├── .gitattributes ├── .langignore ├── Backup.php ├── LICENSE ├── README.md ├── Restore.php ├── Webrtc.class.php ├── functions.inc.php ├── functions.inc ├── get_set.php └── hook_core.php ├── i18n ├── cs │ └── LC_MESSAGES │ │ ├── webrtc.mo │ │ └── webrtc.po ├── de_DE │ └── LC_MESSAGES │ │ ├── webrtc.mo │ │ └── webrtc.po ├── es_ES │ └── LC_MESSAGES │ │ ├── webrtc.mo │ │ └── webrtc.po ├── fa_IR │ └── LC_MESSAGES │ │ ├── webrtc.mo │ │ └── webrtc.po ├── fr_FR │ └── LC_MESSAGES │ │ ├── webrtc.mo │ │ └── webrtc.po ├── pl │ └── LC_MESSAGES │ │ ├── webrtc.mo │ │ └── webrtc.po ├── pt_BR │ └── LC_MESSAGES │ │ ├── webrtc.mo │ │ └── webrtc.po ├── ru │ └── LC_MESSAGES │ │ ├── webrtc.mo │ │ └── webrtc.po ├── uk_UA │ └── LC_MESSAGES │ │ ├── webrtc.mo │ │ └── webrtc.po ├── vi │ └── LC_MESSAGES │ │ ├── webrtc.mo │ │ └── webrtc.po ├── webrtc.pot └── zh_CN │ └── LC_MESSAGES │ ├── webrtc.mo │ └── webrtc.po ├── module.xml ├── ucp ├── Webrtc.class.php ├── assets │ ├── images │ │ └── no_user_logo.png │ ├── js │ │ └── global.js │ ├── jssiplibs │ │ ├── sip-0.7.6.min.js │ │ └── sip-0.7.7.min.js │ ├── less │ │ └── bootstrap.less │ └── sounds │ │ ├── US_Ringback.mp3 │ │ ├── US_Ringback.ogg │ │ ├── ring.mp3 │ │ └── ring.ogg └── views │ ├── phone.php │ └── settings.php ├── utests ├── InstallVersionTest.php └── setuptests.php └── views └── ucp_config.php /.gitattributes: -------------------------------------------------------------------------------- 1 | amp_conf/htdocs/admin/assets/js/pbxlib.js merge=ours 2 | module.xml merge=ours 3 | *.po merge=merge-gettext-po 4 | *.pot merge=merge-gettext-po 5 | *.mo merge=ours 6 | -------------------------------------------------------------------------------- /.langignore: -------------------------------------------------------------------------------- 1 | jssip-0.6.30.js 2 | -------------------------------------------------------------------------------- /Backup.php: -------------------------------------------------------------------------------- 1 | $this->dumpTables(), 8 | 'kvstore' => $this->dumpKVStore(), 9 | ]; 10 | $this->addDependency('userman'); 11 | $this->addDependency('certman'); 12 | $this->addConfigs($configs); 13 | } 14 | } -------------------------------------------------------------------------------- /LICENSE: -------------------------------------------------------------------------------- 1 | GNU GENERAL PUBLIC LICENSE 2 | Version 3, 29 June 2007 3 | 4 | Copyright (C) 2007 Free Software Foundation, Inc. 5 | Everyone is permitted to copy and distribute verbatim copies 6 | of this license document, but changing it is not allowed. 7 | 8 | Preamble 9 | 10 | The GNU General Public License is a free, copyleft license for 11 | software and other kinds of works. 12 | 13 | The licenses for most software and other practical works are designed 14 | to take away your freedom to share and change the works. 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Limitation of Liability. 601 | 602 | IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING 603 | WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MODIFIES AND/OR CONVEYS 604 | THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES, INCLUDING ANY 605 | GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING OUT OF THE 606 | USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED TO LOSS OF 607 | DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY YOU OR THIRD 608 | PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER PROGRAMS), 609 | EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF 610 | SUCH DAMAGES. 611 | 612 | 17. Interpretation of Sections 15 and 16. 