name="video_codecs">Video Codecs</string><string name="audio_settings">Audioeinstellungen</string><string name="backup">Sicherungskopie</string><string name="aec_help">Falls angekreuzt, wird versucht, Echounterdrückung auf den Anruf anzuwenden.</string><string name="rtcp_mux">RTCP Multiplexverfahren</string><string name="invalid_opus_packet_loss">Ungültiger Opus Paketverlustprozentsatz</string><string name="default_call_volume">Standardmäßige Anruflautstärke</string><string name="stun_server">STUN/TURN Server</string><string name="stun_server_uri">STUN/TURN Server URI</string><string name="dtmf_inband">RTP Ergeignisse im Frequenzband</string><string name="audio_codecs">Audio Codecs</string><string name="tls_ca_file">TLS CA Datei</string><string name="contacts_consent">Wenn Android Kontakte gewählt ist, können diese beim Telefonieren und Verschicken von Direktnachrichten als Referenzen für SIP und tel URIs genutzt werden. Die baresip App speichert keine Android Kontakte und teilt diese auch nicht. Um Android Kontakte in baresip verfügbar zu machen, erfordert Google Ihre Zustimmung zur Nutzung der Kontakte nach den Angaben in der <a href="https://raw.githubusercontent.com/juha-h/baresip-studio/master/PrivacyPolicy.txt">Datenschutzerklärung</a> der App.</string><string name="attended">Mit Rückfrage (attended)</string><string name="choose_destination_uri">Ziel-U
|
|||
|
|
<h1> Agente de usuario SIP basado en la biblioteca Baresip </h1>
|
|||
|
|
<p> Juha Heinanen < jh@tutpro.com > </p>
|
|||
|
|
<p> Versión %1$s </p>
|
|||
|
|
<h2> Consejos de uso </h2>
|
|||
|
|
<ul>
|
|||
|
|
<li> Compruebe que los valores predeterminados en la configuración de baresip se ajusten a sus necesidades
|
|||
|
|
(toque los títulos de los elementos para obtener ayuda). </li>
|
|||
|
|
<li> Luego, en Cuentas, crea una o más cuentas (nuevamente toca los títulos de los elementos para obtener ayuda). </li>
|
|||
|
|
<li> El estado de registro de una cuenta se muestra con un punto de color: verde (registro
|
|||
|
|
exitoso), amarillo (el registro está en proceso), rojo (el registro falló), blanco (el registro está en progreso).
|
|||
|
|
no ha sido activado). </li>
|
|||
|
|
<li> Al tocar el punto se accede directamente a la configuración de la cuenta. </li>
|
|||
|
|
<li> El gesto de deslizar hacia abajo provoca el nuevo registro de la cuenta mostrada actualmente. </li>
|
|||
|
|
<li> Al mantener pulsada la cuenta que se muestra actualmente, se habilita o deshabilita el registro de la cuenta. </li>
|
|||
|
|
<li> El gesto de deslizar hacia la izquierda o hacia la derecha alterna entre las cuentas. </li>
|
|||
|
|
<li> Se puede volver a seleccionar la persona que llamó anteriormente tocando el ícono de llamada cuando el destinatario esté vacío. </li>
|
|||
|
|
<li> Los pares de llamadas y mensajes se pueden agregar a los contactos mediante toques prolongados. </li>
|
|||
|
|
<li> Los toques prolongados también se pueden utilizar para eliminar llamadas, chats, mensajes y contactos. </li>
|
|||
|
|
<li> Se puede utilizar el toque o toque prolongado del ícono de contacto para instalar o eliminar la imagen del avatar. </li>
|
|||
|
|
<li> Consulte <a href="https://github.com/juha-h/baresip-studio/wiki"> Wiki </a> para obtener más información.
|
|||
|
|
información. </li>
|
|||
|
|
</ul>
|
|||
|
|
<h2> Política de privacidad </h2>
|
|||
|
|
La política de privacidad está disponible <a href="https://raw.githubusercontent.com/juha-h/baresip-studio/master/PrivacyPolicy.txt"> aquí </a> .
|
|||
|
|
<h2> Código fuente </h2>
|
|||
|
|
El código fuente está disponible en <a href="https://github.com/juha-h/baresip-studio"> GitHub </a> ,
|
|||
|
|
donde también se pueden reportar problemas.
|
|||
|
|
<h2> Licencias </h2>
|
|||
|
|
<ul>
|
|||
|
|
<li> <b> Cláusula BSD-3 </b> excepto lo siguiente: </li>
|
|||
|
|
<li> <b> Códecs AMR y seguridad TLS de Apache 2.0 </b>
|
|||
|
|
<li> <b> AGPLv4 </b> Cifrado de medios ZRTP </li>
|
|||
|
|
<li> <b> GNU LGPL 2.1 </b> Códecs G.722, G.726 y Codec2 </li>
|
|||
|
|
<li> <b> GNU GPLv3 </b> Códec G.729 </li>
|
|||
|
|
</ul>
|
|||
|
|
]]></string><string name="account">Cuenta</string><string name="display_name">Nombre para mostrar</string><string name="your_name">Su nombre</string><string name="display_name_help">Nombre (si lo hay) utilizado en el URI de origen de las solicitudes salientes.</string><string name="authentication_username">Nombre de usuario de autenticación</string><string name="authentication_username_help">Nombre de usuario de autenticación si se requiere la autenticación de las solicitudes SIP. El valor por defecto es el nombre de usuario de la cuenta.</string><string name="authentication_password">Contraseña de autenticación</string><string name="authentication_password_help">Contraseña de autenticación de hasta 64 caracteres. Si se proporciona el nombre de usuario, pero no la contraseña, esta se le pedirá cuando inicie baresip.</string><string name="outbound_proxies">Proxies salientes</string><string name="outbound_proxies_help">URI SIP de uno o dos proxies que deben utilizarse al enviar las solicitudes. Si se dan dos, las solicitudes de REGISTRO se envían a ambos y las demás solicitudes se envían a uno que responda. Si no se indica ningún proxy de salida, las solicitudes se envían basándose en la búsqueda del registro DNS NAPTR/SRV/A de la parte de host del URI del destinatario. Si la parte del host del URI SIP es una dirección IPv6, la dirección debe escribirse entre corchetes [].
|
|||
|
|
\nEjemplos:
|
|||
|
|
\n - sip:ejemplo.com:5061;transporte=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 del servidor proxy</string><string name="sip_uri_of_another_proxy_server">SIP URI de otro servidor proxy</string><string name="register">Registrar</string><string name="register_help">Si está marcado, el registro está habilitado y las solicitudes de REGISTRO se envían en el intervalo especificado por Intervalo de registro.</string><string name="audio_codecs">Códecs de audio</string><string name="audio_codecs_help">Lista de códecs de audio por orden de prioridad. Arrastre para reordenar, deslice a la derecha para activar o desactivar.</string><string name="media_nat">Media NAT transversal</string><string name="media_nat_help">Selecciona el protocolo transversal de NAT media (si lo hay). Las posibles opciones son STUN
|
|||
|
|
(Utilidades de recorrido de sesión para NAT, RFC 5389) e ICE (conectividad interactiva
|
|||
|
|
Establecimiento, RFC 5245).
|
|||
|
|
</string><string name="stun_server">Servidor STUN / TURN</string><string name="stun_server_help">Un URI de servidor STUN/TURN de la forma scheme:host[:port][\?transport=udp|tcp], donde scheme es \'stun\', \'stuns\', \'turn\', o \'turns\'. El servidor STUN predeterminado de fábrica para los protocolos STUN e ICE es \'stun:stun.l.google.com:19302\' que apunta al servidor STUN público de Google. No hay servidor TURN por defecto.</string><string name="media_encryption">Cifrado de medios</string><string name="media_encryption_help">Selecciona el protocolo de cifrado de transporte de medios (si lo hay).
|
|||
|
|
\n • ZRTP (recomendado) significa que la negociación de cifrado de medios de extremo a extremo de ZRTP se intenta después que
|
|||
|
|
la llamada ha sido establecida.
|
|||
|
|
\n • DTLS-SRTPF significa que UDP / TLS / RTP / SAVPF se ofrece en llamadas salientes y que RTP / SAVP,
|
|||
|
|
RTP / SAVPF, UDP / TLS / RTP / SAVP o UDP / TLS / RTP / SAVPF se usa si se ofrece en la llamada entrante.
|
|||
|
|
\n • SRTP-MANDF significa que RTP / SAVPF se ofrece en llamadas salientes y se requiere en llamadas entrantes.
|
|||
|
|
\n • SRTP-MAND significa que RTP / SAVP se ofrece en llamadas salientes y se requiere en llamadas entrantes.
|
|||
|
|
\n • SRTP significa que RTP / AVP se ofrece en llamadas salientes y que se utiliza RTP / SAVP o RTP / SAVPF
|
|||
|
|
si se ofrece en llamada entrante.
|
|||
|
|
</string><string name="answer_mode">Modo de contestación</string><string name="answer_mode_help">Selecciona cómo se contestan las llamadas entrantes.</string><string name="manual">Manual</string><string name="auto">Automático</string><string name="voicemail_uri">URI de correo de voz</string><string name="voicemain_uri_help">URI de SIP para comprobar los mensajes de correo de voz. Si se deja vacío, no se suscribirá a los mensajes de correo de voz (indicaciones de mensaje en espera).</string><string name="default_account">Cuenta predeterminada</string><string name="default_account_help">Si está marcada, esta cuenta se selecciona cuando se inicia baresip.
|
|||
|
|
</string><string name="accounts">Cuentas</string><string name="user_domain">usuario@dominio</string><string name="invalid_aor">usuario@dominio[:puerto][;transport=udp|tcp|tls] «%1$s» no válido</string><string name="account_exists">Ya existe la cuenta «%1$s».</string><string name="account_allocation_failure">Error al asignar la cuenta nueva.</string><string name="encrypt_password">Contraseña de cifrado</string><string name="decrypt_password">Contraseña para descifrar</string><string name="delete_account">¿Quiere eliminar la cuenta «%1$s»\?</string><string name="answer">Contestar</string><string name="reject">Rechazar</string><string name="incoming_call_from">Llamada entrante de</string><string name="transfer_request">Solicitud de transferencia</string><string name="message_from">Mensaje de</string><string name="call_history">Historial de llamadas</string><string name="call">Llamada</string><string name="calls_calls">llamadas</string><string name="calls_call">llamar</string><string name="calls_call_message_question">¿Quiere llamar o enviar un mensaje a «%1$s»\?</string><string name="calls_add_delete_question">¿Quiere añadir a «%1$s» a los contactos o eliminar %2$s del historial de llamadas\?</string><string name="calls_delete_question">¿Quiere eliminar «%1$s» %2$s del historial de llamadas\?</string><string name="delete_history">Eliminar</string><string name="disable_history">Desactivar</string><string name="enable_history">Activar</string><string name="delete_history_alert">¿Quiere eliminar el historial de llamadas de la cuenta «%1$s»\?</string><string name="chat">Mensajes de chat</string><string name="chat_with">Chatear con %1$s</string><string name="new_message">Mensaje nuevo</string><string name="long_message_question">¿Quiere eliminar el mensaje o añadir el par «%1$s» a los contactos\?</string><string name="short_message_question">¿Quiere eliminar el mensaje\?</string><string name="add_contact">Añadir contacto</string><string name="sending_failed">Envío de mensaje fallido</string><string name="message_failed">Ha fallado</string><string name="chats">Historial de chat</string><string name="today">Hoy</string><string name="you">Usted</string><string name="new_chat_peer">Nuevo compañero de chat</string><string name="invalid_chat_peer_uri">URI de chat no válido</string><string name="long_chat_question">¿Quieres eliminar el chat con un compañero? \'%1$s\' o
|
|||
|
|
agregar pares a los contactos?</string><string name="short_chat_question">¿Quiere eliminar el chat con «%1$s»\?</string><string name="delete_chats">Eliminar</string><string name="delete_chats_alert">¿Quiere eliminar el historial de chat de la cuenta «%1$s»\?</string><string name="configuration">Configuración</string><string name="start_automatically">Comenzar automáticamente</string><string name="start_automatically_help">Si está marcado, baresip se inicia automáticamente después de que el dispositivo se (re)inicia.</string><string name="listen_address">Dirección de escucha</string><string name="listen_address_help">Dirección IP y puerto de formulario \'address:port\' en el que escucha baresip
|
|||
|
|
para solicitudes SIP entrantes. Si la dirección IP es una dirección IPv6, debe escribirse dentro
|
|||
|
|
soportes []. La dirección IPv4 0.0.0.0 o la dirección IPv6 [::] hace que la escucha de baresip sea
|
|||
|
|
Direcciones disponibles. Si se deja vacío (predeterminado de fábrica), baresip escucha en el puerto 5060 de
|
|||
|
|
todas las direcciones disponibles.
|
|||
|
|
</string><string name="invalid_listen_address">Dirección de escucha no válida</string><string name="dns_servers">Servidores DNS</string><string name="dns_servers_help">Lista separada por comas de direcciones de servidores DNS. Si no se da,
|
|||
|
|
Las direcciones del servidor DNS se obtienen dinámicamente del sistema. Cada dirección DNS es de forma
|
|||
|
|
\'ip:port\' o \'ip\'. Si se omite el puerto, el valor predeterminado es 53. Si ip es una dirección IPv6 y
|
|||
|
|
también se da puerto, IP debe
|
|||
|
|
estar escrito entre corchetes []. Como ejemplo, lista \'8.8.8.8:53,[2001:4860:4860::8888]:53\'
|
|||
|
|
apunta a direcciones IPv4 e IPv6 de servidores DNS públicos de Google.</string><string name="invalid_dns_servers">Servidores DNS no válidos</string><string name="failed_to_set_dns_servers">Error al establecer servidores DNS</string><string name="tls_certificate_file">Archivo de certificado TLS</string><string name="tls_certificate_file_help">Si se marca, se ha cargado o se cargará un archivo que contiene el certificado TLS y la clave privada de esta instancia de baresip. En las versiones Android 9, se carga un archivo llamado \'cert.pem\' desde la carpeta Download. Por razones de seguridad, elimine el archivo después de cargarlo.</string><string name="tls_ca_file">Archivo de CA de TLS</string><string name="tls_ca_file_help">Si está marcada, se ha cargado o se cargará un archivo que contiene certificados TLS de dichas Autoridades de Certificación que no están incluidas en el SO Android. En las versión de Android 9, se carga un archivo llamado \'ca_certs.crt\' desde la carpeta Download.</string><string name="aec">Cancelación de eco acústico</string><string name="aec_help">Si está marcada, se intenta la cancelación del eco en el audio de la llamada.</string><string name="opus_bit_rate">Tasa de bits de Opus</string><string name="opus_bit_rate_help">Velocidad de bits máxima promedio utilizada por la transmisión de audio Opus.
|
|||
|
|
Los valores válidos son 6000-510000. El valor predeterminado de fábrica es 28000.</string><string name="opus_packet_loss">Pérdida de paquetes de Opus esperada</string><string name="opus_packet_loss_help">Porcentaje esperado de la pérdida de los paquetes del flujo de audio Opus, de 0 a 100. El valor predeterminado de fábrica es 1. El valor 0 también desactiva la corrección de errores hacia delante (FEC) de Opus. El valor predeterminado de fábrica es 1. El valor 0 también desactiva Corrección de errores de reenvío de Opus (FEC).</string><string name="invalid_opus_bitrate">Tasa de bits de Opus no válida</string><string name="invalid_opus_packet_loss">Porcentaje de pérdida de paquetes opus no válido</string><string name="default_call_volume">Volumen de llamada predeterminado</string><string name="default_call_volume_help">Si está configurado, el volumen de audio predeterminado de la llamada en escala 1–10.</string><string name="debug">Depurar</string><string name="debug_help">Si se marca, proporciona mensajes de registro de nivel de depuración e información a Logcat.</string><string name="reset_config">Restablecer los valores de fábrica</string><string name="reset_config_help">Si está marcado, la configuración se restablece
|
|||
|
|
a los valores predeterminados de fábrica.</string><string name="read_cert_error">Fallo al leer el archivo \'cert.pem\'.</string><string name="read_ca_certs_error">Error al leer el archivo \'ca_certs.crt\'.</string><string name="contact">Contacto</string><string name="new_contact">Contacto nuevo</string><string name="contact_name">Nombre</string><string name="invalid_contact">El nombre de contacto «%1$s» no es válido</string><string name="contact_already_exists">Ya existe el contacto «%1$s».</string><string name="invalid_contact_uri">URI de SIP no válido</string><string name="contacts">Contactos</string><string name="contact_action_question">¿Quieres llamar o enviar un mensaje a \'%1$s\'?</string><string name="send_message">Enviar mensaje</string><string name="contact_delete_question">¿Quiere eliminar el contacto «%1$s»\?</string><string name="contacts_exceeded">Su número máximo de contactos %1$d se ha excedido.</string><string name="alert">Alerta</string><string name="info">Información</string><string name="notice">Aviso</string><string name="cancel">Cancelar</string><string name="ok">De acuerdo</string><string name="yes">Sí</string><string name="no">No</string><string name="accept">Aceptar</string><string name="deny">Denegar</string><string name="user_id">Id. de usuario</string><string name="password">Contraseña</string><string name="add">Añadir</string><string name="delete">Eliminar</string><string name="edit">Editar</string><string name="send">Enviar</string><string name="status">Status</string><string name="error">Error</string><string name="backup">Copia de respaldo</string><string name="restore">Restaurar</string><string name="about">Acerca de</string><string name="restart">Reiniciar</string><string name="quit">Salir</string><string name="outgoing_call_to_dots">Llamar a …</string><string name="incoming_call_from_dots">Llamada de …</string><string name="transferring_call_to_dots">Transfiriendo llamada a …</string><string name="invalid_sip_uri">URI de SIP no válido \'%1$s\'</string><string name="callee">Destinatario</string><string name="hangup">Colgar</string><string name="hold">Retención/desconexión de llamadas</string><string name="dtmf">DTMF</string><string name="call_info">Información de llamada</string><string name="duration">Duración: %1$d (segs)</string><string name="codecs">Códecs</string><string name="rate">Velocidad actual: %1$s (Kbits/s)</string><string name="voicemail">Mensaje de voz</string><string name="voicemail_messages">Mensajes de correo de voz</string><string name="you_have">Tiene</string><string name="one_new_message">un mensaje nuevo</string><string name="new_messages">mensajes nuevos</string><string name="one_old_message">un mensaje antiguo</string><string name="old_messages">mensajes antiguos</string><string name="and">y</string><string name="no_messages">No tiene mensajes</string><string name="listen">Escuchar</string><string name="messages">Mensajes</string><string name="dialpad">Teclado</string><string name="call_already_active">Ya tiene una llamada activa.</string><string name="start_failed">Baresip no se ha podido iniciar. Esto puede deberse a un valor de configuración no válido. Compruebe la dirección de escucha, el archivo de certificado TLS y el archivo TLS CA. Luego reinicie baresip.</string><string name="registering_failed">Registro de \`%1$s\` ha fallado.</string><string name="verify">Verificar la solicitud</string><string name="verify_sas">¿Quieres verificar SAS <%1$s>\?</string><string name="transfer_request_query">¿Acepta transferir la llamada a «%1$s»\?</string><string name="call_failed">Llamada fallida</string><string name="call_closed">La llamada está cerrada</string><string name="call_not_secure">¡Esta llamada NO es segura!</string><string name="peer_not_verified">¡Esta llamada es SEGURA, pero el par NO está verificado!</string><string name="call_is_secure">¡Esta llamada es SEGURA y el compañero está VERIFICADO!