613 | 614 | If the disclaimer of warranty and limitation of liability provided 615 | above cannot be given local legal effect according to their terms, 616 | reviewing courts shall apply local law that most closely approximates 617 | an absolute waiver of all civil liability in connection with the 618 | Program, unless a warranty or assumption of liability accompanies a 619 | copy of the Program in return for a fee. 620 | 621 | END OF TERMS AND CONDITIONS 622 | 623 | How to Apply These Terms to Your New Programs 624 | 625 | If you develop a new program, and you want it to be of the greatest 626 | possible use to the public, the best way to achieve this is to make it 627 | free software which everyone can redistribute and change under these terms. 628 | 629 | To do so, attach the following notices to the program. It is safest 630 | to attach them to the start of each source file to most effectively 631 | state the exclusion of warranty; and each file should have at least 632 | the "copyright" line and a pointer to where the full notice is found. 633 | 634 | 635 | Copyright (C) 636 | 637 | This program is free software: you can redistribute it and/or modify 638 | it under the terms of the GNU General Public License as published by 639 | the Free Software Foundation, either version 3 of the License, or 640 | (at your option) any later version. 641 | 642 | This program is distributed in the hope that it will be useful, 643 | but WITHOUT ANY WARRANTY; without even the implied warranty of 644 | MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the 645 | GNU General Public License for more details. 646 | 647 | You should have received a copy of the GNU General Public License 648 | along with this program. If not, see . 649 | 650 | Also add information on how to contact you by electronic and paper mail. 651 | 652 | If the program does terminal interaction, make it output a short 653 | notice like this when it starts in an interactive mode: 654 | 655 | Copyright (C) 656 | This program comes with ABSOLUTELY NO WARRANTY; for details type `show w'. 657 | This is free software, and you are welcome to redistribute it 658 | under certain conditions; type `show c' for details. 659 | 660 | The hypothetical commands `show w' and `show c' should show the appropriate 661 | parts of the General Public License. Of course, your program's commands 662 | might be different; for a GUI interface, you would use an "about box". 663 | 664 | You should also get your employer (if you work as a programmer) or school, 665 | if any, to sign a "copyright disclaimer" for the program, if necessary. 666 | For more information on this, and how to apply and follow the GNU GPL, see 667 | . 668 | 669 | The GNU General Public License does not permit incorporating your program 670 | into proprietary programs. If your program is a subroutine library, you 671 | may consider it more useful to permit linking proprietary applications with 672 | the library. If this is what you want to do, use the GNU Lesser General 673 | Public License instead of this License. But first, please read 674 | . 675 | -------------------------------------------------------------------------------- /README.md: -------------------------------------------------------------------------------- 1 | ``` 2 | ______ _____ ______ __ 3 | | ____| | __ \| _ \ \ / / 4 | | |__ _ __ ___ ___| |__) | |_) \ V / 5 | | __| '__/ _ \/ _ \ ___/| _ < > < 6 | | | | | | __/ __/ | | |_) / . \ 7 | |_| |_| \___|\___|_| |____/_/ \_\ 8 | Your Open Source Asterisk PBX GUI Solution 9 | ``` 10 | ### What? 11 | webrtc 12 | This is a module for [FreePBX©](http://www.freepbx.org/ "FreePBX Home Page"). [FreePBX](http://www.freepbx.org/ "FreePBX Home Page") is an open source GUI (graphical user interface) that controls and manages [Asterisk©](http://www.asterisk.org/ "Asterisk Home Page") (PBX). FreePBX is licensed under GPL. 13 | [FreePBX](http://www.freepbx.org/ "FreePBX Home Page") is a completely modular GUI for Asterisk written in PHP and Javascript. Meaning you can easily write any module you can think of and distribute it free of cost to your clients so that they can take advantage of beneficial features in [Asterisk](http://www.asterisk.org/ "Asterisk Home Page") 14 | 15 | ### Setting up a FreePBX system 16 | [See our