|
|||
|
|
¿Quieres desverificar al compañero?</string><string name="unverify">No verificar</string><string name="backed_up">Los datos de la aplicación han sido respaldados en el archivo \'%1$s\'. En las versión de Android 9, el archivo está en la carpeta Download.</string><string name="backup_failed">Ha fallado la copia de seguridad de los datos de la aplicación en el archivo \'%1$s\'. Compruebe Apps → baresip → Permisos → Almacenamiento.</string><string name="restored">Se restauraron los datos de la aplicación. Baresip necesita reiniciarse. ¿Quiere reiniciar ahora\?</string><string name="restore_failed">No se han podido restaurar los datos de la aplicación. Compruebe que ha dado la contraseña correcta y que el archivo de copia de seguridad es de esta aplicación. En las versión de Android 9, compruebe también Aplicaciones → baresip → Permisos → Almacenamiento y que el archivo \'%1$s\' existe en la carpeta Download.</string><string name="config_restart">Es necesario reiniciar Baresip para que surta efecto la configuración nueva. ¿Quiere reiniciar ahora\?</string><string name="audio_modules_title">Módulos de audio</string><string name="audio_modules_help">Las cuentas pueden utiilzar códecs de audio provistos por los módulos comprobados.</string><string name="failed_to_load_module">Falló la carga del módulo.</string><string name="no_calls">baresip necesita permiso de \"Micrófono\" para las llamadas de voz.</string><string name="invalid_authentication_password">Contraseña incorrecta%1$s</string><string name="invalid_authentication_username">Nombre de usuario no válido \'%1$s\'</string><string name="video_codecs_help">Lista de códecs de vídeo por orden de prioridad. Arrastre para reordenar, deslice a la derecha para activar o desactivar.</string><string name="show_password">Mostrar contraseña</string><string name="no_cameras">No tiene ninguna cámara de video compatible.</string><string name="no_video_calls">Concede permiso a \"Cámara\" para realizar o responder videollamadas.</string><string name="restart_request">Solicitud de reinicio</string><string name="call_info_not_available">No hay información disponible</string><string name="transfer_failed">Error en la transferencia</string><string name="transfer">Transferir</string><string name="transfer_destination">Destino de transferencia</string><string name="call_transfer">Transferencia de llamadas</string><string name="mic">Micrófono encendido/apagado</string><string name="allow_video_recv">¿Aceptar la recepción de vídeo de \'%1$s\'\?</string><string name="allow_video_send">¿Acepta el envío de video a \'%1$s\'\?</string><string name="allow_video">¿Aceptar el envío y la recepción de vídeo con \'%1$s\'\?</string><string name="video_request">Solicitud de vídeo</string><string name="video_call">Videollamada</string><string name="confirmation">Confirmación</string><string name="help">Ayuda</string><string name="android_contact_help">Si está marcada, este contacto se añade a los contactos de Android.</string><string name="reset">Restablecer</string><string name="reset_config_alert">¿Estás seguro de que quieres restablecer los valores de fábrica\?</string><string name="sip_trace_help">Si está marcada y si Debug está marcada, los mensajes Logcat incluyen también la petición SIP y la traza de respuesta. Desmarcado automáticamente al iniciar baresip.</string><string name="sip_trace">Seguimiento SIP</string><string name="video_size_help">Tamaño de los cuadros de vídeo transmitidos (ancho x alto)</string><string name="video_size">Tamaño del fotograma de vídeo</string><string name="dark_theme_help">Forzar el tema de la pantalla oscura</string><string name="dark_theme">Tema oscuro</string><string name="aec_extended_filter_help">Si está marcada, la cancelación de eco está utilizando el filtro extendido.</string><string name="aec_extended_filter">Filtro extendido AEC</string><string name="verify_server_help">Si está marcada, baresip verifica los certificados TLS del Agente de Usuario SIP y de los Servidores Proxy SIP cuando
|
|||
|
|
<h1>Agente de usuario SIP basado en biblioteca Baresip con videollamadas</h1>
|
|||
|
|
<p>Juha Heinanen <jh@tutpro.com></p>
|
|||
|
|
<p>Versión %1$s</p>
|
|||
|
|
<h2>Consejos de uso</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li>Comprobar que los valores por defecto en la configuración de baresip+ se ajustan a sus necesidades
|
|||
|
|
(Toque los títulos de los elementos para obtener ayuda). </li>
|
|||
|
|
<li>Luego, en Cuentas, cree una o más cuentas (otra vez toque los títulos de los elementos para obtener ayuda). </li>
|
|||
|
|
<li>El estado de registro de una cuenta se muestra con un punto de color: verde (registro
|
|||
|
|
Ha sido aprobado), amarillo (el registro está en curso), rojo (el registro falló), blanco (el registro está en curso
|
|||
|
|
no se ha activado). </li>
|
|||
|
|
<li>Tocar el punto lleva directamente a la configuración de la cuenta. </li>
|
|||
|
|
<li>Deslizar el dedo hacia abajo provoca que la cuenta mostrada actualmente vuelva a registrarse. </li>
|
|||
|
|
<li>El toque largo en la cuenta mostrada actualmente activa o desactiva el registro de la cuenta. </li>
|
|||
|
|
<li>El gesto de deslizar hacia la izquierda/derecha alterna entre las cuentas. </li>
|
|||
|
|
<li>La llamada anterior puede ser reseleccionada tocando el icono de llamada cuando Callee está vacío. </li>
|
|||
|
|
<li>Los pares de llamadas y mensajes se pueden agregar a contactos con toques largos. </li>
|
|||
|
|
<li>Los toques largos también se pueden usar para eliminar llamadas, chats, mensajes y contactos. </li>
|
|||
|
|
<li>El icono de contacto puede utilizarse para instalar/eliminar el avatar de la imagen. </li>
|
|||
|
|
<li>Ver <a href="https://github.com/juha-h/baresip-studio/wiki">Wiki</a> para más información
|
|||
|
|
información. </li>
|
|||
|
|
</ul>
|
|||
|
|
<h2>Problemas conocidos</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li>En las videollamadas, el dispositivo debe mantenerse en horizontal
|
|||
|
|
modo girado 90 grados a la izquierda de la orientación vertical. </li>
|
|||
|
|
<li>La vista de usuario no se muestra correctamente cuando se transmite vídeo. </li>
|
|||
|
|
</ul>
|
|||
|
|
<h2>Política de privacidad</h2>
|
|||
|
|
La política de privacidad está disponible <a href="https://raw.githubusercontent.com/juha-h/baresip-studio/video/PrivacyPolicy.txt">aquí</a>.
|
|||
|
|
<h2>Código fuente</h2>
|
|||
|
|
El código fuente está disponible en <a href="https://github.com/juha-h/baresip-studio">GitHub</a>,
|
|||
|
|
También se pueden comunicar problemas.
|
|||
|
|
<h2>Licencias</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li><b>Cláusula BSD-3</b> excepto lo siguiente:</li>
|
|||
|
|
<li><b>Apache 2.0</b> Códecs AMR y seguridad TLS</li>
|
|||
|
|
<li><b>AGPLv4</b> Cifrado de medios ZRTP</li>
|
|||
|
|
<li><b>Codecs GNU LGPL 2.1</b> G.722, G.726 y Codec2</li>
|
|||
|
|
<li><b>El códec G.729 de la GNU GPLv3</b> </li>
|
|||
|
|
<li><b>Codecs H.264 y H.265 de la GNU GPLv2</b></li>
|
|||
|
|
<li><b>Codec AV1 de AOMedia</b> </li>
|
|||
|
|
</ul>
|
|||
|
|
]]></string><string name="about_title_plus">Acerca de baresip+</string><string name="reg_int">Intervalo de inscripción</string><string name="invalid_reg_int">Intervalo de registro no válido\'%1$s\'</string><string name="rtcp_mux">Multiplexación RTCP</string><string name="country_code">Código del país</string><string name="reg_int_help">Indica la frecuencia (en segundos) con la que baresip envía peticiones REGISTRO. Los valores válidos van de 60 a 3600.</string><string name="rtcp_mux_help">Si está marcada, los paquetes RTP y RTCP se multiplexan en un único puerto (RFC 5761).</string><string name="account_nickname">Apodo de la cuenta</string><string name="account_nickname_help">Apodo (si lo hay) utilizado para identificar esta cuenta dentro de la aplicación baresip.</string><string name="invalid_account_nickname">Apodo de la cuenta no válido \"%1$s</string><string name="non_unique_account_nickname">El apodo \'%1$s\' ya existe</string><string name="nickname">Apodo</string><string name="both">Ambos</string><string name="country_code_hint">+código</string><string name="call_details">Detalles de la llamada</string><string name="peer">Interlocutores</string><string name="direction">Dirección</string><string name="calls_duration">Duración</string><string name="audio_settings">Configuración del audio</string><string name="user_domain_or_number">usuario@dominio o número de teléfono</string><string name="avatar_image">Imagen del perfil</string><string name="telephony_provider">Proveedor de telefonía</string><string name="telephony_provider_help">Parte del host SIP URI utilizada en las llamadas a números de teléfono. Por defecto es el dominio de la cuenta. Si no se indica, esta cuenta no puede utilizarse para llamar a números de teléfono.</string><string name="telephony_provider_hint">Parte del host SIP URI</string><string name="invalid_country_code">Código del país \"%1$s\" no válido</string><string name="invalid_sip_uri_hostpart">Parte del host SIP URI \'%1$s\' no válida</string><string name="codec_action">Reordenar</string><string name="contacts_help">Elige si se utilizan contactos de Baresip, contactos de Android o ambos. Si se utilizan ambos y existe un contacto con el mismo nombre en ambos contactos, se elegirá el contacto bareip.</string><string name="consent_request">Solicitud de autorización</string><string name="sip_or_tel_uri">SIP o teléfono URI</string><string name="call_is_on_hold">Llamada en espera</string><string name="no_restore">No es posible restaurar la copia de seguridad sin el permiso de \"Almacenamiento\".</string><string name="no_network">¡No hay conexion de red!</string><string name="no_telephony_provider">La cuenta \'%1$s\' no tiene proveedor de telefonía</string><string name="average_rate">Velocidad media: %1$s (Kbits/s)</string><string name="lost">Perdidos</string><string name="jitter">Oscilación: %1$s (ms)</string><string name="permissions_rationale">Fundamentos de los permisos</string><string name="blind">Oculto</string><string name="attended">Asistió</string><string name="missed_calls_count">%1$d llamadas perdidas</string><string name="battery_optimizations">Optimizaciones de la batería</string><string name="no_backup">No podrá crear copias de seguridad sin el permiso \"Almacenamiento\".</string><string name="anonymous">Anónimo</string><string name="unknown">Desconocido</string><string name="invalid_sip_or_tel_uri">URI del teléfono o SIP no válido \'%1$s\'</string><string name="packets">Paquetes</string><string name="audio_permissions">baresip necesita el permiso \"Micrófono\" para las llamadas de voz, el permiso \"Dispositivos cercanos\" para la detección de micrófonos/altavoces Bluetooth y el permiso \"Notificaciones\" para la publicación de notificaciones.</string><string name="missed_calls">Llamadas perdidas</string><string name="country_code_help">Código de país E.164 de esta cuenta. Si la parte de usuario From URI de la llamada entrante o del mensaje contiene un número de teléfono que no empieza por el signo \"+\" y si falla la búsqueda de contacto,
|
|||
|
|
<![CDATA[
|
|||
|
|
<h1>Baresip-kirjastoon perustuva, äänipuhelut ja pikaviestit mahdollistava SIP-sovellus</h1>
|
|||
|
|
<p>Juha Heinanen <jh@tutpro.com></p>
|
|||
|
|
<p>Versio %1$s</p>
|
|||
|
|
<h2>Käyttövihjeitä</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li>Tarkista, että baresip-sovelluksen Asetukset vastaavat tarpeitasi. Kunkin otsikon kosketus tarjoaa apua.</li>
|
|||
|
|
<li>Sen jälkeen luo yksi tai useampi tili. Jälleen kunkin otsikon kosketus tarjoaa apua.</li>
|
|||
|
|
<li>Tilin rekisteröintitila kerrotaan värillisellä pisteellä: vihreä (rekisteröinti onnistui),
|
|||
|
|
keltainen (rekisteröinti on meneillään), punainen (rekisteröinti epäonnistui),
|
|||
|
|
valkoinen (rekisteröintiä ei ole aktivoitu).</li> <li>Pisteen kosketus johtaa suoraan tilin asetuksiin.</li>
|
|||
|
|
<li>Pyyhkäisy alas aikaansaa parhaillaan näkyvissä olevan tilin uudelleen rekisteröinnin.</li>
|
|||
|
|
<li>Parhaillaan näkyvissä olevan tilin pitkä kosketus aktivoi tai passivoi tilin rekisteröinnin.</li>
|
|||
|
|
<li>Pyyhkäisyllä vasemmalle/oikealle voit vaihtaa näkyvissä olevaa tiliä.</li>
|
|||
|
|
<li>Voit valita edellisen puhelun uudelleen koskettamalla virheää luuria silloin, kun puhelun kohde on tyhjä.</li>
|
|||
|
|
<li>Voit lisätä puheluiden ja viestien kohteet yhteystietoihin pitkällä kosketuksella.</li>
|
|||
|
|
<li>Pitkillä kosketuksilla voit myös poistaa puheluita, viestiketjuja, viestejä ja yhteystietoja.</li>
|
|||
|
|
<li>Voit lisätä/poistaa yhteystiedon avatar-kuvan koskettamalla yhteystiedon ikonia lyhyesti/pitkästi.</li>
|
|||
|
|
<li>Katso lisätietoja <a href=https://github.com/juha-h/baresip-studio/wiki>Wiki</a>-sivulta.</li>
|
|||
|
|
</ul>
|
|||
|
|
<h2>Tietosuoja</h2>
|
|||
|
|
Sovelluksen tietosuojakäytäntö on luettavissa <a href="https://raw.githubusercontent.com/juha-h/baresip-studio/master/PrivacyPolicy.txt">täältä</a>.
|
|||
|
|
<h2>Lähdekoodi</h2>
|
|||
|
|
Lähdekoodi on saatavilla <a href=https://github.com/juha-h/baresip-studio>GitHub</a>:ssa,
|
|||
|
|
missä voi myös raportoida virheistä.
|
|||
|
|
<h2>Lisenssit</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li><b>BSD-3-Clause</b> except the following:</li>
|
|||
|
|
<li><b>BSD-3-Clause</b> paitsi seuraavat:</li>
|
|||
|
|
<li><b>Apache 2.0</b> AMR koodausmenetelmät ja TLS turvallisuus</li>
|
|||
|
|
<li><b>AGPLv4</b> ZRTP median salausmenetelmä</li>
|
|||
|
|
<li><b>GNU LGPL 2.1</b> G.722, G.726, ja Codec2 koodausmenetelmät</li>
|
|||
|
|
<li><b>GNU GPLv3</b> G.729 koodausmenetelmä</li>
|
|||
|
|
</ul>
|
|||
|
|
]]>
|
|||
|
|
</string><string name="about_text_plus">
|
|||
|
|
<![CDATA[
|
|||
|
|
<h1>Baresip-kirjastoon perustuva, videopuhelut ja pikaviestit mahdollistava SIP-sovellus</h1>
|
|||
|
|
<p>Juha Heinanen <jh@tutpro.com></p>
|
|||
|
|
<p>Versio %1$s</p>
|
|||
|
|
<h2>Käyttövihjeitä</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li>Tarkista, että baresip+-sovelluksen Asetukset vastaavat tarpeitasi. Kunkin otsikon kosketus tarjoaa apua.</li>
|
|||
|
|
<li>Sen jälkeen luo yksi tai useampi tili. Jälleen kunkin otsikon kosketus tarjoaa apua.</li>
|
|||
|
|
<li>Tilin rekisteröintitila kerrotaan värillisellä pisteellä: vihreä (rekisteröinti onnistui),
|
|||
|
|
keltainen (rekisteröinti on meneillään), punainen (rekisteröinti epäonnistui),
|
|||
|
|
valkoinen (rekisteröintiä ei ole aktivoitu).</li>
|
|||
|
|
<li>Pisteen kosketus johtaa suoraan tilin asetuksiin.</li>
|
|||
|
|
<li>Pyyhkäisy alas aikaansaa parhaillaan näkyvissä olevan tilin uudelleen rekisteröinnin.</li>
|
|||
|
|
<li>Parhaillaan näkyvissä olevan tilin pitkä kosketus aktivoi tai passivoi tilin rekisteröinnin.</li>
|
|||
|
|
<li>Pyyhkäisyllä vasemmalle/oikealle voit vaihtaa näkyvissä olevaa tiliä.</li>
|
|||
|
|
<li>Voit valita edellisen puhelun uudelleen koskettamalla virheää luuria silloin, kun puhelun kohde on tyhjä.</li>
|
|||
|
|
<li>Voit lisätä puheluiden ja viestien kohteet yhteystietoihin pitkällä kosketuksella.</li>
|
|||
|
|
<li>Pitkillä kosketuksilla voit myös poistaa puheluita, viestiketjuja, viestejä ja yhteystietoja.</li>
|
|||
|
|
<li>Voit lisätä/poistaa yhteystiedon avatar-kuvan koskettamalla yhteystiedon ikonia lyhyesti/pitkästi.</li>
|
|||
|
|
<li>Katso lisätietoja <a href=https://github.com/juha-h/baresip-studio/wiki>Wiki</a>-sivulta.</li>
|
|||
|
|
</ul>
|
|||
|
|
<h2>Tunnetut ongelmat</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li>Videopuheluissa laitetta on pidettävä vaakasuorassa asennossa kiertämällä sitä 90 astetta vasempaan
|
|||
|
|
pystysuorasta asennosta.</li>
|
|||
|
|
<li>Oman videokuvan näyttö ei toimi, jos video on vain lähetyssuuntainen.</li>
|
|||
|
|
</ul>
|
|||
|
|
<h2>Tietosuoja</h2>
|
|||
|
|
Sovelluksen tietosuojakäytäntö on luettavissa <a href="https://raw.githubusercontent.com/juha-h/baresip-studio/video/PrivacyPolicy.txt">täältä</a>.
|
|||
|
|
<h2>Lähdekoodi</h2>
|
|||
|
|
Lähdekoodi on saatavilla <a href=https://github.com/juha-h/baresip-studio>GitHub</a>:ssa,
|
|||
|
|
missä voi myös raportoida virheistä.
|
|||
|
|
<h2>Lisenssit</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li><b>BSD-3-Clause</b> except the following:</li>
|
|||
|
|
<li><b>BSD-3-Clause</b> paitsi seuraavat:</li>
|
|||
|
|
<li><b>Apache 2.0</b> AMR koodausmenetelmät ja TLS turvallisuus</li>
|
|||
|
|
<li><b>AGPLv4</b> ZRTP median salausmenetelmä</li>
|
|||
|
|
<li><b>GNU LGPL 2.1</b> G.722, G.726, ja Codec2 koodausmenetelmät</li>
|
|||
|
|
<li><b>GNU GPLv3</b> G.729 koodausmenetelmä</li>
|
|||
|
|
<li><b>GNU GPLv2</b> H.264 ja H.265 koodausmenetelmät</li>
|
|||
|
|
<li><b>AOMedia</b> AV1 koodausmenetelmä</li>
|
|||
|
|
</ul>
|
|||
|
|
]]>
|
|||
|
|
</string><string name="account">Tili</string><string name="account_nickname">Tilin lempinimi</string><string name="account_nickname_help">Lempinimi (jos annettu) millä tämä tili identifioidaan
|
|||
|
|
baresip sovelluksessa.</string><string name="nickname">Lempinimi</string><string name="invalid_account_nickname">Virheellinen tilin lempinimi \'%1$s\'</string><string name="non_unique_account_nickname">Lempinimi \'%1$s\' on jo olemassa</string><string name="display_name">Tilin käyttäjän nimi</string><string name="your_name">Tilin käyttäjän nimi</string><string name="display_name_help">Tilin käyttäjän nimi, joka esiintyy
|
|||
|
|
SIP-sanomien From URI:ssa (vapaaehtoinen).</string><string name="invalid_display_name">Virheellinen tilin käyttäjän nimi \'%1$s\'</string><string name="authentication_username">Käyttäjätunnus</string><string name="authentication_username_help">Todentamiseen käytettävä
|
|||
|
|
käyttäjätunnus, jos välityspalvelin vaatii sellaisen. Oletusarvo on
|
|||
|
|
tilin käyttäjätunnus.
|
|||
|
|
</string><string name="invalid_authentication_username">Virheellinen todennuskäyttäjänimi \'%1$s\'</string><string name="authentication_password">Todennuksen salasana</string><string name="authentication_password_help">Todentamiseen käytettävä
|
|||
|
|
salasana, jonka pituus on enintään 64 ASCII merkkiä. Jos
|
|||
|
|
käyttäjätunnus on annettu, mutta salasanaa ei ole annettu, se
|
|||
|
|
kysytään, kun baresip käynnistetään.
|
|||
|
|
</string><string name="invalid_authentication_password">Virheellinen todennussalasana \'%1$s\'</string><string name="outbound_proxies">Välityspalvelimet</string><string name="outbound_proxies_help">Yhden tai kahden
|
|||
|
|
välityspalvelimen SIP URI, joille SIP-sanomat lähetetään. Jos
|
|||
|
|
välityspalvelimia on annettu kaksi, REGISTER-sanomat yritetään
|
|||
|
|
lähettää molemmille välityspalvelimille ja muut sanomat yhdelle
|
|||
|
|
toiminnassa olevalle välityspalvelimelle. Jos välityspalvelimia ei
|
|||
|
|
ole annettu, SIP-sanomat lähetetään palvelimelle, mikä selviää
|
|||
|
|
kohteen domainille tehtävien NAPTR/SRV/Animipalvelukyselyiden perusteella.
|
|||
|
|
Jos välityspalvelimen osoite SIP
|
|||
|
|
URI:ssa on IPv6-osoite, osoite pitää kirjoittaa sulkujen [] sisään.
|
|||
|
|
\nEsimerkkejä:
|
|||
|
|
\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">Välityspalvelimen SIP URI</string><string name="sip_uri_of_another_proxy_server">Toisen
|
|||
|
|
välityspalvelimen SIP URI</string><string name="invalid_proxy_server_uri">Virheellinen välityspalvelimen URI \'%1$s\'</string><string name="register">Rekisteröi</string><string name="register_help">Jos merkitty, rekisteröinti on aktiivinen ja REGISTER-sanomat
|
|||
|
|
lähetetään Rekisteröintitaajuus-asetuksen mukaisesti.</string><string name="reg_int">Rekisteröintitaajuus</string><string name="reg_int_help">Kertoo kuinka usein baresip lähettää REGISTER-sanomia. Arvon on
|
|||
|
|
oltava välillä 60-3600 (sekunttia).</string><string name="invalid_reg_int">Virheellinen Rekisteröintitaajuus \'%1$s\'</string><string name="media_nat">Media NAT hallinta</string><string name="media_nat_help">Valitsee media NAT hallintaprotokollan
|
|||
|
|
(vapaaehtoinen). Vaihtoehtoja ovat STUN (Session Traversal Utilities
|
|||
|
|
for NAT, RFC 5389) ja ICE (Interactive Connectivity Establishment, RFC 5245).
|
|||
|
|
</string><string name="stun_server">STUN/TURN-palvelin</string><string name="stun_server_help">STUN/TURN-palvelimen muotoa \'kaava:palvelin[:portti][\?transport=udp|tcp]\'
|
|||
|
|
oleva URI, missä kaava on \'stun\', \'stuns\', \'turn\' tai \'turns\'. Oletus STUN-palvelin
|
|||
|
|
STUN- ja ICE-protokollille on \'stun:stun.l.google.com:19302\', joka osoittaa Google:n
|
|||
|
|
julkiseen STUN-palvelimeen. TURN-palvelimella ei ole oletusarvoa.
|
|||
|
|
</string><string name="stun_server_uri">STUN/TURN-palvelimen URI</string><string name="invalid_stun_server">Virheellinen STUN/TURN-palvelimen URI \'%1$s\'</string><string name="stun_username">STUN/TURN-käyttäjätunnus</string><string name="stun_username_help">Käyttäjätunnus jos STUN/TURN-palvelin vaatii sellaisen</string><string name="invalid_stun_username">Virheellinen käyttäjänimi %1$s</string><string name="stun_password">STUN/TURN-salasana</string><string name="stun_password_help">Salasana jos STUN/TURN-palvelin vaatii sellaisen</string><string name="invalid_stun_password">Virheellinen salasana \'%1$s\'</string><string name="media_encryption">Median salaus</string><string name="media_encryption_help">Valitsee median salausprotokollan (vapaaehtoinen).
|
|||
|
|
\n • ZRTP (suositeltu) tarkoittaa, että ZRTP-salausta yritetään neuvotella sen jälkeen, kun puhelu on alkanut.
|
|||
|
|
\n • DTLS-SRTPF tarkoittaa, että UDP/TLS/RTP/SAVPF-protokollaa tarjotaan lähteviin puheluihin
|
|||
|
|
ja että RTP/SAVP-, RTP/SAVPF-, UDP/TLS/RTP/SAVP- tai
|
|||
|
|
UDP/TLS/RTP/SAVPF-protokollaa käytetään, jos sellaista tarjotaan tulevassa puhelussa.
|
|||
|
|
\n • SRTP-MANDF tarkoittaa, että RTP/SAVPF-protokollaa tarjotaan
|
|||
|
|
lähtevissä puheluissa ja että se vaaditaan tulevissa puheluissa.
|
|||
|
|
\n • SRTP-MAND takoittaa, että RTP/SAVP-protokollaa tarjotaan
|
|||
|
|
lähteviin puheluihin ja että se vaaditaan tulevissa puheluissa.
|
|||
|
|
\n • SRTP tarkoittaa, että RTP/AVP-protokollaa tarjotaan
|
|||
|
|
lähteviin puheluihin ja että RTP/SAVP- tai RTP/SAVPF-protokollaa
|
|||
|
|
käytetään, jos sellaista tarjotaan tulevissa puheluissa.
|
|||
|
|
</string><string name="prefer_ipv6_media">Suosi IPv6-mediaprotokollaa</string><string name="prefer_ipv6_media_help">Jos merkitty, tarjoa IPv6-mediaprotokollaa (mikäli
|
|||
|
|
se on käytettävissä), jos puhelun molempien osapuolten tukemaa mediaprotokollaa
|
|||
|
|
ei saada automaattisesti selvitettyä.
|
|||
|
|
</string><string name="rtcp_mux">RTCP-multipleksaus</string><string name="rtcp_mux_help">Jos merkitty, RTP- and RTCP-paketit multipleksataan samaan porttiin
|
|||
|
|
(RFC 5761).
|
|||
|
|
</string><string name="rel_100">Luotettavat alustavat vastaukset</string><string name="rel_100_help">Jos merkitty, kutsussa ilmoitetaan luotettavien alustavien vastausten
|
|||
|
|
tukemisesta (RFC 3262).</string><string name="dtmf_mode">DTMF-moodi</string><string name="dtmf_mode_help">Valitsee tavan, miten DTMF-merkit 0–9, #, * ja A-D lähetetään.</string><string name="dtmf_inband">RTP-tapahtuma</string><string name="dtmf_info">SIP INFO -pyyntö</string><string name="dtmf_auto">RTP tai SIP INFO</string><string name="answer_mode">Vastaustapa</string><string name="answer_mode_help">Valitsee tulevien puheluiden vastaustavan.</string><string name="manual">Manuaalinen</string><string name="auto">Automaattinen</string><string name="redirect_mode">Uudelleenohjaustapa</string><string name="redirect_mode_help">Valitsee toteutetaanko puhelun uudelleenohjauspyyntö
|
|||
|
|
automaattisesti vai kysytäänkö vahvistusta.</string><string name="voicemail_uri">Puhepostin URI</string><string name="voicemain_uri_help">SIP URI, jota käytetään
|
|||
|
|
puhepostiviestien kuunteluun. Jos URI:a ei ole annettu, tietoa
|
|||
|
|
mahdollista puhepostiviesteistä (Message Waiting Indications) ei tilata.
|
|||
|
|
</string><string name="invalid_voicemail_uri">Virheellinen puhepostin URI \'%1$s\'</string><string name="country_code">Maakoodi</string><string name="country_code_help">Tämän tilin E.164-maakoodi. Jos tulevan puhelun tai viestin
|
|||
|
|
From URI:n käyttäjäosa sisältää puhelinnumeron, joka ei ala \'+\' merkillä, ja jos sitä
|
|||
|
|
ei löydy yhteystiedoista, niin tämä maakoodi lisätään numeron eteen ja etsintä tehdään
|
|||
|
|
uudelleen. Jos puhelinnumero alkaa yhdellä numerolla \'0\', niin numero \'0\'
|
|||
|
|
poistetaan ennen maakoodin lisäämistä.
|
|||
|
|
</string><string name="country_code_hint">+koodi</string><string name="invalid_country_code">Virheellinen maakoodi \'%1$s\'</string><string name="telephony_provider">Puhelinpalvelun tarjoaja</string><string name="telephony_provider_help">SIP URI:n domain-osa, jota käytetään soitettaessa
|
|||
|
|
puhelinnumeroihin. Tehdasasetus on tilin domain-osa. Jos tyhjä, niin tätä tiliä ei
|
|||
|
|
voi käyttää soitettaessa puhelinnumeroihin.
|
|||
|
|
</string><string name="telephony_provider_hint">SIP URI:n domain osa</string><string name="invalid_sip_uri_hostpart">Virheellinen SIP URI:n domain-osa \'%1$s\'</string><string name="default_account">Oletustili</string><string name="default_account_help">Jos merkitty, niin tämä tili on
|
|||
|
|
valittuna, kun baresip käynnistetään.
|
|||
|
|
</string><string name="accounts">Tilit</string><string name="new_account">Uusi tili</string><string name="accounts_help">Kun uusi tili luodaan, voidaan haluttaessa antaa myös porttinumero
|
|||
|
|
ja tiedonsiirtoprotokolla: <käyttäjä>@<domain>[:<portti>][;transport=udp|tcp|tls].
|
|||
|
|
Jos <portti> on annettu, mutta protokollaa ei ole annettu, protokolla on udp.
|
|||
|
|
Jos <portti> ei ole annettu, mutta protokolla on annettu, portti on joko 5060 tai 5061 (tls).
|
|||
|
|
Jos kumpaakaan ei ole annettu eikä välityspalvelinta ole määritelty, tilin mahdollinen
|
|||
|
|
rekisteröintipalvelin päätellään pelkästään domainin DNS-informaation perusteella.
|
|||
|
|
</string><string name="user_domain">käyttäjä@domain</string><string name="invalid_aor">Virheellinen käyttäjä@domain[:portti][;transport=udp|tcp|tls] \'%1$s\'
|
|||
|
|
</string><string name="account_exists">Tili \'%1$s\' on jo olemassa.</string><string name="account_allocation_failure">Uuden tilin luonti epäonnistui.</string><string name="encrypt_password">Tallenna salasanalla</string><string name="decrypt_password">Palauta salasanalla</string><string name="delete_account">Haluatko poistaa tilin \'%1$s\'\?</string><string name="answer">Vastaa</string><string name="reject">Hylkää</string><string name="incoming_call_from">Puhelu soittajalta</string><string name="missed_call_from">Vastaamaton puhelu soittajalta</string><string name="missed_calls">Vastaamattomia puheluita</string><string name="missed_calls_count">%1$d vastaamatonta puhelua</string><string name="transfer_request_to">Puhelun siirtopyyntö kohteeseen</string><string name="message_from">Viesti lähettäjältä</string><string name="call_auto_rejected">Automaattisesti hylätty puhelu soittajalta \`%1$s\`</string><string name="call_history">Puheluhistoria</string><string name="call">Soita</string><string name="calls_calls">puhelut</string><string name="calls_call">puhelun</string><string name="direction">Suunta</string><string name="peer">Kumppani</string><string name="time">Aika</string><string name="calls_duration">Kesto</string><string name="calls_call_message_question">Haluatko soittaa tai lähettää viestin kohteeseen
|
|||
|
|
\'%1$s\'\?
|
|||
|
|
</string><string name="calls_add_delete_question">Haluatko luoda uuden yhteystiedon \'%1$s\' tai poistaa
|
|||
|
|
%2$s puheluhistoriasta\?
|
|||
|
|
</string><string name="calls_delete_question">Haluatko poistaa \'%1$s\' %2$s puheluhistoriasta\?</string><string name="delete_history">Tyhjennä</string><string name="disable_history">Poista käytöstä</string><string name="enable_history">Ota käyttöön</string><string name="delete_history_alert">Haluatko tyhjentää tilin \'%1$s\' puheluhistorian\?</string><string name="chat">Viestiketju</string><string name="chat_with">Viestiketju %1$s</string><string name="new_message">Uusi viesti</string><string name="long_message_question">Haluatko poistaa viestin tai
|
|||
|
|
luoda uuden yhteystiedon \'%1$s\'\?</string><string name="short_message_question">Haluatko poistaa viestin\?</string><string name="add_contact">Lisää yhteystieto</string><string name="sending_failed">Viestin lähetys epäonnistui</string><string name="message_failed">Epäonnistui</string><string name="chats">Viestiketjut</string><string name="today">Tänään</string><string name="you">Sinä</string><string name="new_chat_peer">Uusi viestin kohde</string><string name="invalid_chat_peer_uri">Virheellinen viestikumppanin URI</string><string name="long_chat_question">Haluatko poistaa viestiketjun tai
|
|||
|
|
luoda uuden yhteystiedon \'%1$s\'\?</string><string name="short_chat_question">Haluatko poistaa viestiketjun \'%1$s\'\?</string><string name="delete_chats">Tyhjennä</string><string name="delete_chats_alert">Haluatko tyhjentää tilin \'%1$s\' viestihistorian\?</string><string name="audio_codecs">Audion koodausmenetelmät</string><string name="audio_codecs_help">Luettelo audion koodausmenetelmistä prioriteettijärjestyksessä.
|
|||
|
|
Järjestä raahaamalla ja pudottamalla, ota käyttöön tai poista käytöstä pyyhkäisemällä oikealle.
|
|||
|
|
</string><string name="video_codecs">Videon koodausmenetelmät</string><string name="video_codecs_help">Luettelo videon koodausmenetelmistä prioriteettijärjestyksessä.
|
|||
|
|
Järjestä raahaamalla ja pudottamalla, ota käyttöön tai poista käytöstä pyyhkäisemällä oikealle.
|
|||
|
|
</string><string name="codec_action">Järjestä</string><string name="configuration">Asetukset</string><string name="start_automatically">Käynnistä automaattisesti</string><string name="start_automatically_help">Jos merkitty, baresip
|
|||
|
|
käynnistyy automaattisesti, kun laite käynnistyy.</string><string name="appear_on_top_permission">Automaattinen käynnistys vaatii "Näytä päällimmäisenä"
|
|||
|
|
käyttöoikeuden.</string><string name="battery_optimizations">Akun käytön optimointi</string><string name="battery_optimizations_help">Ota akun käytön optimointi pois päältä (suositeltu),
|
|||
|
|
jos haluat vähentää todennäköisyyttä, että Android rajoittaa baresip-sovelluksen toimintaa
|
|||
|
|
ja pääsyä verkkoon.</string><string name="default_phone_app">Oletuspuhelinsovellus</string><string name="dialer_role_not_available">Puhelinrooli ei ole saatavana</string><string name="default_phone_app_help">Jos merkity, baresip on oletuspuhelinsovellus. Älä merkitse, jos laitteesi täytyy hallita myös muita kuin SIP-puheluita tai -viestejä.</string><string name="listen_address">Kuunteluosoite</string><string name="listen_address_help">IP-osoite ja portti muotoa
|
|||
|
|
\'osoite:portti\', missä baresip kuuntelee sisään tulevia
|
|||
|
|
SIP-sanomia. Jos IP-osoite on IPv6-osoite, se täytyy kirjoittaa
|
|||
|
|
sulkujen [] sisään. IPv4-osoite 0.0.0.0 tai IPv6-osoite [::]
|
|||
|
|
tarkoittaa kaikkia käytössä olevia osoitteita. Oletusarvo on tyhjä,
|
|||
|
|
jolloin baresip kuuntelee porttia 5060 kaikilla käytössä olevilla
|
|||
|
|
osoitteilla.
|
|||
|
|
</string><string name="invalid_listen_address">Virheellinen kuunteluosoite</string><string name="address_family">Osoiteperhe</string><string name="address_family_help">Valitsee, mitä IP-osoitteita baresip käyttää. Jos IPv4 tai IPv6
|
|||
|
|
on valittu, baresip käyttää ainoastaan IPv4- tai IPv6-osoitteita. Jos kumpaakaan ei ole
|
|||
|
|
valittu, baresip käyttää sekä IPv4- että IPv6-osoitteita.