WIKI](http://sangomakb.atlassian.net/wiki/spaces/FP/pages/9732130/Install+FreePBX) 17 | ### License 18 | [This modules code is licensed as GPLv3+](http://www.gnu.org/licenses/gpl-3.0.txt) 19 | ### Contributing 20 | To contribute code or modules back into the [FreePBX](http://www.freepbx.org/ "FreePBX Home Page") ecosystem you must fully read our Code License Agreement. We are not able to look at or accept patches or code of any kind until this document is filled out. Please take a look at [http://sangomakb.atlassian.net/wiki/spaces/FP/pages/10682663/Contributor+License+Agreement](http://sangomakb.atlassian.net/wiki/spaces/FP/pages/10682663/Contributor+License+Agreement) for more information 21 | ### Issues 22 | Please file bug reports at http://github.com/FreePBX/issue-tracker/issues -------------------------------------------------------------------------------- /Restore.php: -------------------------------------------------------------------------------- 1 | getConfigs(); 7 | $this->importTables($configs['tables']); 8 | $bmo = $this->FreePBX->Webrtc; 9 | $clients = $bmo->getClientsEnabled(); 10 | foreach($clients as $client) { 11 | $bmo->createDevice($client['user'],$client['certid']); 12 | } 13 | } 14 | 15 | public function processLegacy($pdo, $data, $tables, $unknownTables){ 16 | $this->restoreLegacyDatabase($pdo); 17 | //recreate all extensions on restore 18 | $bmo = $this->FreePBX->Webrtc; 19 | $clients = $bmo->getClientsEnabled(); 20 | foreach($clients as $client) { 21 | $bmo->createDevice($client['user'],$client['certid']); 22 | } 23 | } 24 | } 25 | -------------------------------------------------------------------------------- /Webrtc.class.php: -------------------------------------------------------------------------------- 1 | ["transport" => "wss,ws", "avpf" => "yes", "force_avp" => "yes", "icesupport" => "yes", "encryption" => "yes", "rtcp_mux" => "yes"], "pjsip" => ["media_use_received_transport" => "yes", "avpf" => "yes", "icesupport" => "yes", "rtcp_mux" => "yes"]]; 15 | 16 | /** 17 | * Prefix added to all WebRTC Extensions 18 | * @type {int} 19 | */ 20 | private string $prefix = '99'; 21 | 22 | public function doConfigPageInit($page) { 23 | 24 | } 25 | 26 | public function install() { 27 | $status = $this->validVersion(); 28 | if($status !== true) { 29 | out($status); 30 | throw new Exception($status); 31 | } 32 | //Remove Old Link if need be 33 | if(file_exists($this->FreePBX->Config->get('ASTETCDIR').'/http.conf') && is_link($this->FreePBX->Config->get('ASTETCDIR').'/http.conf') && (readlink($this->FreePBX->Config->get('ASTETCDIR').'/http.conf') == __DIR__.'/etc/httpd.conf')) { 34 | unlink($this->FreePBX->Config->get('ASTETCDIR').'/http.conf'); 35 | } 36 | 37 | 38 | if($this->FreePBX->Config->conf_setting_exists('HTTPENABLED')) { 39 | $this->FreePBX->Config->set_conf_values(['HTTPENABLED' => true],true); 40 | } 41 | 42 | try { 43 | $sql = "SELECT * FROM webrtc_settings"; 44 | $sth = $this->Database->prepare($sql); 45 | $sth->execute(); 46 | $settings = $sth->fetchAll(PDO::FETCH_ASSOC); 47 | if(!empty($settings)) { 48 | foreach($settings as $setting) { 49 | $this->setConfig($setting['key'], $setting['value']); 50 | } 51 | } 52 | $sql = "DROP TABLE IF EXISTS `webrtc_settings`"; 53 | $sth = $this->Database->prepare($sql); 54 | $sth->execute(); 55 | } catch(Exception) {} 56 | 57 | $prefix = $this->getConfig('prefix'); 58 | if(empty($prefix)) { 59 | $this->setConfig('prefix','99'); 60 | } 61 | 62 | $clients = $this->getClientsEnabled(); 63 | foreach($clients as $client) { 64 | $prefix = $client['prefix'] ?? $this->prefix; 65 | $module = isset($client['module'])&& ($client['module']!='')?