|
|||
|
|
</string><string name="dns_servers">DNS-palvelimet</string><string name="dns_servers_help">Pilkulla toisistaan erotettu luettelo DNS-palvelijoiden
|
|||
|
|
osoitteita. Jos jätetään antamatta (oletus), osoitteet hankitaan dynaamisesti järjestelmästä.
|
|||
|
|
Kukin osoite on muotoa \'ip:portti\' tai \'ip\', missä ip on IPv4 tai IPv6 osoite. Jos ip
|
|||
|
|
on IPv6-osoite ja myös portti annetaan, pitää osoite kirjoittaa sulkujen [] sisään.
|
|||
|
|
Esimerkiksi luettelo \'8.8.8.8:53,[2001:4860:4860::8888]:53\' osoittaa Googlen julkisiin
|
|||
|
|
DNS-palvelijoihin.</string><string name="invalid_dns_servers">Virheelliset DNS-palvelimet</string><string name="failed_to_set_dns_servers">DNS-palvelinten asetus epäonnistui</string><string name="tls_certificate_file">TLS-sertifikaattitiedosto</string><string name="tls_certificate_file_help">Jos merkitty, tiedosto joka sisältää
|
|||
|
|
tämän baresip-sovelluksen julkisen ja yksityisen TLS-sertifikaatin, on joko jo ladattu
|
|||
|
|
tai tullaan lataamaan. Android versiossa 9 tiedosto nimeltään \'cert.pem\'
|
|||
|
|
ladataan Download-kansiosta. Turvallisuusyistä tuhoa tiedosto heti lataamisen jälkeen.
|
|||
|
|
</string><string name="verify_server">Tarkista palvelinten sertifikaatit</string><string name="verify_server_help">Jos merkitty, baresip tarkistaa SIP-palvelinten
|
|||
|
|
sertifikaatit, kun TLS-tiedonsiirto on käytössä.
|
|||
|
|
</string><string name="tls_ca_file">TLS CA-tiedosto</string><string name="tls_ca_file_help">Jos merkitty, tiedosto on joko jo ladattu tai tullaan
|
|||
|
|
lataamaan, joka sisältää sellaisten sertifikaattiauktoriteettien (CA) julkiset sertifikaatit,
|
|||
|
|
jotka eivät sisälly Android-käyttöjärjestelmään. Android versiossa 9 tiedosto
|
|||
|
|
nimeltään \'ca_certs.crt\' ladataan Download-kansiosta.
|
|||
|
|
</string><string name="audio_settings">Audio-asetukset</string><string name="speaker_phone">Kaiutin</string><string name="speaker_phone_help">Jos merkitty, kaiutin kytketään sutomaattisesti päälle
|
|||
|
|
kun puhelu alkaa.
|
|||
|
|
</string><string name="audio_modules_title">Audio-modulit</string><string name="audio_modules_help">Merkittyjen modulien tarjoamat audio-koodekit ovat tilien
|
|||
|
|
käytettävissä.
|
|||
|
|
</string><string name="failed_to_load_module">Modulin lataaminen epäonnistui.</string><string name="aec">Akustinen kaiun poisto (AKP)</string><string name="aec_help">Jos merkitty, kaikua yritetään poistaa ohjelmiston avulla
|
|||
|
|
puheluiden aikana.</string><string name="aec_extended_filter">AKP laajennettu suodatin</string><string name="microphone_gain">Mikrofonin äänikerroin</string><string name="microphone_gain_help">Kertoo mikrofonin äänen voimakkuuden tällä desimaaliluvulla.
|
|||
|
|
Minimikerroin on 1.0 (tehdasasetus), mikä ei muuta mikrofonin äänen voimakkuutta. Suuremmat
|
|||
|
|
arvot voivat vaikuttaa negatiivisesti äänen laatuun.</string><string name="invalid_microphone_gain">Virheellinen mikrofonin äänikerroin</string><string name="_1.0" translatable="false">1.0</string><string name="opus_bit_rate">Opus-koodekin bittinopeus</string><string name="opus_bit_rate_help">Opus-koodekin käyttämä
|
|||
|
|
keskimääräinen enimmäisnopeus. Mahdollisia arvoja ovat
|
|||
|
|
6000-510000. Oletusarvo on 28000.
|
|||
|
|
</string><string name="opus_packet_loss">Odotettu opus-pakettihäviö</string><string name="opus_packet_loss_help">Odotettu opus audio virran pakettihäviö prosentteina.
|
|||
|
|
Mahdolliset arvot ovat 0-100. Oletusarvo on 1. Arvo 0 poistaa käytöstä
|
|||
|
|
ennakoivan virheenkorjauksen.</string><string name="invalid_opus_bitrate">Virheellinen Opus-koodekin bittinopeus</string><string name="invalid_opus_packet_loss">Virheellinen Opus-koodekin odotettu pakettihäviö</string><string name="audio_delay">Audioviive</string><string name="audio_delay_help">Audion odotusviive (millisekunneissa) soitetun puhelun alkaessa.
|
|||
|
|
Aseta korkeampi arvo, jos et kuule vastaajan ääntä heti, kun puhelu alkaa.</string><string name="invalid_audio_delay">Virheellinen audioviive \'%1$s\'. Sallittu arvo on välillä 100–3000.</string><string name="default_call_volume">Oletus äänen voimakkuus</string><string name="default_call_volume_help">Jos valittu, puhelun äänen voimakkuus
|
|||
|
|
asteikolla 1–10.</string><string name="tone_country">Puheluäänien maa</string><string name="tone_country_help">Soitto-, odotus- ja varattuäänien maa</string><string name="dark_theme">Tumma teema</string><string name="dark_theme_help">Käytä aina tummaa näyttöteemaa</string><string name="video_size">Videon kehyskoko</string><string name="video_size_help">Lähetettävän videon kehyskoko (leveys x korkeus)</string><string name="video_fps">Videokehysten lähetystaajuus</string><string name="video_fps_help">Videokehysten lähetystaajuus per sekuntti, jota tarjotaan
|
|||
|
|
SDP-neuvottelun aikana. Arvon on oltava välillä 10-30.</string><string name="invalid_fps">Virheellinen videokehysten lähetystaajuus \'%1$d\'</string><string name="user_agent">User Agent</string><string name="user_agent_help">Räätälöity SIP sanoman User-Agent otsikkokentän arvo</string><string name="invalid_user_agent">Virheellinen User-Agent otsikkokentän arvo</string><string name="contacts_help">Valitsee, käytetäänkö Baresip-yhteystietoja, Android-yhteystietoja vai molempia. Jos molempia käytetään ja molemmissa yhteystiedoissa on samanniminen yhteystieto, valitaan baresip-yhteystieto.</string><string name="both">Molemmat</string><string name="debug">Lokiviestit</string><string name="debug_help">Jos merkitty, baresip tuottaa debug- ja info-tason Logcat-viestejä.</string><string name="sip_trace">SIP-sanomien jäljitys</string><string name="sip_trace_help">Jos merkitty ja jos Lokiviestit on merkitty, Logcat-viestit
|
|||
|
|
sisältävät myös lähetetyt ja vastaanotetut SIP-sanomat. On aina merkitsemättä sovelluksen
|
|||
|
|
käynnistyessä.</string><string name="reset_config">Palauta oletusasetukset</string><string name="reset_config_help">Jos merkitty, oletusasetukset palautetaan, kun
|
|||
|
|
baresip seuraavan kerran käynnistetään.
|
|||
|
|
</string><string name="reset_config_alert">Oletko varma, että haluat palauttaa oletusasetukset\?</string><string name="reset">Palauta</string><string name="read_cert_error">Tiedoston \'cert.pem\' luku epäonnistui.</string><string name="read_ca_certs_error">Tiedoston \'ca_certs.crt\' epäonnistui.</string><string name="config_restart">baresip täytyy käynnistää uudelleen, jotta saat uudet asetukset
|
|||
|
|
käyttöön. Käynnistä nyt\?
|
|||
|
|
</string><string name="consent_request">Suostumuspyyntö</string><string name="contacts_consent">Jos Androidin yhteystiedot on valittu, voit käyttää niitä puheluissa ja viesteissä SIP ja tel URI:en ohella. Baresip ei talleta Androidin yhteystietoja eikä jaa niitä kenellekään. Jotta Baresip saisi käyttöönsä Androidin yhteystiedot, Google vaatii, että annat siihen suostumuksen hyväksymällä Baresip-sovelluksen <a href="https://github.com/juha-h/baresip-studio/blob/master/PrivacyPolicy.txt">yksityisyyskäytännöt</a>.</string><string name="contact">Yhteystieto</string><string name="new_contact">Uusi yhteystieto</string><string name="contact_name">Nimi</string><string name="sip_or_tel_uri">SIP tai tel URI</string><string name="user_domain_or_number">käyttäjä@domain tai puhelinnumero</string><string name="favorite">Suosikki</string><string name="invalid_contact">Virheellinen yhteystiedon nimi \'%1$s\'</string><string name="contact_already_exists">Yhteystieto \'%1$s\' on jo olemassa.</string><string name="invalid_contact_uri">Virheellinen SIP:n URI</string><string name="android_contact_help">Jos merkitty, tämä yhteystieto lisätään Androidin yhteystietoihin.</string><string name="avatar_image">Profiilikuva</string><string name="contacts">Yhteystiedot</string><string name="contact_action_question">Haluatko soittaa tai lähettää
|
|||
|
|
viestin ositteeseen \'%1$s\'\?</string><string name="send_message">Lähetä viesti</string><string name="contact_delete_question">Haluatko poistaa yhteystiedon \'%1$s\'\?</string><string name="contacts_exceeded">Yhteystietojesi enimmäismäärä %1$d on ylittynyt.</string><string name="alert">Varoitus</string><string name="info">Tieto</string><string name="notice">Huomio</string><string name="cancel">Peruuta</string><string name="ok">OK</string><string name="yes">Kyllä</string><string name="no">Ei</string><string name="accept">Hyväksy</string><string name="deny">Estä</string><string name="user_id">Käyttäjätunnus</string><string name="password">Salasana</string><string name="add">Lisää</string><string name="delete">Poista</string><string name="edit">Muokkaa</string><string name="send">Lähetä</string><string name="status">Tila</string><string name="error">Virhe</string><string name="help">Apua</string><string name="confirmation">Vahvistus</string><string name="anonymous">Anonyymi</string><string name="unknown">Tuntematon</string><string name="invalid_sip_or_tel_uri">Virheellinen SIP tai puh. URI \'%1$s\'</string><string name="baresip" translatable="false">baresip</string><string name="android" translatable="false">Android</string><string name="backup">Tallenna</string><string name="restore">Palauta</string><string name="about">Tietoja</string><string name="restart">Käynnistä uudelleen</string><string name="quit">Lopeta</string><string name="outgoing_call_to_dots">Puhelu ulos …</string><string name="incoming_call_from_dots">Puhelu sisään …</string><string name="diverted_by_dots">Siirtäjä …</string><string name="transferring_call_to_dots">Puhelun siirto …</string><string name="invalid_sip_uri">Virheellinen SIP URI \'%1$s\'</string><string name="no_telephony_provider">Tilille \'%1$s\' ei ole konfiguroitu puhelinpalvelun
|
|||
|
|
tarjoajaa</string><string name="callee">Puhelun kohde</string><string name="hangup">Lopeta</string><string name="video_call">Videopuhelu</string><string name="video_request">Videopyyntö</string><string name="allow_video">Salli video videon lähetys ja vastaanotto puhelussa \'%1$s\'\?</string><string name="allow_video_send">Salli video videon lähetys puhelussa \'%1$s\'\?</string><string name="allow_video_recv">Salli video videon vastaanotto puhelussa \'%1$s\'\?</string><string name="hold">Puhelu pitoon/pidosta</string><string name="call_is_on_hold">Puhelu on pidossa</string><string name="mic">Mikrofoni päälle/pois</string><string name="rec_in_call">Tallennus voidaan asettaa päälle tai pois vain silloin, kun puhelu
|
|||
|
|
ei ole yhdistetty</string><string name="call_transfer">Puhelun siirto</string><string name="blind">Sokeasti</string><string name="attended">Osallistu</string><string name="transfer_destination">Siirron kohde</string><string name="choose_destination_uri">Valitse kohteen URI</string><string name="transfer">Siirto</string><string name="transfer_failed">Siirto epäonnistui</string><string name="dtmf">DTMF</string><string name="call_info">Puhelutiedot</string><string name="call_info_not_available">Ei saatavilla</string><string name="duration">Kesto: %1$d (sek)</string><string name="codecs">Koodekit</string><string name="rate">Nykyinen nopeus: %1$s (Kbit/s)</string><string name="average_rate">Keskinopeus: %1$s (Kbit/s)</string><string name="packets">Paketit</string><string name="lost">Hukkunut</string><string name="jitter">Vaihtelu: %1$s (ms)</string><string name="voicemail">Puheposti</string><string name="voicemail_messages">Puhepostiviestit</string><string name="you_have">Sinulla on</string><string name="one_new_message">yksi uusi viesti</string><string name="new_messages">uutta viestiä</string><string name="one_old_message">yksi vanha viesti</string><string name="old_messages">vanhaa viestiä</string><string name="and">ja</string><string name="no_messages">Sinulla ei ole viestejä</string><string name="listen">Kuuntele</string><string name="messages">Viestit</string><string name="dialpad">Numeronäppäimistö</string><string name="call_already_active">Sinulla on jo puhelu käynnissä.</string><string name="start_failed">Sovelluksen käynnistäminen epäonnistui. Tämä voi johtua virheellisestä
|
|||
|
|
Asetukset arvosta. Tarkista Kuunteluosoite, TLS-varmennintiedosto ja TLS CA-tiedosto.
|
|||
|
|
Sen jälkeen käynnistä baresip uudelleen.</string><string name="registering_failed">Tilin \'%1$s\' rekisteröinti epäonnistui.</string><string name="verify">Todennuspyyntö</string><string name="verify_sas">Todennatko SAS:n <%1$s>\?</string><string name="transfer_request">Siirtopyyntö</string><string name="transfer_request_query">Hyväksytkö tämän puhelun siirron kohteeseen \'%1$s\'\?</string><string name="call_request">Soittopyyntö</string><string name="call_request_query">Hyväksytkö pyynnön soittaa kohteeseen \'%1$s\'\?</string><string name="redirect_notice">Automaattinen uudelleenohjaus kohteeseen \'%1$s\'\</string><string name="redirect_request">Uudelleenohjauspyyntö</string><string name="redirect_request_query">Hyväksytkö puhelun uudelleenohjauksen kohteeseen \'%1$s\'\?</string><string name="call_failed">Puhelu epäonnistui</string><string name="call_closed">Puhelu on päättynyt</string><string name="call_not_secure">Tämä puhelu EI ole turvallinen!</string><string name="peer_not_verified">Tämä puhelu on turvallinen, mutta kohdetta ei ole
|
|||
|
|
todennettu!</string><string name="call_is_secure">Tämä puhelu on turvallinen ja kohde on
|
|||
|
|
todennettu! Haluatko poistaa todennuksen\?</string><string name="unverify">Poista todennus</string><string name="backed_up">Sovelluksen tiedot (äänityksiä lukuunottamatta)
|
|||
|
|
talletettiin tiedostoon \'%1$s\'. Android versiossa 9 tiedosto löytyy
|
|||
|
|
Download-kansiosta.</string><string name="backup_failed">Sovelluksen tietojen talletus
|
|||
|
|
tiedostoon \'%1$s\' epäonnistui. Android versiossa 9 tarkista
|
|||
|
|
Asetukset → Sovellukset → baresip → Käyttöluvat → Tallennustila.</string><string name="restart_request">Uudelleenkäynnistyspyyntö</string><string name="restored">Sovelluksen tiedot palautettiin. baresip pitää käynnistää
|
|||
|
|
uudelleen. Käynnistä uudelleen nyt\?</string><string name="restore_failed">Sovelluksen tietojen palauttaminen epäonnistui. Tarkista, että
|
|||
|
|
annoit oikean salasanan ja että palautustiedosto kuuluu tällä sovellukselle.
|
|||
|
|
Android versiossa 9 tarkista myös Asetukset → Sovellukset → baresip → Käyttöluvat →
|
|||
|
|
Tallennustila ja että tiedosto \'%1$s\' on Download-kansiossa.</string><string name="restore_unzip_failed">Sovelluksen tietojen palauttaminen epäonnistui. Android
|
|||
|
|
versiosta 14 alkaen ei salli tietojen palauttamista, jos ne on tallettu ennen %1$s versioa
|
|||
|
|
%2$s.</string><string name="no_notifications">Et voi käyttää tätä sovellusta ilman Ilmoitukset-lupaa.</string><string name="no_calls">Et voi soittaa puheluita tai vastata niihin ilman Mikrofoni-lupaa.</string><string name="no_bluetooth">baresip ei voi havaita Bluetooth-yhteyttä ilman
|
|||
|
|
\"Lähellä olevat laitteet\" lupaa.</string><string name="no_video_calls">Salli Kamera-lupa jotta voit soittaa videopuheluita ja vastata niihin.</string><string name="no_backup">Et voi tallentaa sovelluksen tietoja ilman Tallennustila-lupaa.</string><string name="no_restore">Et voi palauttaa sovelluksen tietoja ilman Tallennustila-lupaa.</string><string name="no_cameras">Sinulla ei ole yhtään tuettua video-kameraa.</string><string name="show_password">Näytä salasana</string><string name="no_network">Ei verkkoyhteyttä!</string><string name="call_details">Puhelutiedot</string><string name="no_android_contacts">Et voi käyttää Androidin yhteystietoja ilman Yhteystiedot-lupaa.</string><string name="audio_focus_denied">Audio fokus evätty!</string><string name="permissions_rationale">Tarvittavat luvat</string><string name="audio_permissions">baresip tarvitsee Mikrofoni-luvan puheluita varten ja Lähellä olevat laitteet -luvan Bluetooth-mikrofonin/kaiuttimen havaitsemista varten ja Ilmoitukset-luvan ilmoitusten lähettämistä varten.</string><string name="audio_and_video_permissions">baresip+ tarvitsee Mikrofoni-luvan puheluita varten, Kamera-luvan videopuheluita varten ja Lähellä olevat laitteet -luvan Bluetooth-mikrofonin/kaiuttimen havaitsemista varten ja Ilmoitukset-luvan ilmoitusten näyttämistä varten.</string></file><file path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\values-fr\strings.xml" qualifiers="fr"><string name="transfer">Transférer</string><string name="video_request">Demande de vidéo</string><string name="video_call">Appel vidéo</string><string name="hangup">Raccrocher</string><string name="transferring_call_to_dots">Transfert d’appel à…</string><string name="incoming_call_from_dots">Appel entrant de…</string><string name="outgoing_call_to_dots">Appel sortant à…</string><string name="quit">Quitter</string><string name="restart">Redémarrer</string><string name="about">À propos</string><string name="restore">Restaurer</string><string name="backup">Sauvegarder</string><string name="confirmation">Confirmation</string><string name="help">Aide</string><string name="error">Erreur</string><string name="status">Statut</string><string name="send">Envoyer</string><string name="edit">Modifier</string><string name="delete">Supprimer</string><string name="add">Ajouter</string><string name="password">Mot de passe</string><string name="deny">Refuser</string><string name="accept">Accepter</string><string name="no">Non</string><string name="yes">Oui</string><string name="ok">OK</string><string name="cancel">Annuler</string><string name="info">Infos</string><string name="send_message">Envoyer le message</string><string name="contacts">Contacts</string><string name="contact_already_exists">Le contact « %1$s » existe déjà.</string><string name="invalid_contact">Nom de contact invalide « %1$s »</string><string name="contact_name">Nom</string><string name="new_contact">Nouveau contact</string><string name="contact">Contact</string><string name="debug">Débogage</string><string name="dns_servers">Serveurs DNS</string><string name="start_automatically">Démarrer automatiquement</string><string name="configuration">Paramètres</string><string name="delete_chats">Supprimer</string><string name="you">Vous</string><string name="today">Aujourd’hui</string><string name="sending_failed">Échec de l’envoi du message</string><string name="add_contact">Ajouter un contact</string><string name="new_message">Nouveau message</string><string name="delete_history_alert">Voulez-vous supprimer l’historique des appels du compte « %1$s » \?</string><string name="enable_history">Activer</string><string name="disable_history">Désactiver</string><string name="delete_history">Supprimer</string><string name="calls_call_message_question">Voulez-vous appeler ou envoyer un message à « %1$s » \?</string><string name="calls_call">appel</string><string name="calls_calls">appels</string><string name="call">Appel</string><string name="call_history">Historique des appels</string><strin
|
|||
|
|
<![CDATA[
|
|||
|
|
<h1>Baresip library based SIP User Agent</h1>
|
|||
|
|
<p>Juha Heinanen <jh@tutpro.com></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 <jh@tutpro.com></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$s です</string><string name="stun_username">STUN/TURNユーザー名</string><st
|
|||
|
|
</string><string name="answer_mode_help">着信した電話にどのように応答するかを選択します。</string><string name="start_automatically_help">チェックを入れると、デバイスの再起動後に自動的にbaresipが起動します</string><string name="aec">音響エコーキャンセル</string><string name="aec_extended_filter_help">チェックを入れると、エコーキャンセルは拡張フィルタを使用しています。</string><string name="default_call_volume_help">設定されている場合、デフォルトの通話音声の音量は1～10段階です</string><string name="dark_theme_help">ダークテーマを強制する</string><string name="debug_help">チェックを入れると、デバッグおよび情報レベルのログメッセージをLogcatに提供します。</string><string name="reset_config_help">チェックを入れると、設定は工場出荷時のデフォルト値にリセットされます。</string><string name="contact_action_question"> %1$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 <%1$s> を検証しますか\?</string><string name="transfer_request_query"> %1$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$s にバックアップされました</string><string name="no_calls">マイクへの権限付与がなく電話をかけたり、応答したりすることはできません。</string></file><file path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\values-ko\strings.xml" qualifiers="ko"><string name="app_name" translatable="false">ritosip</string><string name="app_name_plus" translatable="false">ritosip</string><string name="about_title">ritosip 소개</string><string name="about_title_plus">ritosip 소개</string><string name="about_text">
|
|||
|
|
<![CDATA[
|
|||
|
|
<h1>Ritosip 라이브러리를 기반으로 한 SIP 사용자 에이전트</h1>
|
|||
|
|
<p>Juha Heinanen <jh@tutpro.com></p>
|
|||
|
|
<p>버전 %1$s</p>
|
|||
|
|
<h2>사용 방법</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li>baresip 설정에서 기본값이 적절한지 확인하세요 (항목 제목을 터치하면 도움말 제공).</li>
|
|||
|
|
<li>계정 메뉴에서 하나 이상의 계정을 생성하세요 (다시 한 번 항목 제목을 터치하면 도움말 제공).</li>
|
|||
|
|
<li>계정의 등록 상태는 색깔 점으로 표시됩니다: 초록색 (등록 성공), 노란색 (등록 진행 중), 빨간색 (등록 실패), 흰색 (등록되지 않음).</li>
|
|||
|
|
<li>점(●)을 터치하면 계정 설정으로 바로 이동합니다.</li>
|
|||
|
|
<li>아래로 스와이프하면 현재 표시된 계정을 다시 등록합니다.</li>
|
|||
|
|
<li>길게 터치하면 계정 등록을 활성화하거나 비활성화할 수 있습니다.</li>
|
|||
|
|
<li>좌우로 스와이프하면 다른 계정으로 전환됩니다.</li>
|
|||
|
|
<li>이전 통화 상대를 다시 선택하려면 통화 아이콘을 터치하세요 (수신자가 비어 있을 경우).</li>
|
|||
|
|
<li>통화 및 메시지 상대를 길게 터치하면 연락처에 추가할 수 있습니다.</li>
|
|||
|
|
<li>길게 터치하면 통화 기록, 채팅, 메시지, 연락처를 삭제할 수 있습니다.</li>
|
|||
|
|
<li>연락처 아이콘을 터치하거나 길게 터치하면 프로필 이미지를 추가/제거할 수 있습니다.</li>
|
|||
|
|
<li>더 많은 정보를 보려면 <a href="https://github.com/juha-h/baresip-studio/wiki">Wiki</a>를 방문하세요.</li>
|
|||
|
|
</ul>
|
|||
|
|
<h2>개인정보 보호정책</h2>
|
|||
|
|
개인정보 보호정책은 <a href="https://raw.githubusercontent.com/juha-h/baresip-studio/master/PrivacyPolicy.txt">여기</a>에서 확인할 수 있습니다.
|
|||
|
|
<h2>소스 코드</h2>
|
|||
|
|
소스 코드는 <a href="https://github.com/juha-h/baresip-studio">GitHub</a>에서 확인할 수 있으며, 문제 보고도 가능합니다.
|
|||
|
|
<h2>라이선스</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li><b>BSD-3-Clause</b> (다음 예외 포함):</li>
|
|||
|
|
<li><b>Apache 2.0</b> AMR 코덱 및 TLS 보안</li>
|
|||
|
|
<li><b>AGPLv4</b> ZRTP 미디어 암호화</li>
|
|||
|
|
<li><b>GNU LGPL 2.1</b> G.722, G.726, Codec2 코덱</li>
|
|||
|
|
<li><b>GNU GPLv3</b> G.729 코덱</li>
|
|||
|
|
</ul>
|
|||
|
|
]]>
|
|||
|
|
</string><string name="account">계정</string><string name="account_nickname">장치명</string><string name="account_nickname_help">이 앱에서 계정을 식별하는 데 사용될 장치명입니다.</string><string name="nickname">장치명</string><string name="invalid_account_nickname">잘못된 장치명 \'%1$s\'</string><string name="non_unique_account_nickname">이미 존재하는 장치명 \'%1$s\'</string><string name="display_name">표시 이름</string><string name="your_name">당신의 이름</string><string name="display_name_help">발신 요청의 From URI에 표시될 이름입니다.</string><string name="invalid_display_name">잘못된 표시 이름 \'%1$s\'</string><string name="authentication_username">인증 사용자 이름</string><string name="authentication_username_help">SIP 요청의 인증이 필요한 경우 사용할 인증 사용자 이름입니다. 기본값은 계정의 사용자 이름입니다.</string><string name="invalid_authentication_username">잘못된 인증 사용자 이름 \'%1$s\'</string><string name="authentication_password">인증 비밀번호</string><string name="authentication_password_help">최대 64자의 인증 비밀번호입니다. 인증 사용자 이름이 입력되었지만 비밀번호가 입력되지 않은 경우, ritosip이 시작될 때 요청됩니다.</string><string name="invalid_authentication_password">잘못된 인증 비밀번호 \'%1$s\'</string><string name="outbound_proxies">아웃바운드 프록시</string><string name="outbound_proxies_help">SIP 요청을 보낼 때 사용해야 하는 하나 또는 두 개의 프록시 서버의 SIP URI입니다.
|
|||
|
|
두 개를 지정하면 REGISTER 요청이 모두에게 전송되며, 기타 요청은 응답하는 한 곳으로 전송됩니다.
|
|||
|
|
아웃바운드 프록시를 지정하지 않으면 대상 URI의 호스트 부분에 대한 DNS NAPTR/SRV/A 레코드 조회를 기반으로 요청이 전송됩니다.
|
|||
|
|
SIP URI의 호스트 부분이 IPv6 주소인 경우, 반드시 대괄호([])로 감싸야 합니다.
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예제:
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• sip:example.com:5061;transport=tls
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• sip:[2001:67c:223:777::10];transport=tcp
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• sip:192.168.43.50:443;transport=wss
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</string><string name="sip_uri_of_proxy_server">프록시 서버의 SIP URI</string><string name="sip_uri_of_another_proxy_server">다른 프록시 서버의 SIP URI</string><string name="invalid_proxy_server_uri">잘못된 프록시 서버 URI \'%1$s\'</string><string name="register">등록</string><string name="register_help">체크하면 등록이 활성화되며, 지정된 등록 간격에 따라 REGISTER 요청이 전송됩니다.</string><string name="reg_int">등록 간격</string><string name="reg_int_help">ritosip이 REGISTER 요청을 전송하는 간격(초 단위)입니다. 유효한 값은 60에서 3600 사이입니다.</string><string name="invalid_reg_int">잘못된 등록 간격 \'%1$s\'</string><string name="media_nat">미디어 NAT 트래버설</string><string name="media_nat_help">미디어 NAT 트래버설 프로토콜을 선택합니다 (선택 사항). 가능한 선택지는 STUN (Session Traversal Utilities for NAT, RFC 5389) 및 ICE (Interactive Connectivity Establishment, RFC 5245)입니다.</string><string name="stun_server">STUN/TURN 서버</string><string name="stun_server_help">STUN/TURN 서버 URI 형식: scheme:host[:port][\?transport=udp|tcp],
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여기서 scheme은 \'stun\', \'stuns\', \'turn\', \'turns\' 중 하나입니다. STUN 및 ICE 프로토콜의 기본 STUN 서버는 \'stun:stun.l.google.com:19302\'이며, 이는 공개 Google STUN 서버를 가리킵니다. 기본 TURN 서버는 없습니다.
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</string><string name="stun_server_uri">STUN/TURN 서버 URI</string><string name="invalid_stun_server">잘못된 STUN/TURN 서버 URI \'%1$s\'</string><string name="stun_server_default" translatable="false">stun:stun.l.google.com:19302</string><string name="stun_username">STUN/TURN 사용자 이름</string><string name="stun_username_help">STUN/TURN 서버에서 요구하는 경우 입력하는 사용자 이름입니다.</string><string name="invalid_stun_username">잘못된 사용자 이름 \'%1$s\'</string><string name="stun_password">STUN/TURN 비밀번호</string><string name="stun_password_help">STUN/TURN 서버에서 요구하는 경우 입력하는 비밀번호입니다.</string><string name="invalid_stun_password">잘못된 비밀번호 \'%1$s\'</string><string name="media_encryption">미디어 암호화</string><string name="media_encryption_help">미디어 전송 암호화 프로토콜을 선택합니다 (선택 사항).
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• ZRTP (추천) - 통화가 설정된 후 ZRTP 엔드 투 엔드 미디어 암호화 협상을 시도합니다.
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• DTLS-SRTPF - 발신 통화에서 UDP/TLS/RTP/SAVPF를 제공하며, 수신 통화에서 RTP/SAVP, RTP/SAVPF, UDP/TLS/RTP/SAVP 또는 UDP/TLS/RTP/SAVPF 중 하나를 사용합니다.
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• SRTP-MANDF - 발신 통화에서 RTP/SAVPF를 제공하며, 수신 통화에서 이를 필수로 요구합니다.
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• SRTP-MAND - 발신 통화에서 RTP/SAVP를 제공하며, 수신 통화에서 이를 필수로 요구합니다.
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• SRTP - 발신 통화에서 RTP/AVP를 제공하며, 수신 통화에서 RTP/SAVP 또는 RTP/SAVPF를 사용할 경우 이를 허용합니다.
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</string><string name="prefer_ipv6_media">IPv6 미디어 우선 사용</string><string name="prefer_ipv6_media_help">체크하면, 상대방의 미디어 프로토콜을 자동으로 결정할 수 없는 경우 IPv6 미디어 프로토콜을 사용하도록 제안합니다.</string><string name="rtcp_mux">RTCP 멀티플렉싱</string><string name="rtcp_mux_help">체크하면 RTP와 RTCP 패킷이 하나의 포트에서 멀티플렉싱됩니다 (RFC 5761).</string><string name="rel_100">신뢰할 수 있는 임시 응답</string><string name="rel_100_help">체크하면 신뢰할 수 있는 임시 응답(RFC 3262)을 지원함을 나타냅니다.</string><string name="dtmf_mode">DTMF 모드</string><string name="dtmf_mode_help">DTMF 톤(0–9, #, *, A-D)을 전송하는 방식을 선택합니다.</string><string name="dtmf_inband">인밴드 RTP 이벤트</string><string name="dtmf_info">SIP INFO 요청</string><string name="dtmf_auto">인밴드 RTP 또는 SIP INFO</string><string name="answer_mode">응답 모드</string><string name="answer_mode_help">수신 전화를 응답하는 방식을 선택합니다.</string><string name="redirect_mode">리디렉트 모드</string><string name="redirect_mode_help">전화 리디렉트 요청을 자동으로 따를지, 확인을 요청할지를 선택합니다.</string><string name="manual">수동</string><string name="auto">자동</string><string name="voicemail_uri">음성사서함 URI</string><string name="voicemain_uri_help">음성 메시지를 확인하기 위한 SIP URI입니다. 비워두면 음성 메시지(메시지 대기 표시)를 구독하지 않습니다.</string><string name="invalid_voicemail_uri">잘못된 음성사서함 URI \'%1$s\'</string><string name="country_code">국가 코드</string><string name="country_code_help">이 계정의 E.164 국가 코드입니다. 수신 전화 또는 메시지의 발신자 URI 사용자 부분이 \'+\' 기호로 시작하지 않는 전화번호이고, 연락처 검색에 실패한 경우 이 국가 코드가 번호 앞에 추가되어 다시 검색됩니다. 전화번호가 \'0\'으로 시작하는 경우, \'0\'을 제거한 후 국가 코드가 추가됩니다.</string><string name="country_code_hint">+코드</string><string name="invalid_country_code">잘못된 국가 코드 \'%1$s\'</string><string name="telephony_provider">전화 서비스 제공자</string><string name="telephony_provider_help">전화번호로 전화를 걸 때 사용하는 SIP URI의 호스트 부분입니다. 기본값은 계정의 도메인입니다. 입력하지 않으면 이 계정을 사용하여 전화번호로 전화를 걸 수 없습니다.</string><string name="telephony_provider_hint">SIP URI 호스트 부분</string><string name="invalid_sip_uri_hostpart">잘못된 SIP URI 호스트 부분 \'%1$s\'</string><string name="default_account">기본 계정</string><string name="default_account_help">체크하면, ritosip이 시작될 때 이 계정이 기본 계정으로 선택됩니다.</string><string name="accounts">계정</string><string name="new_account">새 계정</string><string name="accounts_help">새 계정을 생성할 때, 계정의 포트 번호 및 전송 프로토콜을 선택적으로 지정할 수 있습니다: <user>@<domain>[:<port>][;transport=udp|tcp|tls].
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포트 번호를 지정하고 전송 프로토콜을 지정하지 않으면 기본값은 UDP입니다.
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포트를 지정하지 않고 전송 프로토콜을 지정하면, 기본 포트는 5060 또는 5061(TLS)로 설정됩니다.
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둘 다 지정되지 않고 아웃바운드 프록시도 없으면, 계정의 등록 서버(있는 경우)는 도메인의 DNS 정보를 기반으로 결정됩니다.