$client['module']:'UCP'; 66 | $this->createDevice($client['user'],$client['certid'],$prefix,$module); 67 | } 68 | // update settings from core which are already saved 69 | $sql = "SELECT DISTINCT(`user`) FROM webrtc_clients"; 70 | $sth = $this->Database->prepare($sql); 71 | $sth->execute(); 72 | $results = $sth->fetchAll(PDO::FETCH_ASSOC); 73 | if(!empty($results)) { 74 | foreach($results as $row) { 75 | $setting = []; 76 | $id= $row['user']; 77 | $q1 = "SELECT `data` FROM sip where id=? AND `keyword` = ?"; 78 | $sth1 = $this->Database->prepare($q1); 79 | $sth1->execute([$id, 'accountcode']); 80 | $r1 = $sth1->fetch(PDO::FETCH_ASSOC); 81 | $setting['devinfo_accountcode'] = (is_array($r1) && array_key_exists('data', $r1)) ? $r1['data'] : ''; 82 | // 83 | $sth1->execute([$id, 'namedcallgroup']); 84 | $r2 = $sth1->fetch(PDO::FETCH_ASSOC); 85 | $setting['devinfo_namedcallgroup'] = (is_array($r2) && array_key_exists('data', $r2)) ? $r2['data'] : ''; 86 | // 87 | $sth1->execute([$id, 'namedpickupgroup']); 88 | $r3 = $sth1->fetch(PDO::FETCH_ASSOC); 89 | $setting['devinfo_namedpickupgroup'] = (is_array($r3) && array_key_exists('data', $r3)) ? $r3['data'] : ''; 90 | $this->updatefromcore($id,$setting); 91 | unset($setting); 92 | } 93 | } 94 | return true; 95 | } 96 | public function uninstall() { 97 | $sql = "SELECT * FROM webrtc_clients"; 98 | $sth = $this->Database->prepare($sql); 99 | $sth->execute(); 100 | $results = $sth->fetchAll(PDO::FETCH_ASSOC); 101 | if(!empty($results)) { 102 | foreach($results as $row) { 103 | $this->removeDevice($row['user']); 104 | } 105 | } 106 | return true; 107 | } 108 | 109 | public function genConfig() { 110 | } 111 | 112 | /** 113 | * Enable WebRTC and originate for said user 114 | * @param string $username Username 115 | */ 116 | public function migrationEnable($username) { 117 | $user = $this->FreePBX->Userman->getUserByUsername($username); 118 | if(!empty($user) && !empty($user['default_extension']) && $user['default_extension'] != "none") { 119 | if($this->FreePBX->Certman->checkCAexists()) { 120 | $certs = $this->FreePBX->Certman->getAllManagedCertificates(); 121 | if(!empty($certs)) { 122 | $this->createDevice($user['default_extension']); 123 | } 124 | } 125 | } 126 | } 127 | 128 | public function ucpDelGroup($id,$display,$data) { 129 | if(!empty($data['users'])) { 130 | foreach($data['users'] as $id) { 131 | $enabled = $this->FreePBX->Ucp->getCombinedSettingByID($id, 'Webrtc', 'enabled'); 132 | 133 | $user = $this->FreePBX->Userman->getUserByID($id); 134 | if(!empty($user['default_extension']) && $user['default_extension'] != 'none' && $enabled) { 135 | if(!$this->checkEnabled($user['default_extension'])) { 136 | $this->createDevice($user['default_extension'],$_REQUEST['webrtc_cert']); 137 | } 138 | } else { 139 | if($this->checkEnabled($user['default_extension'])) { 140 | $this->removeDevice($user['default_extension']); 141 | } 142 | } 143 | } 144 | } 145 | } 146 | 147 | public function ucpAddGroup($id, $display, $data) { 148 | $this->ucpUpdateGroup($id,$display,$data); 149 | } 150 | 151 | public function ucpUpdateGroup($id,$display,$data) { 152 | if($display == "userman") { 153 | if(isset($_POST['webrtc_enable']) && $_POST['webrtc_enable'] == 'yes') { 154 | $this->FreePBX->Ucp->setSettingByGID($id,'Webrtc','enabled',true); 155 | } else { 156 | $this->FreePBX->Ucp->setSettingByGID($id,'Webrtc','enabled',false); 157 | } 158 | } 159 | 160 | $group = $this->FreePBX->Userman->getGroupByGID($id); 161 | foreach($group['users'] as $user) { 162 | $enabled = $this->FreePBX->Ucp->getCombinedSettingByID($user, 'Webrtc', 'enabled'); 163 | $user = $this->FreePBX->Userman->getUserByID($user); 164 | if(!empty($user['default_extension']) && $user['default_extension'] != 'none' && $enabled) { 165 | $dev = $this->FreePBX->Core->getDevice($user['default_extension']); 166 | $id = $this->prefix.$user['default_extension']; 167 | $settings = $this->FreePBX->Certman->getDTLSOptions($id); 168 | $defaultCert = $this->FreePBX->Certman->getDefaultCertDetails(); 169 | if(empty($defaultCert)) { 170 | return false; 171 | } 172 | 173 | if(!empty($dev) && (!