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</string><string name="user_domain">사용자@도메인</string><string name="invalid_aor">잘못된 사용자@도메인[:포트][;transport=udp|tcp|tls] \'%1$s\'</string><string name="account_exists">계정 \'%1$s\'이(가) 이미 존재합니다.</string><string name="account_allocation_failure">새 계정을 할당하지 못했습니다.</string><string name="encrypt_password">비밀번호 암호화</string><string name="decrypt_password">비밀번호 복호화</string><string name="delete_account">계정 \'%1$s\'을(를) 삭제하시겠습니까?</string><string name="answer">응답</string><string name="reject">거절</string><string name="incoming_call_from">수신 전화: %1$s</string><string name="missed_call_from">부재중 전화: %1$s</string><string name="missed_calls">부재중 전화</string><string name="missed_calls_count">%1$d개의 부재중 전화</string><string name="transfer_request_to">통화 전환 요청: %1$s</string><string name="message_from">메시지 발신자: %1$s</string><string name="call_auto_rejected">자동 거부된 전화: \'%1$s\'</string><string name="call_history">통화 기록</string><string name="call_details">통화 상세 정보</string><string name="call">통화</string><string name="calls_calls">통화들</string><string name="calls_call">통화</string><string name="peer">상대</string><string name="direction">방향</string><string name="time">시간</string><string name="calls_duration">통화 시간</string><string name="calls_call_message_question">\'%1$s\'에게 전화를 걸거나 메시지를 보내시겠습니까?</string><string name="calls_add_delete_question">\'%1$s\'을(를) 연락처에 추가하거나 %2$s을(를) 통화 기록에서 삭제하시겠습니까?</string><string name="calls_delete_question">\'%1$s\' %2$s을(를) 통화 기록에서 삭제하시겠습니까?</string><string name="delete_history">삭제</string><string name="disable_history">비활성화</string><string name="enable_history">활성화</string><string name="delete_history_alert">계정 \'%1$s\'의 통화 기록을 삭제하시겠습니까?</string><string name="chat">채팅 메시지</string><string name="chat_with">\'%1$s\'님과 채팅</string><string name="new_message">새 메시지</string><string name="long_message_question">메시지를 삭제하거나 \'%1$s\'을(를) 연락처에 추가하시겠습니까?</string><string name="short_message_question">메시지를 삭제하시겠습니까?</string><string name="add_contact">연락처 추가</string><string name="sending_failed">메시지 전송 실패</string><string name="message_failed">실패</string><string name="chats">채팅 기록</string><string name="today">오늘</string><string name="you">당신</string><string name="new_chat_peer">새 채팅 상대</string><string name="invalid_chat_peer_uri">잘못된 채팅 상대 URI</string><string name="long_chat_question">\'%1$s\'과의 채팅을 삭제하거나 연락처에 추가하시겠습니까?</string><string name="short_chat_question">\'%1$s\'과의 채팅을 삭제하시겠습니까?</string><string name="delete_chats">삭제</string><string name="delete_chats_alert">계정 \'%1$s\'의 채팅 기록을 삭제하시겠습니까?</string><string name="audio_codecs">오디오 코덱</string><string name="audio_codecs_help">우선순위에 따라 정렬된 오디오 코덱 목록입니다. 드래그하여 순서를 변경하고, 오른쪽으로 스와이프하여 활성화 또는 비활성화할 수 있습니다.</string><string name="video_codecs">비디오 코덱</string><string name="video_codecs_help">우선순위에 따라 정렬된 비디오 코덱 목록입니다. 드래그하여 순서를 변경하고, 오른쪽으로 스와이프하여 활성화 또는 비활성화할 수 있습니다.</string><string name="codec_action">재정렬</string><string name="configuration">설정</string><string name="start_automatically">자동 시작</string><string name="start_automatically_help">체크하면 기기가 (재)시작될 때 ritosip이 자동으로 실행됩니
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<a href="https://raw.githubusercontent.com/juha-h/baresip-studio/master/PrivacyPolicy.txt">개인정보 보호정책</a>
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</string><string name="contact">연락처</string><string name="new_contact">새 연락처</string><string name="contact_name">이름</string><string name="sip_or_tel_uri">SIP 또는 전화 URI</string><string name="user_domain_or_number">사용자@도메인 또는 전화번호</string><string name="favorite">즐겨찾기</string><string name="invalid_contact">잘못된 연락처 이름 \'%1$s\'</string><string name="contact_already_exists">연락처 \'%1$s\'이(가) 이미 존재합니다.</string><string name="invalid_contact_uri">잘못된 SIP URI</string><string name="android_contact_help">체크하면 이 연락처가 Android 연락처에 추가됩니다.</string><string name="avatar_image">프로필 이미지</string><string name="contacts">연락처</string><string name="contact_action_question">\'%1$s\'에게 전화를 걸거나 메시지를 보내시겠습니까?</string><string name="send_message">메시지 보내기</string><string name="contact_delete_question">연락처 \'%1$s\'을(를) 삭제하시겠습니까?</string><string name="contacts_exceeded">최대 연락처 수(%1$d)를 초과했습니다.</string><string name="alert">알림</string><string name="info">정보</string><string name="notice">공지</string><string name="cancel">취소</string><string name="ok">확인</string><string name="yes">예</string><string name="no">아니오</string><string name="accept">수락</string><string name="deny">거부</string><string name="user_id">사용자 ID</string><string name="password">비밀번호</string><string name="sip_uri" translatable="false">SIP URI</string><string name="add">추가</string><string name="delete">삭제</string><string name="edit">편집</string><string name="send">보내기</string><string name="status">상태</string><string name="error">오류</string><string name="help">도움말</string><string name="confirmation">확인</string><string name="anonymous">익명</string><string name="unknown">알 수 없음</string><string name="dots" translatable="false">…</string><string name="bullet_item" translatable="false">• %1$s</string><string name="invalid_sip_or_tel_uri">잘못된 SIP 또는 전화 URI \'%1$s\'</string><string name="baresip" translatable="false">baresip</string><string name="android" translatable="false">Android</string><string name="backup">백업</string><string name="restore">복원</string><string name="logcat" translatable="false">Logcat</string><string name="about">정보</string><string name="restart">재시작</string><string name="quit">종료</string><string name="outgoing_call_to_dots">전화 걸기</string><string name="incoming_call_from_dots">수신 전화 …</string><string name="diverted_by_dots">다른 경로로 연결됨 …</string><string name="transferring_call_to_dots">통화 전환 중 …</string><string name="invalid_sip_uri">잘못된 SIP URI \'%1$s\'</string><string name="no_telephony_provider">계정 \'%1$s\'에 전화 서비스 제공자가 없습니다.</string><string name="callee">수신자</string><string name="hangup">통화 종료</string><string name="video_call">영상 통화</string><string name="video_request">영상 요청</string><string name="allow_video">\'%1$s\'와(과) 영상 송수신을 허용하시겠습니까?</string><string name="allow_video_send">\'%1$s\'에게 영상 전송을 허용하시겠습니까?</string><string name="allow_video_recv">\'%1$s\'로부터 영상 수신을 허용하시겠습니까?</string><string name="hold">통화 대기/해제</string><string name="call_is_on_hold">통화가 대기 상태입니다.</string><string name="mic">마이크 켜기/끄기</string><string name="rec_in_call">통화가 연결되지 않은 경우에만 녹음을 켜거나 끌 수 있습니다.</string><string name="call_transfer">통화 전환</string><string name="blind">블라인드</string><string name="attended">참석 전환</string><string name="transfer_destination">전환 대상</string><string name="choose_destination_uri">대상 URI 선택</string><string name="transfer">전환</string><string name="tran
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수신 주소, TLS 인증서 파일 및 TLS CA 파일을 확인한 후 baresip을 다시 시작하세요.
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</string><string name="registering_failed">\'%1$s\' 등록 실패</string><string name="verify">확인 요청</string><string name="verify_sas">SAS <%1$s>를 확인하시겠습니까?</string><string name="transfer_request">전환 요청</string><string name="transfer_request_query">이 전화를 \'%1$s\'로 전환하시겠습니까?</string><string name="call_request">통화 요청</string><string name="call_request_query">\'%1$s\'와(과) 통화를 수락하시겠습니까?</string><string name="redirect_notice">\'%1$s\'로 자동 연결됨</string><string name="redirect_request">리디렉션 요청</string><string name="redirect_request_query">통화를 \'%1$s\'로 리디렉션하는 것을 수락하시겠습니까?</string><string name="call_failed">통화 실패</string><string name="call_closed">통화 종료됨</string><string name="call_not_secure">이 통화는 안전하지 않습니다!</string><string name="peer_not_verified">이 통화는 안전하지만 상대방이 인증되지 않았습니다!</string><string name="call_is_secure">이 통화는 안전하며 상대방이 인증되었습니다!
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상대방 인증을 해제하시겠습니까?</string><string name="unverify">인증 해제</string><string name="backed_up">애플리케이션 데이터(녹음 제외)가 파일 \'%1$s\'에 백업되었습니다. Android 9 버전에서는 이 파일이 다운로드 폴더에 저장됩니다.</string><string name="backup_failed">애플리케이션 데이터를 \'%1$s\' 파일로 백업하는 데 실패했습니다. 앱 권한에서 "저장소" 권한을 확인하세요.</string><string name="restart_request">재시작 요청</string><string name="restored">애플리케이션 데이터가 복원되었습니다. ritosip을 다시 시작해야 합니다. 지금 다시 시작하시겠습니까?</string><string name="restore_failed">애플리케이션 데이터 복원 실패. 올바른 비밀번호를 입력했는지 확인하고, 백업 파일이 이 애플리케이션에서 생성된 것인지 확인하세요. Android 9 버전에서는 앱 권한에서 "저장소" 권한을 확인하고 다운로드 폴더에 \'%1$s\' 파일이 있는지 확인하세요.</string><string name="restore_unzip_failed">애플리케이션 데이터 복원 실패. Android 14 이상 버전에서는 %1$s 버전 %2$s 이전에 백업된 데이터를 복원할 수 없습니다.</string><string name="no_notifications">"알림" 권한이 없으면 이 애플리케이션을 사용할 수 없습니다.</string><string name="no_calls">ritosip에서 음성 통화를 하려면 "마이크" 권한이 필요합니다.</string><string name="no_bluetooth">"주변 장치" 권한이 없으면 ritosip에서 블루투스 연결을 감지할 수 없습니다.</string><string name="no_video_calls">영상 통화를 걸거나 받으려면 "카메라" 권한을 허용하세요.</string><string name="no_backup">"저장소" 권한이 없으면 백업을 생성할 수 없습니다.</string><string name="no_restore">"저장소" 권한이 없으면 백업을 복원할 수 없습니다.</string><string name="no_android_contacts">"연락처" 권한이 없으면 Android 연락처에 접근할 수 없습니다.</string><string name="no_cameras">지원되는 비디오 카메라가 없습니다.</string><string name="show_password">비밀번호 표시</string><string name="no_network">네트워크 연결 없음!</string><string name="audio_focus_denied">오디오 포커스 거부됨!</string><string name="permissions_rationale">권한 안내</string><string name="audio_permissions">ritosip은 음성 통화를 위해 "마이크" 권한이 필요하며, 블루투스 마이크/스피커 감지를 위해 "주변 장치" 권한이 필요합니다. 또한 알림을 게시하려면 "알림" 권한이 필요합니다.</string><string name="audio_and_video_permissions">ritosip은 음성 통화를 위해 "마이크" 권한, 영상 통화를 위해 "카메라" 권한, 블루투스 마이크/스피커 감지를 위해 "주변 장치" 권한, 알림을 게시하려면 "알림" 권한이 필요합니다.</string></file><file path="D:\Workspace\RITO\Develop\RitoSIP\ritosip\app\src\main\res\values-nb-rNO\strings.xml" qualifiers="nb-rNO"><string name="about_title">Om</string><string name="account">Konto</string><string name="display_name">Visningsnavn</string><string name="your_name">Ditt navn</string><string name="display_name_help">Navn (hvis noe) brukt i skjema-URI for utgående forespørsler.</string><string name="authentication_username">Brukernavn for identitetsbekreftelse</string><string name="authentication_username_help">Brukernavn for identitetsbekreftelse av SIP-forespørsler er påkrevd. Forvalgt verdi er kontoens brukernavn.</string><string name="authentication_password">Passord for identitetsbekreftelse</string><string name="authentication_password_help">Passord for identitetsbekreftelse opptil 64 tegn. Hvis det angis, og passord ikke angis, vil det bli anmodet inntastet når baresip startes.</string><string name="outbound_proxies">Utgående mellomtjenere</string><string name="audio_codecs">Lyd-kodeker</string><string name="audio_codecs_help">Liste over støttede lydkodeker i
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\nEksempler:
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\n • sip:foo.com:5060;transport=tls
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\n • sip:[2001:67c:223:777::10]:5060;transport=tcp</string><string name="sip_uri_of_proxy_server">SIP URI for mellomtjeneren</string><string name="sip_uri_of_another_proxy_server">SIP URI til en annen mellomtjener</string><string name="register">Registrer</string><string name="register_help">Skrur på registrering og REGISTER-forespørsler blir sendt hvert tolvte minutt.</string><string name="media_nat">Media NAT-gjennomgang</string><string name="media_nat_help">Velger media NAT-gjennomgangsprotokoll (hvis noen). Mulige valg er STUN (øktgjennomgangsverktøy for NAT, RFC 5389) og ICE (interaktivt tilkoblingstilknytning, RFC 5245).</string><string name="stun_server_help">En STUN-tjener i utførelsen host[:port]. Forvalgt fabrikkverdi er \'stun.l.google.com:19302\', som peker til offentlig Google STUN-tjener. Brukernavn og passord støttes foreløpig ikke.</string><string name="stun_server_default">stun.l.google.com:19302</string><string name="media_encryption_help">Velger krypteringsprotokoll for mediatransport (hvis noen).
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\n • ZRTP (anbefalt) betyr at ZRTP ende-til-ende -forhandling av mediakryptering forsøkes etter at samtalen har blitt opprettet.
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\n • DTLS-SRTPF betyr at UDP/TLS/RTP/SAVPF tilbys i utgående samtale, og at RTP/SAVP, RTP/SAVPF, UDP/TLS/RTP/SAVP, eller UDP/TLS/RTP/SAVPF brukes hvis tilbudt i innkomende anrop.
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\n • SRTP-MANDF betyr at RTP/SAVPF tilbys i utgående anrop, og krever et innkommende anrop.
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\n • SRTP-MAND betyr at RTP/SAVP tilbys i utgående anrop, og kreves i innkommende anrop.
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\n • SRTP betyr at RTP/AVP tilbys i utgående anrop, og at RTP/SAVP eller RTP/SAVPF brukes hvis tilbudt i innkommende anrop.</string><string name="voicemail_uri">Telefonsvarer-URI</string><string name="voicemain_uri_help">SIP-URI for sjekk av telefonsvarermeldinger. Hvis levnet tom, vil telefonsvarermeldinger (melding venter-anvisninger) ikke bli levert.</string><string name="default_account_help">Velger denne kontoen når baresip startes.</string><string name="transfer_request">Samtaleoverføringsforespørsel til</string><string name="calls_delete_question">Ønsker du å slette \"%1$s\" %2$s fra anropshistorikk\?</string><string name="message_failed">Mislyktes</string><string name="short_chat_question">Ønsker du å slette sludringen med \"%1$s\"\?</string><string name="start_automatically_help">Starter baresip automatisk sammen med enheten.</string><string name="listen_address_help">IP-adresse og port i utførelsen \"adresse:port\" som baresip lytter til for innkommende SIP-forespørsler. Hvis en IP-adresse er en IPv6-adresse, må den skrives i klammeparenteser []. IPv4-adresse 0.0.0.0 eller IPv6-adresse [::] får baresip til å lytte til alle tilgjengelige adresser. Hvis levnet tom (fabrikksforvalg), vil baresip lytte til port 5060 for alle tilgjengelige adresser.</string><string name="dns_servers_help">Kommainndelt liste over adresser til DNS-tjenere. Hvis ikke angitt, vil DNS-tjeneradresser hentes dynamisk fra systemet. Hver DNS-adresse er utførelsen \"ip:port\" eller \"ip\". Hvis porten utelates, brukes forvalget 53. Hvis IP-en er en IPv6-adresse, og port også angis, må IP-en skrives i hakeparenteser []. Som etksempel, list \'8.8.8.8:53,[2001:4860:4860::8888]:53\' peker til en IPv4-adresser og IPv6-adresser for offentlige Google DNS-tjenere.</string><string name="failed_to_load_module">Klarte ikke å laste inn modul.</string><string name="aec">Kansellering av akustisk ekko</string><string name="aec_help">Prøver å unngå ekko under samtaler.</string><string name="opus_bit_rate_help">Gjennomsnittlig maksimal bitrate ved bruk av Opus-lydstrøm. Gyldig verdi er 6000-510000. Fabrikksforvalg er 28000.</string><string name="opus_packet_loss">Forventet Opus-pakketap</string><string name="opus_packet_loss_help">Forventet prosentvis pakketap for Opus-lydstrøm. Gyldige verdier fra 0–100. Fabrikksforvalg er 0, som skrur av Opus-fremtidsfeilkorrigering (FEC).</string><string name="default_call_volume">Forvalgt samtalelydstyrke</string><string name="default_call_volume_help">Setter samtalelydstyrken på en skala fra 1-10.</string><string name="debug">Feilretting</string><string name="debug_help">Gjør feilrettings- og infonivå-loggmeldinger tilgjengelige i Logcat.</string><string name="reset_config_help">Tilbakestiller oppsettet til fabrikkforvalgte verdier.</string><string name="contact_already_exists">Kontakten \"%1$s\" finnes allerede.</string><string name="invalid_contact_uri">Ugyldig SIP URI</string><string name="contact_action_question">Ønsker du å ringe eller sende melding til \"%1$s\"\?</string><string name="contacts_exceeded">Ditt har nådd ditt maksimale antall kontakter på %1$d.</string><string name="invalid_sip_uri">Ugyldig SIP URI \'%1$s\'</string><string name="callee">Ringer</string><string name="hangup">Legg på</string><string name="hold">Sett på vent</string><string name="dtmf">Tonesignalering</string><string name="call_info">Samtaleinfo</string><string name="rate">Takt: %1$s</string><string name="voicemail">Telefonsvarer</string><string name="voicemail_messages">Telefonsvarermeldinger</string><string name="dialpad">Nummerskive</string><string name="call_already_active">Du har allerede et aktivt anrop.</string><string name="start_failed">Baresip kunne ikke starte. Dette kan ha oppstått på grunn av en ugyldig oppstartsverdi. Sjekk lytteadressen, TLS-sertifikatsfilen, og TLS CA-filen. Start så programmet på ny.</string><string name="registering_failed">Registrering av %1$s mislyktes.</string><string name="verify_sas">Ønsker du å bekrefte SAS <%1$s>\?</string><string name="call_failed">Anro
|
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\nExemplos:
|
|||
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\n • sip:example.com:5061;transport=tls
|
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\n • sip:[2001:67c:223:777::10];transport=tcp
|
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\n • sip:192.168.43.50:443;transport=wss</string><string name="send_message">Enviar Mensagem</string><string name="account_allocation_failure">Falha ao alocar nova conta.</string><string name="call_info_not_available">Nenhuma informação disponível</string><string name="deny">Recusar</string><string name="delete_history_alert">Quer apagar o histórico de chamadas da conta \'%1$s\'\?</string><string name="add_contact">Adicionar Contato</string><string name="stun_server_help">Um URI do servidor STUN/TURN na forma de scheme:host[:porta][\?transport=udp|tcp]. Onde o scheme é \'stun\', \'stuns\', \'turn\', ou \'turns\'. O Servidor predefinido de fábrica para o STUN e os protocolos ICE são \'stun:stun.l.google.com:19302\' apontando para o servidor STUN público do Google. Não há qualquer servidor STUN predefinido de fábrica.</string><string name="start_failed">Baresip falhou ao iniciar. Isso pode ser devido a um valor inválido das Configurações. Verifique Endereço de Escuta, Ficheiro de Certificado TLS e Ficheiro CA TLS. Em seguida, reinicie o baresip.</string><string name="call_already_active">Já tem uma chamada ativa.</string><string name="calls_call">chamada</string><string name="accounts_help">Quando uma nova conta é criada, a informação do número da porta, da conta e o protocolo de transporte é opcional: <utilizador>@@<domínio>[:<porta>][;transport=udp|tcp|tls]. Caso a <porta> seja informada e o protocolo de transporte não, o padrão será udp. Caso a <porta> não seja informada e o protocolo de transporte seja, a <porta> padrão é 5060 ou 5061 (TLS). Caso nenhum dos dois ou nenhum proxy seja informado, o utilizador da conta (caso haja) é determinado exclusivamente com base nas informações do DNS do domínio.</string><string name="stun_password">Palavra-passe STUN/TURN</string><string name="reset_config">Redefinir os Padrões de Fábrica</string><string name="allow_video">Aceitar o envio e o recebimento de vídeo com \'%1$s\'\?</string><string name="incoming_call_from_dots">Chamada de …</string><string name="add">Adicionar</string><string name="alert">Alerta</string><string name="prefer_ipv6_media">Preferir Mídia IPv6</string><string name="one_new_message">uma nova mensagem</string><string name="contact_delete_question">Quer apagar o contato \'%1$s\'\?</string><string name="default_call_volume">Volume de Chamadas prefinido</string><string name="encrypt_password">Criptografar Palavra-passe</string><string name="new_account">Nova Conta</string><string name="contact_name">Nome</string><string name="chat">Mensagem do Chat</string><string name="contact_action_question">Quer ligar ou enviar mensagem para \'%1$s\'\?</string><string name="tls_ca_file">Ficheiro TLS CA</string><string name="aec_extended_filter_help">Caso esteja marcado, o cancelamento do eco estará utilizando um filtro estendido.</string><string name="sip_uri_of_another_proxy_server">URI SIP de outro Servidor Proxy</string><string name="contact_already_exists">Contato \'%1$s\' já existe.</string><string name="sip_uri_of_proxy_server">URI SIP do Servidor Proxy</string><string name="backup">Cópia de Segurança</string><string name="audio_modules_title">Módulos de Áudio</string><string name="transfer">Transferir</string><string name="status">Estado</string><string name="peer_not_verified">A chama é SEGURA, porém o par NÃO é verificado!</string><string name="disable_history">Desativar</string><string name="long_message_question">Quer apagar a mensagem ou adicionar o ponto \'%1$s\' aos contatos\?</string><string name="voicemain_uri_help">URI SIP para verificar mensagens de correio de voz. Se deixado em branco, as mensagens do correio de voz (Indicação de Mensagem em Espera) não serão assinadas.</string><string name="register_help">Se marcado, o registo é ativado e as solicitações de REGISTO são enviadas no intervalo determinado pelo Intervalo de registo.</string><string name="video_codecs_help">Lista dos codecs de vídeo em ordem de prioridade. Arraste para reordenar, deslize para a direita para ativar ou
|
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\n • ZRTP (recomendado) significa que a negociação de criptografia de mídia ZRTP ponta-a-ponta é tentada depois que a ligação foi estabelecida.
|
|||
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\n • DTLS-SRTPF significa que UDP/TLS/RTP/SAVPF é oferecido em chamadas efetuadas que RTP/SAVP, RTP/SAVPF, UDP/TLS/RTP/SAVP, ou UDP/TLS/RTP/SAVPF é usado se oferecida em chamadas recebidas.
|
|||
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\n • SRTP-MANDF significa que RTP/SAVPF é oferecido em chamadas efetuadas e requerido em chamadas recebidas.
|
|||
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\n • SRTP-MAND significa que RTP/SAVP é oferecido em chamadas efetuadas e requerido em chamadas recebidas.
|
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\n • SRTP significa que RTP/AVP é oferecido em chamadas efetuadas e que RTP/SAVP ou RTP/SAVPF é usado se oferecido em chamadas recebidas.</string><string name="new_messages">novas mensagens</string><string name="about_text"><h1>Biblioteca Baresip baseado em agente de utilizador SIP</h1> <p>Juha Heinanen <jh@tutpro.com></p> <p>Version %1$s</p> <h2>Dicas de Uso</h2> <ul> <li>Verifique se os valores predefinidos nas Configurações atendem as suas necessidades (clique nos títulos para ajuda).</li> <li>Então em Contas, crie uma ou mais contas (novamente clique nos títulos para ajuda).</li> <li>Uma nova conta pode ser parcialmente configurada automaticamente. Consulte <a href=https://github.com/juha-h/baresip-studio/wiki/Automatic-Account-Configuration>Wiki</a> para obter mais informações.</li> <li>O estado de uma conta é mostrado com um ponto colorido: verde (registo bem sucedido), amarelo (registo em progresso), vermelho (registo falhou), branco (registo não foi ativado).</li> <li>Toque nos três pontos para ir direto à configuração da conta.</li> <li>O toque longo na conta atual, ativa ou não o registo da conta.</li> <li>Gestos para esquerda/direita alternam as contas.</li> <li>O ícone na parte superior do ecrã principal alterna o alto-falante.</li> <li>Os ícones na parte inferior da ecrã principal levam ao correio de voz (caso a URI do correio de voz tenha sido configurada para a conta), contatos, mensagens e histórico de chamadas que permitem alternar entre o teclado numérico e o alfanumérico</li> <li>O interlocutor anterior pode ser re-lacionado a tocar no ícone da chamada quando o \"calee\" estiver vazio</li> <li>O gesto de passar o dedo para baixo no ecrã faz com que o ecrã de registo da conta seja exibida novamente.</li> <li>Peers de chamada e mensagens podem ser adicionadas aos contatos com toque longo.</li> <li>Toques longos também podem ser usados para remover chamadas, chats, mensagens e contatos.</li> <li>Toque/toque longo em ícones de contato podem ser usados para adicionar/remover avatar.</li> </ul> <h2>Política de privacidade</h2> A política de privacidade está disponível <a href=https://raw.githubusercontent.com/juha-h/baresip-studio/master/PrivacyPolicy.txt>aqui</a>. <h2>Código Fonte</h2> Código fonte disponível em <a href=https://github.com/juha-h/baresip-studio>GitHub</a>, onde também falhas podem ser reportadas. <h2>Licenças</h2> <ul> <li><b>BSD-3-Clause</b> exceto o seguinte</li> <li><b>Apache 2.0</b> AMR codec, TLS security e criptografia de mídia ZRTP</li> <li><b>LGPL 2.1</b> G722 e codecs G726</li> <b>GNU GPLv3</b> G.729 codec</li> </ul></string><string name="your_name">Seu Nome</string><string name="video_size">Tamanho do Quadro do Vídeo</string><string name="restore_failed">Falha ao restaurar os dados da aplicação. Verifique se inseriu a palavra-passe correta e que o ficheiro de backup seja deste aplicação, Nas versões do Android 9 e anteriores, verifique também em Apps → baresip → Permissões → Armazenamento e se o ficheiro \'%1$s\' existe na pasta Download.</string><string name="restart">Reiniciar</string><string name="authentication_username">Nome de Utilizador de Autenticação</string><string name="messages">Mensagens</string><string name="authentication_username_help">Nome de utilizador de autenticação se a autenticação de solicitações SIP for necessária. O valor predefinido é o nome de utilizador da conta.</string><string name="yes">Sim</string><string name="media_nat_help">Selecione o protocolo de NAT transversal de mídia (se houver). Escolhas possíveis são STUN (Session Traversal Utilities para NAT, RFC 5389) e ICE (Interactive Connectivity Establishment, RFC 5245).</string><string name="invalid_stun_server">Servidor URI STUN/TURN Inválido \'%1$s\'</string><string name="invalid_authentication_username">Nome de Utilizador de Aut
|
|||
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\n • ZRTP (recomendado) significa que a negociação de criptografia de mídia ZRTP ponta-a-ponta é tentada depois que a ligação foi estabelecida.
|
|||
|
|
\n • DTLS-SRTPF significa que UDP/TLS/RTP/SAVPF é oferecido em chamadas efetuadas que RTP/SAVP, RTP/SAVPF, UDP/TLS/RTP/SAVP, ou UDP/TLS/RTP/SAVPF é usado se oferecida em chamadas recebidas.
|
|||
|
|
\n • SRTP-MANDF significa que RTP/SAVPF é oferecido em chamadas efetuadas e requerido em chamadas recebidas.
|
|||
|
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\n • SRTP-MAND significa que RTP/SAVP é oferecido em chamadas efetuadas e requerido em chamadas recebidas.
|
|||
|
|
\n • SRTP significa que RTP/AVP é oferecido em chamadas efetuadas e que RTP/SAVP ou RTP/SAVPF é usado se oferecido em chamadas recebidas.</string><string name="media_encryption">Criptografia de Mídia</string><string name="invalid_stun_server">Servidor URI STUN/TURN Inválido \'%1$s\'</string><string name="stun_server_help">Um URI do servidor STUN/TURN na forma de scheme:host[:porta][\?transport=udp|tcp]. Onde o scheme é \'stun\', \'stuns\', \'turn\', ou \'turns\'. O Servidor predefinido de fábrica para o STUN e os protocolos ICE são \'stun:stun.l.google.com:19302\' apontando para o servidor STUN público do Google. Não há qualquer servidor STUN predefinido de fábrica.</string><string name="stun_server">Servidor STUN/TURN</string><string name="media_nat_help">Selecione o protocolo de NAT transversal de mídia (se houver). Escolhas possíveis são STUN (Session Traversal Utilities para NAT, RFC 5389) e ICE (Interactive Connectivity Establishment, RFC 5245).</string><string name="media_nat">NAT Transversal de Mídia</string><string name="audio_codecs_help">Lista dos codecs de áudio em ordem de prioridade. Arraste para reordenar, deslize para a direita para ativar ou desativar.</string><string name="audio_codecs">Codecs de Áudio</string><string name="register_help">Se marcado, o registro é ativado e as solicitações de REGISTRO são enviadas no intervalo determinado pelo Intervalo de registro.</string><string name="register">Registro</string><string name="invalid_proxy_server_uri">URI do Servidor Proxy Inválido \'%1$s\'</string><string name="sip_uri_of_another_proxy_server">URI SIP de outro Servidor Proxy</string><string name="sip_uri_of_proxy_server">URI SIP do Servidor Proxy</string><string name="outbound_proxies_help">URI SIP de um ou dois proxies que precisam ser usados para enviar solicitações. Se dois forem informados, requisições de REGISTRO serão enviados para ambos e as outras requisições serão enviadas para aquele que respondeu. Se nenhum proxy foi informado, requisições serão enviadas baseadas em DNS NAPTR/SRV/A pesquisa de registro da URI do host do destinatário. Se o host da URI SIP for um endereço IPv6, o endereço precisa ser escrito dentro de colchetes [].
|
|||
|
|
\nExemplos:
|
|||
|
|
\n • sip:example.com:5061;transport=tls
|
|||
|
|
\n • sip:[2001:67c:223:777::10];transport=tcp
|
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|
|
\n • sip:192.168.43.50:443;transport=wss</string><string name="outbound_proxies">Próxies de Saída</string><string name="invalid_authentication_password">Senha de Autenticação Inválida \'%1$s\'</string><string name="authentication_password_help">Senha de Autenticação com até 64 caracteres. Se o Nome de Usuário de Autenticação for informado mas a senha não, ela será solicitada quando o baresip for iniciado.</string><string name="authentication_password">Senha de Autenticação</string><string name="invalid_authentication_username">Nome de Usuário de Autenticação Inválido \'%1$s\'</string><string name="authentication_username_help">Nome de usuário de autenticação se a autenticação de solicitações SIP for necessária. O valor padrão é o nome de usuário da conta.</string><string name="authentication_username">Nome de Usuário de Autenticação</string><string name="invalid_display_name">Nome de Exibição Inválido\'%1$s\'</string><string name="display_name_help">Nome (se houver) usado em From URI de solicitações de saída.</string><string name="your_name">Seu Nome</string><string name="display_name">Nome de Exibição</string><string name="account">Conta</string><string name="about_text"><![CDATA[
|
|||
|
|
<h1>Biblioteca Baresip baseado em agente de usuário SIP</h1>
|
|||
|
|
<p>Juha Heinanen <jh@tutpro.com></p>
|
|||
|
|
<p>Versão %1$s</p>
|
|||
|
|
<h2>Dicas de Uso</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li>Verifique se os valores predefinidos nas Configurações atendem as suas necessidades
|
|||
|
|
(clique nos títulos para ajuda).</li>
|
|||
|
|
<li>Então em Contas, crie uma ou mais contas (novamente clique nos títulos para ajuda).</li>
|
|||
|
|
<li>O estado de uma conta é mostrado com um ponto colorido: verde (registro
|
|||
|
|
bem sucedido), amarelo (registro em progresso), vermelho (registro falhou), branco
|
|||
|
|
(registro não foi ativado).</li>
|
|||
|
|
<li>Toque nos três pontos para ir direto para a configuração da conta.</li>
|
|||
|
|
<li>O gesto de deslizar para baixo causa o novo registro da conta exibida no momento.</li>
|
|||
|
|
<li>Um toque longo na conta exibida no momento ativa ou desativa o registro da conta.</li>
|
|||
|
|
<li>Gestos para a esquerda/direita alternam as contas.</li>
|
|||
|
|
<li>O participante da chamada anterior pode ser selecionado novamente tocando no ícone
|
|||
|
|
de chamada quando o "Callee" estiver vazio.</li>
|
|||
|
|
<li>Os pares de chamadas e mensagens podem ser adicionados aos contatos por meio de toques longos.</li>
|
|||
|
|
<li>Os toques longos também podem ser usados para remover chamadas, bate-papos, mensagens e contatos.</li>
|
|||
|
|
<li>Toque/toque longo no ícone do contato pode ser usado para instalar/remover o avatar da imagem.</li>
|
|||
|
|
<li>Consulte a <a href="https://github.com/juha-h/baresip-studio/wiki">Wiki</a> para obter
|
|||
|
|
mais informações.</li>
|
|||
|
|
</ul>
|
|||
|
|
<h2>Política de privacidade</h2>
|
|||
|
|
A política de privacidade está disponível <a href="https://raw.githubusercontent.com/juha-h/baresip-studio/master/PrivacyPolicy.txt">aqui</a>.
|
|||
|
|
<h2>Código fonte</h2>
|
|||
|
|
O código fonte está disponível no <a href="https://github.com/juha-h/baresip-studio">GitHub</a>,
|
|||
|
|
onde os problemas encontrados também podem ser reportados.
|
|||
|
|
<h2>Licenças</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li><b>Cláusula BSD-3</b> exceto os seguintes:</li>
|
|||
|
|
<li><b>Apache 2.0</b>, codecs AMR e segurança TLS</li>
|
|||
|
|
<li><b>AGPLv4</b>, criptografia de mídia ZRTP</li>
|
|||
|
|
<li><b>GNU LGPL 2.1</b> G.722, G.726, e codecs Codec2</li>
|
|||
|
|
<li><b>GNU GPLv3</b> G.729 codec</li>
|
|||
|
|
</ul>
|
|||
|
|
]]></string><string name="about_title">Sobre o baresip</string><string name="no_cameras">Você não possui câmeras de vídeo compatíveis.</string><string name="no_video_calls">Conceda permissão à \"Câmera\" para fazer ou atender chamadas de vídeo.</string><string name="restart_request">Solicitação para Reinicialização</string><string name="transfer">Transferir</string><string name="allow_video">Aceitar o envio e o recebimento de vídeo com \'%1$s\'\?</string><string name="video_request">Pedido de Vídeo</string><string name="video_call">Chamada de vídeo</string><string name="confirmation">Confirmação</string><string name="video_size_help">Tamanho dos quadros de vídeo que foram transmitidos (largura x altura)</string><string name="video_size">Tamanho do Quadro do Vídeo</string><string name="aec_extended_filter_help">Caso esteja marcado, o cancelamento do eco estará utilizando um filtro estendido.</string><string name="aec_extended_filter">Filtro Estendido do AEC</string><string name="invalid_stun_password">Senha inválida \'%1$s\'</string><string name="stun_password_help">Senha, caso seja necessário pelo servidor STUN/TURN</string><string name="stun_password">Senha STUN/TURN</string><string name="invalid_stun_username">Nome do usuário inválido \'%1$s\'</string><string name="stun_username_help">Nome do usuário, caso seja necessário pelo servidor STUN/TURN</string><string name="stun_username">Nome do Usuário STUN/TURN</string><string name="video_codecs_help">Lista dos codecs de vídeo em ordem de prioridade. Arraste para reordenar, deslize para a direita para ativar ou desativar.</string><string name="video_codecs">Codecs de Vídeo</string><string name="transfer_request_query">Você aceita transferir esta chamada para \'%1$s\'\?</string><string name="transfer_failed">A transferência falhou</string><string name="transfer_destination">Destino da transferência</string><string name="call_transfer">Transferência de chamada</string><string name="transfer_request_to">Chamada de solicitação de transferência para</string><string name="about_text_plus"><![CDATA[
|
|||
|
|
<h1>Biblioteca Baresip baseado em agente de usuário SIP com vídeo chamadas</h1>
|
|||
|
|
<p>Juha Heinanen <jh@tutpro.com></p>
|
|||
|
|
<p>Versão %1$s</p>
|
|||
|
|
<h2>Dicas de uso</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li>Verifique se os valores padrão nas configurações do baresip+\ atendem às suas necessidades
|
|||
|
|
(toque nos títulos dos itens para obter ajuda).</li>
|
|||
|
|
<li>Então em Contas, crie uma ou mais contas (novamente clique nos títulos para ajuda).</li>
|
|||
|
|
<li>O estado de uma conta é mostrado com um ponto colorido: verde (registro
|
|||
|
|
bem sucedido), amarelo (registro em progresso), vermelho (registro falhou), branco
|
|||
|
|
(registro não foi ativado).</li>
|
|||
|
|
<li>Toque nos três pontos para ir direto para a configuração da conta.</li>
|
|||
|
|
<li>O gesto de deslizar para baixo causa o novo registro da conta exibida no momento.</li>
|
|||
|
|
<li>Um toque longo na conta exibida no momento ativa ou desativa o registro da conta.</li>
|
|||
|
|
<li>Gestos para a esquerda/direita alternam as contas.</li>
|
|||
|
|
<li>O participante da chamada anterior pode ser selecionado novamente tocando no ícone
|
|||
|
|
de chamada quando o "Callee" estiver vazio.</li>
|
|||
|
|
<li>Os pares de chamadas e mensagens podem ser adicionados aos contatos por meio de toques longos.</li>
|
|||
|
|
<li>Os toques longos também podem ser usados para remover chamadas, bate-papos, mensagens e contatos.</li>
|
|||
|
|
<li>Toque/toque longo no ícone do contato pode ser usado para instalar/remover o avatar da imagem.</li>
|
|||
|
|
<li>Consulte a <a href="https://github.com/juha-h/baresip-studio/wiki">Wiki</a> para obter
|
|||
|
|
mais informações.</li>
|
|||
|
|
</ul>
|
|||
|
|
<h2>Problemas conhecidos</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li>Nas chamadas com vídeo, o dispositivo precisa ser mantido no modo paisagem,
|
|||
|
|
girado 90 graus para a esquerda em relação à orientação retrato.</li>
|
|||
|
|
<li>A própia visualização automática não é exibida corretamente quando o
|
|||
|
|
fluxo de vídeo é somente envio.</li>
|
|||
|
|
</ul>
|
|||
|
|
<h2>Política de privacidade</h2>
|
|||
|
|
A política de privacidade está disponível <a href="https://raw.githubusercontent.com/juha-h/baresip-studio/video/PrivacyPolicy.txt">aqui</a>.
|
|||
|
|
<h2>Código fonte</h2>
|
|||
|
|
O código fonte está disponível no <a href="https://github.com/juha-h/baresip-studio">GitHub</a>,
|
|||
|
|
onde os problemas encontrados também podem ser reportados.