$this->checkEnabled($user['default_extension']) || ($this->checkEnabled($user['default_extension']) && $settings['cid'] != $defaultCert['cid']))) { 174 | $this->createDevice($user['default_extension']); 175 | } 176 | } elseif($user['default_extension'] != 'none') { 177 | if($this->checkEnabled($user['default_extension'])) { 178 | $this->removeDevice($user['default_extension']); 179 | } 180 | } 181 | } 182 | } 183 | 184 | /** 185 | * Hook functionality from userman when a user is deleted 186 | * @param {int} $id The userman user id 187 | * @param {string} $display The display page name where this was executed 188 | * @param {array} $data Array of data to be able to use 189 | */ 190 | public function ucpDelUser($id, $display, $ucpStatus, $data) { 191 | $enabled = $this->FreePBX->Ucp->getCombinedSettingByID($id, 'Webrtc', 'enabled'); 192 | 193 | $user = $this->FreePBX->Userman->getUserByID($id); 194 | if(!empty($user['default_extension']) && $user['default_extension'] != 'none' && $enabled) { 195 | } else { 196 | if(isset($user['default_extension']) && $this->checkEnabled($user['default_extension'])) { 197 | $this->removeDevice($user['default_extension']); 198 | } 199 | } 200 | } 201 | 202 | /** 203 | * Hook functionality from userman when a user is added 204 | * @param {int} $id The userman user id 205 | * @param {string} $display The display page name where this was executed 206 | * @param {array} $data Array of data to be able to use 207 | */ 208 | public function ucpAddUser($id, $display, $ucpStatus, $data) { 209 | $this->ucpUpdateUser($id, $display, $ucpStatus, $data); 210 | } 211 | 212 | /** 213 | * Hook functionality from userman when a user is updated 214 | * @param {int} $id The userman user id 215 | * @param {string} $display The display page name where this was executed 216 | * @param {array} $data Array of data to be able to use 217 | */ 218 | public function ucpUpdateUser($id, $display, $ucpStatus, $data) { 219 | if($display == "userman") { 220 | if(isset($_POST['webrtc_enable']) && $_POST['webrtc_enable'] == 'yes') { 221 | $this->FreePBX->Ucp->setSettingByID($id,'Webrtc','enabled',true); 222 | } elseif(isset($_POST['webrtc_enable']) && $_POST['webrtc_enable'] == 'no') { 223 | $this->FreePBX->Ucp->setSettingByID($id,'Webrtc','enabled',false); 224 | } elseif(isset($_POST['webrtc_enable']) && $_POST['webrtc_enable'] == 'inherit') { 225 | $this->FreePBX->Ucp->setSettingByID($id,'Webrtc','enabled',null); 226 | } 227 | } 228 | 229 | $enabled = $this->FreePBX->Ucp->getCombinedSettingByID($id, 'Webrtc', 'enabled'); 230 | 231 | $user = $this->FreePBX->Userman->getUserByID($id); 232 | if(!empty($user['default_extension']) && $user['default_extension'] != 'none' && $enabled) { 233 | $id = $this->prefix.$user['default_extension']; 234 | $defaultCert = $this->FreePBX->Certman->getDefaultCertDetails(); 235 | if(empty($defaultCert)) { 236 | return false; 237 | } 238 | $settings = $this->FreePBX->Certman->getDTLSOptions($id); 239 | if(!$this->checkEnabled($user['default_extension']) || ($this->checkEnabled($user['default_extension']) && $settings['cid'] != $defaultCert['cid'])) { 240 | $this->createDevice($user['default_extension']); 241 | } 242 | } else { 243 | if($this->checkEnabled($user['default_extension'])) { 244 | $this->removeDevice($user['default_extension']); 245 | } 246 | } 247 | // 248 | } 249 | 250 | public function ucpConfigPage($mode, $user, $action) { 251 | $html = []; 252 | $defaultCert = $this->FreePBX->Certman->getDefaultCertDetails(); 253 | if(empty($user)) { 254 | $enabled = ($mode == 'group') ? true : null; 255 | } else { 256 | if($mode == 'group') { 257 | $enabled = $this->FreePBX->Ucp->getSettingByGID($user['id'],'Webrtc','enabled'); 258 | $enabled = !($enabled) ? false : true; 259 | } else { 260 | $enabled = $this->FreePBX->Ucp->getSettingByID($user['id'],'Webrtc','enabled'); 261 | } 262 | } 263 | 264 | $html[0] = ["title" => _("Phone"), "rawname" => "webrtc", "content" => ""]; 265 | if($this->validVersion() === true && !empty($defaultCert)) { 266 | $html[0]['content'] = load_view(__DIR__."