|
|||
|
|
<h2>Licenças</h2>
|
|||
|
|
<ul>
|
|||
|
|
<li><b>Cláusula BSD-3</b> exceto os seguintes:</li>
|
|||
|
|
<li><b>Apache 2.0</b>, codecs AMR e segurança TLS</li>
|
|||
|
|
<li><b>AGPLv4</b>, criptografia de mídia ZRTP</li>
|
|||
|
|
<li><b>GNU LGPL 2.1</b> G.722, G.726, e codecs Codec2</li>
|
|||
|
|
<li><b>GNU GPLv3</b> codec G.729</li>
|
|||
|
|
<li><b>GNU GPLv2</b> codecs H.264 e H.265</li>
|
|||
|
|
<li><b>AOMedia</b> codec AV1</li>
|
|||
|
|
</ul>
|
|||
|
|
]]></string><string name="about_title_plus">Sobre o baresip+</string><string name="stun_server_uri">Servidor URI STUN/TURN</string><string name="sip_trace_help">Se selecionado em conjunto com Debug, as mensagens Logcat incluem também o pedido SIP e o rastreamento da resposta. A opção não é ativada automaticamente durante a inicialização do baresip.</string><string name="sip_trace">Rastreio SIP</string><string name="allow_video_recv">Aceitar o recebimento de vídeo a partir do \'%1$s\'\?</string><string name="allow_video_send">Aceitar o envio de vídeo para \'%1$s\'\?</string><string name="missed_call_from">Ligação perdida de</string><string name="dark_theme_help">Impor o uso do tema escuro na tela</string><string name="dark_theme">Tema Escuro</string><string name="android_contact_help">Se for marcado, este contato será adicionado nos contatos do Android.</string><string name="verify_server_help">Se estiver marcado, o baresip verifica os certificados TLS do SIP User Agent e o SIP Proxy Servers quando o transporte TLS for usado.</string><string name="verify_server">Verifique os certificados do servidor</string><string name="dtmf_info">Solicitações INFO SIP</string><string name="dtmf_inband">Eventos na Banda RTP</string><string name="dtmf_mode_help">Selecione como os tons DTMF 0–9, #, *, e A-D serão enviados.</string><string name="dtmf_mode">Modo DTMF</string><string name="reset">Redefinir</string><string name="reset_config_alert">Tem certeza de que deseja redefinir as configurações para os valores predefinidos de fábrica\?</string><string name="mic">Microfone ligado/desligado</string><string name="no_network">Nenhuma conexão de rede!</string><string name="missed_calls_count">%1$d chamadas perdidas</string><string name="missed_calls">Chamadas perdidas</string><string name="avatar_image">Imagem do perfil</string><string name="no_restore">Você não pode restaurar o backup sem a permissão de \"Armazenamento\".</string><string name="no_backup">Você não pode criar backups sem a permissão de \"Armazenamento\".</string><string name="battery_optimizations">Otimizações da bateria</string><string name="battery_optimizations_help">Desative as otimizações da bateria (recomendado) caso queira reduzir a probabilidade do Android restringir o acesso do baresip à rede ou coloque o baresip no modo de espera.</string><string name="lost">Perda</string><string name="average_rate">Taxa média: %1$s (Kbits/s)</string><string name="packets">Pacotes</string><string name="call_is_on_hold">Chamada em espera</string><string name="jitter">Variação: %1$s (ms)</string><string name="audio_settings">Configurações do áudio</string><string name="calls_duration">Duração</string><string name="peer">Par</string><string name="call_details">Detalhes da chamada</string><string name="direction">Direção</string><string name="time">Hora</string><string name="telephony_provider">Operadora de telefonia</string><string name="telephony_provider_hint">Parte do host SIP URI</string><string name="invalid_sip_uri_hostpart">Parte inválida do host SIP URI \'%1$s\'</string><string name="sip_or_tel_uri">SIP ou tel. URI</string><string name="user_domain_or_number">usuário@domínio ou o número de telefone</string><string name="telephony_provider_help">SIP URI host é parcialmente usado nas chamadas para números de telefone. O padrão de fábrica é domínio da conta. Caso não seja informada, essa conta não pode ser usada para ligar para os números de telefone.</string><string name="invalid_sip_or_tel_uri">SIP ou tel. URI inválido \'%1$s\'</string><string name="contacts_help">Escolhe se serão usados os contatos do baresip, os contatos do Android ou ambos. Se ambos forem usados e houver um contato com o mesmo nome em ambos os contatos, o contato baresip será escolhido.</string><string name="both">Ambos</string><string name="no_android_contacts">Você não pode acessar os contatos do Android sem a permissão \"Contatos\".</string><string name="blind">Cego</string><string name="attended">Participou</string><string name="anonymou
|
|||
|
|
\n • ZRTP (рекомендуется) означает, что согласование сквозного шифрования данных ZRTP предпринимается после установления соединения.
|
|||
|
|
\n • DTLS-SRTPF означает, что UDP/TLS/RTP/SAVPF предлагается в исходящем вызове и что RTP/SAVP, RTP/SAVPF, UDP/TLS/RTP/SAVP или UDP/TLS/RTP/SAVPF используется, если предлагается во входящем вызов.
|
|||
|
|
\n • SRTP-MANDF означает, что RTP/SAVPF предлагается при исходящем вызове и требуется при входящем вызове.
|
|||
|
|
\n • SRTP-MAND означает, что RTP/SAVP предлагается при исходящем вызове и требуется при входящем вызове.
|
|||
|
|
\n • SRTP означает, что RTP/AVP предлагается в исходящем вызове и что RTP/SAVP или RTP/SAVPF используется, если предлагается во входящем вызове.</string><string name="message_failed">Неудачно</string><string name="message_from">Сообщение от</string><string name="messages">Сообщения</string><string name="new_account">Новый аккаунт</string><string name="new_chat_peer">Новый партнер чата</string><string name="new_contact">Новый контакт</string><string name="new_message">Новое сообщение</string><string name="new_messages">Новые сообщения</string><string name="no">Нет</string><string name="no_messages">Вы не имеете сообщений</string><string name="notice">Оповещение</string><string name="ok">Ок</string><string name="old_messages">старые сообщения</string><string name="one_new_message">одно новое сообщение</string><string name="one_old_message">одно старое сообщение</string><string name="password">Пароль</string><string name="quit">Выход</string><string name="rate">Текущая скорость: %1$s (Кбит/с)</string><string name="register">Регистрация</string><string name="reject">Отклонить</string><string name="reset_config">Сброс к заводским настройкам</string><string name="restart">Перезагрузка</string><string name="restore">Восстановление</string><string name="verify">Запрос на проверку</string><string name="user_id">ID пользователя</string><string name="no_calls">baresip требуется разрешение доступа к микрофону для голосовых вызовов.</string><string name="long_message_question">Хотите удалить сообщения или добавить пользователя \'%1$s\' в контакты?</string><string name="accept">Принять</string><string name="account">Профиль</string><string name="account_allocation_failure">Не могу создать новый аккаунт.</string><string name="account_exists">Профиль \'%1$s\' уже существует.</string><string name="accounts">Профили</string><string name="aec">Подавление акустического эха</string><string name="aec_help">Если выбрано - работает подавление эха.</string><string name="and">и</string><string name="answer_mode">Режим ответа</string><string name="answer_mode_help">Выберите как отвечать на входящие вызовы.</string><string name="backed_up">Данные приложения сохранены в файл \'%1$s\'. В Android версии 9 и ниже файл находится в каталоге загрузок.</string><string name="call_not_secure">Вызов не безопасен!</string><string name="listen_address">Прослушивать адрес</string><string name="outbound_proxies">Исходящие прокси</string><string name="outgoing_call_to_dots">Вызов на …</string><string name="registering_failed">Регистрация `%1$s` не удалась.</string><string name="restored">Данные приложения восстановлены. Нужна перезагрузка baresip. Перезагрузить\?</string><string name="today">Сегодня</string><string name="transfer_request_query">Принять перевод этого вызова на \'%1$s\'\?</string><string name="transfer_request">Запрос перевода на</string><string name="transferring_call_to_dots">Перевод на …</string><string name="unverify">Отменить подтверждение</string><string name="invalid_opus_packet_loss">Неверный процент потерь Opus</string><string name="media_nat">Прохождени
|
|||
|
|
\nПримеры:
|
|||
|
|
\n • sip:fooexample.com:50601;transport=tls
|
|||
|
|
\n • sip:[2001:67c:223:777::10]:5060;transport=tcp
|
|||
|
|
\n • sip:192.168.43.50:443;transport=wss</string><string name="invalid_authentication_username">Неверное имя пользователя для аутентификации \'%1$s\'</string><string name="authentication_username_help">Имя пользователя для аутентификации, если требуется аутентификация SIP-запросов. Значение по умолчанию - имя пользователя учетной записи.</string><string name="invalid_display_name">Неверное отображаемое имя \'%1$s\'</string><string name="invalid_authentication_password">Неверный пароль аутентификации \'%1$s\'</string><string name="transfer_failed">Ошибка передачи</string><string name="transfer_destination">Переадресовать на</string><string name="call_transfer">Переадресация вызова</string><string name="transfer_request_to">Переадресовать вызов</string><string name="about_text_plus"><h1>SIP-клиент на основе библиотеки Baresip с видеозвонками</h1> <p>Juha Heinanen &lt;jh@tutpro.com&gt;</p> <p>Версия %1$s</p> <h2>Подсказки по использованию</h2> <ul> <li>Убедитесь, что значения параметров по умолчанию в настройках baresip+ соответствуют вашим потребностям (касайтесь их названий для получения справки).</li> <li>Затем создайте одну или несколько учётных записей (опять же касайтесь заголовков элементов для получения справки).</li> <li>Новая учётная запись может частично настраиваться автоматически. Смотрите подробности в <a href=https://github.com/juha-h/baresip-studio/wiki/Automatic-Account-Configuration>вики</a>.</li> <li>Состояние регистрации учётной записи отображается цветной точкой: зелёной (регистрация успешна), жёлтой (регистрация выполняется), красной (регистрация не удалась), белой (регистрация отключена).</li> <li>Касание точки открывает параметры учётной записи.</li> <li>Жест смахивания вниз вызывает повторную регистрацию текущей учётной записи.</li> <li>Долгое нажатие на текущей учётной записи включает или отключет её регистрацию.</li> <li>Смахивание влево или вправо переключает между учётными записями.</li> <li>Значок сверху основного экрана переключает динамик.</li> <li>Значки снизу основного экрана открывают голосовую почту (если её URI указан в учётной записи), контакты, сообщения и историю вызовов, а также позволяют переключаться между цифровой и буквенно-цифровой клавиатурами.</li> <li>Предыдущего собеседника можно выбрать касанием значка вызова при пустом поле «Вызываемый».</li> <li>Собеседников в вызовах и сообщениях можно добавлять в контакты долгими нажатиями.</li> <li>Также долгие нажатия можно использовать для удаления вызовов, чатов, сообщений
|
|||
|
|
\nExempel:
|
|||
|
|
\n • sip:example.com:5061;transport=tls
|
|||
|
|
\n • sip:[2001:67c:223:777::10];transport=tcp
|
|||
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\n • sip:192.168.43.50:443;transport=wss</string><string name="outbound_proxies">Utgående proxyservrar</string><string name="invalid_authentication_password">Ogiltigt lösenord för autentisering \'%1$s\'</string><string name="authentication_password_help">Lösenord för autentisering, upp till 64 tecken. Om användarnamn för autentisering är ifyllt, men lösenordet lämnas tomt kommer baresip fråga om lösenord vid uppstart.</string><string name="authentication_password">Lösenord för autentisering</string><string name="authentication_username_help">Användarnamn som används om SIP-förfrågningar kräver autentisering. Förvalt värde är kontots användarnamn.</string><string name="authentication_username">Användarnamn för autentisering</string><string name="invalid_authentication_username">Ogiltigt användarnamn för autentisering \'%1$s\'</string><string name="invalid_display_name">Ogiltigt visat namn \'%1$s\'</string><string name="display_name_help">Namn som, om det anges, används i From-URI i utgående SIP-förfrågningar.</string><string name="your_name">Ditt namn</string><string name="display_name">Visat namn</string><string name="account">Konto</string><string name="about_title_plus">Om baresip+</string><string name="about_title">Om baresip</string><string name="about_text"><h1>SIP-klient baserad på baresip-biblioteket</h1> <p>Juha Heinanen <jh@tutpro.com></p> <p>Version %1$s</p> <h2>Användartips</h2> <ul> <li>Anpassa baresips inställningar efter dina behov (tryck på en inställnings titel för hjälp).</li> <li>Skapa sedan ett eller flera konton (igen, tryck på en inställnings titel för hjälp).</li> <li>Ett nytt konto kan delvis konfigureras automatiskt. Se <a href=https://github.com/juha-h/baresip-studio/wiki/Automatic-Account-Configuration>Wiki</a> för mer information.</li> <li>Registreringsstatus för ett konto visas med en färgad prick: grön (registrering lyckades), gul (registrering pågår), röd (registrering misslyckades), vit (registrering är inte aktiverad).</li> <li>Om du trycker på den färgade pricken tas du till kontots konfiguration.</li> <li>Om du sveper ner över skärmen registrerar det valda kontot om sig.</li> <li>Långt tryck på det valda kontot aktiverar eller inaktiverar registrering för kontot.</li> <li>Svepningar åt vänster/höger växlar mellan konton.</li> <li>Ikonen överst på startsidan slår på/av högtalartelefonfunktionen.</li> <li>Ikonerna längst ner på startsidan tar dig till röstbrevlådan (om URI för röstbrevlåda har angetts för kontot), dina kontakter, meddelanden, samtalshistorik och låter dig växla mellan numeriskt och alfanumeriskt tangentbord.</li> <li>Senast uppringda nummer kan fyllas i igen genom att trycka på ring upp-ikonen när inget nummer att ringa har angetts.</li> <li>Samtal- och meddelandemotparter kan läggas till i dina kontakter genom ett långt tryck på motparten.</li> <li>Långa tryck kan också användas för att ta bort samtal, chattar, meddelanden och kontakter.</li> <li>Genom tryck/långt tryck på en kontakts ikon kan en bild läggas till/tas bort för kontakten.</li> </ul> <h2>Källkod</h2> Källkoden finns tillgänglig på <a href=https://github.com/juha-h/baresip-studio>GitHub</a> där även fel och problem kan rapporteras. <h2>Licenser</h2> <ul> <li><b>BSD-3-Clause</b> förutom följande:</li> <li><b>Apache 2.0</b> AMR-kodek, TLS-säkerhet och ZRTP-mediakryptering</li> <li><b>LGPL 2.1</b> G.722- och G.726-kodekar</li> <li><b>GNU GPLv3</b> G.729-kodek</li> </ul></string><string name="transfer_request_query">Accepterar du att koppla det här samtalet till \'%1$s\'\?</string><string name="audio_modules_title">Ljudmoduler</string><string name="start_failed">Baresip kunde inte starta. Det kan bero på ett ogiltigt värde för en inställning. Kontrollera Adress att lyssna på, TLS certifikat-fil och TLS CA-fil. Starta
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\n • ZRTP (rekommenderad) innebär att ändpunktskryptering via ZRTP försöker förhandlas fram efter att samtalet kopplats upp.
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\n • DTLS-SRTPF innebär att UDP/TLS/RTP/SAVPF erbjuds i utgående samtal och att RTP/SAVP, RTP/SAVPF, UDP/TLS/RTP/SAVP eller UDP/TLS/RTP/SAVPF används om det erbjuds i inkommande samtal.
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\n • SRTP-MANDF innebär att RTP/SAVPF erbjuds i utgående samtal och är ett krav för inkommande samtal.
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\n • SRTP-MAND innebär att RTP/SAVP erbjuds i utgående samtal och är ett krav för inkommande samtal.
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\n • SRTP innebär att RTP/AVP erbjuds i utgående samtal och att RTP/SAVP eller RTP/SAVPF används om de erbjuds i inkommande samtal.</string><string name="tls_certificate_file">TLS-certifikatfil</string><string name="failed_to_set_dns_servers">Kunde inte ställa in DNS-servrar</string><string name="dns_servers_help">Kommaseparerad lista med adresser på DNS-servrar. Om listan lämnas tom hämtas DNS-serveradresser automatiskt från systemet. Varje DNS-adress skrivs på formatet \'ip:port\' eller \'ip\'. Om port utelämnas används port 53. Om ip är en IPv6-adress och en port anges då måste ip skrivas inom hakparenteser []. Som exempel, i listan \'8.8.8.8:53,[2001:4860:4860::8888]:53\' anges IPv4- och IPv6-adresser för Googles publika DNS-servrar.</string><string name="invalid_listen_address">Ogiltig adress att lyssna på</string><string name="listen_address_help">IP-adress och port på formatet \'adress:port\' där baresip lyssnar efter inkommande SIP-förfrågningar. Om IP-adressen är en IPv6-adress måste den skrivas inom hakparenteser []. IPv4-adressen 0.0.0.0 eller IPv6-adressen [::] betyder att baresip lyssnar på alla tillgängliga adresser. Om fältet lämnas tomt (grundinställning) lyssnar baresip på port 5060 på alla tillgängliga adresser.</string><string name="listen_address">Adress att lyssna på</string><string name="start_automatically_help">Kryssa i för att starta baresip automatiskt när enheten startat.</string><string name="delete_chats_alert">Vill du ta bort chatt-historiken för kontot \'%1$s\'\?</string><string name="short_chat_question">Vill du ta bort chatten med \'%1$s\'\?</string><string name="long_chat_question">Vill du ta bort chatten med motparten \'%1$s\' eller lägga till motparten till dina kontakter\?</string><string name="new_chat_peer">Ny chat-motpart</string><string name="chats">Chat-historik</string><string name="message_failed">Misslyckades</string><string name="sending_failed">Kunde inte skicka meddelande</string><string name="long_message_question">Vill du ta bort meddelandet eller lägga till motparten \'%1$s\' till dina kontakter\?</string><string name="decrypt_password">Lösenord för avkryptering</string><string name="encrypt_password">Lösenord för kryptering</string><string name="accounts_help">När ett nytt konto skapas är det möjligt, men inte nödvändigt, att ange port och transportprotokoll för kontot: <användare>@<domän>[:<port>][;transport=udp|tcp|tls]. Om <port> anges men transportprotokoll utelämnas kommer UDP användas som transportprotokoll. Om <port> utelämnas, men transportprotokoll anges används port 5060 eller 5061 (TLS). Om ingendera anges och ingen utgående proxy är angiven kommer registreringsserver för kontot bestämmas enbart baserat på domänens DNS-information.</string><string name="about_text_plus"><h1>SIP-klient med videostöd baserad på baresip-biblioteket</h1> <p>Juha Heinanen &lt;jh@tutpro.com&gt;</p> <p>Version %1$s</p> <h2>Användartips</h2> <ul> <li>Anpassa baresip+:s inställningar efter dina behov (tryck på en inställnings titel för hjälp).</li> <li>Skapa sedan ett eller flera konton (igen, tryck på en inställnings titel för hjälp).</li> <li>Ett nytt konto kan delvis konfigureras automatiskt. Se <a href=https://github.com/juha-h/baresip-studio/wiki/Automatic-Account-Configuration>Wiki</a> för mer information.</li> <li>Registreringsstatus för ett konto visas med en färgad prick: grön (registrering lyckades), gul (registrering pågår), röd (registrering misslyckades), vit (registrering är inte aktiverad).</li> <li>Om du trycker på den färgade pricken tas du till kontots konfiguration.</li> <li>Om du sveper ner över skärmen registrerar det valda kontot om sig.</li> <li>Långt tryck på det valda kontot aktiverar eller inaktiverar registrering för kontot.</li> <li>Svepningar åt vänster/höger växlar mellan konton.</li> <li>Ikonen överst på startsidan slår på/av högtalartelefonfunktionen.&l
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