/views/ucp_config.php",["mode" => $mode, "enabled" => $enabled, "webrtcmessage" => '', "config" => true]); 267 | } elseif($this->validVersion() === true) { 268 | $html[0]['content'] = load_view(__DIR__."/views/ucp_config.php",["mode" => $mode, "enabled" => $enabled, "webrtcmessage" => sprintf(_('You have no default certificates setup in %s'),''._('Certificate Manager').''), "config" => false]); 269 | } else { 270 | $html[0]['content'] = load_view(__DIR__."/views/ucp_config.php",["mode" => $mode, "enabled" => $enabled, "webrtcmessage" => $this->validVersion(), "config" => false]); 271 | } 272 | return $html; 273 | } 274 | 275 | public function validVersion() { 276 | $version = $this->FreePBX->Config->get('ASTVERSION'); 277 | $vParts = explode(".",(string) $version); 278 | $base = $vParts[0]; 279 | $res_ver = IsAsteriskSupported($base); // method located in framework utility.function.php 280 | if ($res_ver["status"] == false) { 281 | return (sprintf(_("Running an unsupported version of Asterisk. %s Detected Asterisk version: %s "), $res_ver["message"], $version)); 282 | } 283 | return true; 284 | } 285 | 286 | public function getAllSettings() { 287 | return $this->getAll(); 288 | } 289 | 290 | public function getSetting($setting) { 291 | return $this->getConfig($setting); 292 | } 293 | 294 | public function setSetting($setting,$value) { 295 | return $this->setConfig($setting,$value); 296 | } 297 | 298 | public function delSetting($setting) { 299 | return $this->delConfig($setting); 300 | } 301 | 302 | public function checkEnabled($user,$prefix ='') { 303 | $prefix = ($prefix == '')?$this->prefix:$prefix; 304 | $settings = $this->getClientSettingsByUser($user,$prefix); 305 | return !empty($settings); 306 | } 307 | 308 | public function setClientSettings($user,$device,$prefix,$module) { 309 | try { 310 | $sql = "REPLACE INTO webrtc_clients (`user`, `device`,`prefix`,`module`) VALUES(?,?,?,?)"; 311 | $sth = $this->Database->prepare($sql); 312 | return $sth->execute([$user, $device, $prefix, $module]); 313 | } catch(Exception) { 314 | return false; 315 | } 316 | } 317 | 318 | /** This does the same as setClientSettings but adds certid. did not want to break existing calls with a bad certid default */ 319 | public function upsertClientSettings($user,$device,$certid) { 320 | try { 321 | $sql = "REPLACE INTO webrtc_clients (`user`, `device`, `certid`) VALUES(?,?,?)"; 322 | $sth = $this->Database->prepare($sql); 323 | return $sth->execute([$user, $device, $certid]); 324 | } catch(Exception) { 325 | return false; 326 | } 327 | } 328 | 329 | public function getClientsEnabled() { 330 | $sql = "SELECT * FROM webrtc_clients"; 331 | $sth = $this->Database->prepare($sql); 332 | $sth->execute(); 333 | $results = $sth->fetchAll(PDO::FETCH_ASSOC); 334 | if(empty($results)) { 335 | return []; 336 | } 337 | return $results; 338 | } 339 | 340 | public function getClientSettingsByUser($user,$prefix='') { 341 | $prefix = ($prefix == '')?$this->prefix:$prefix; 342 | $sql = "SELECT * FROM webrtc_clients WHERE `user` = ? AND `prefix`=?"; 343 | $sth = $this->Database->prepare($sql); 344 | $sth->execute([$user, $prefix]); 345 | $results = $sth->fetch(PDO::FETCH_ASSOC); 346 | if(empty($results)) { 347 | return false; 348 | } 349 | 350 | $serverparts = explode(":", (string) $_SERVER['HTTP_HOST']); //strip off port because we define it 351 | $sip_server = $serverparts[0]; 352 | $secure = isset($_SERVER['HTTPS']) && $_SERVER['HTTPS'] == "on"; 353 | $dev = $this->FreePBX->Core->getDevice($results['device']); 354 | if(empty($dev)) { 355 | //no device so remove the settings, someone deleted the device basically 356 | $this->removeClientSettingsByUser($user,$prefix); 357 | return false; 358 | } 359 | if($this->FreePBX->Config->get('HTTPTLSENABLE') && $dev['transport'] == "chan_sip" && ($dev['transport'] != "wss" && $dev['transport'] != "wss,ws")) { 360 | return false; 361 | } 362 | //$usr = core_users_get($results['user']); 363 | $results['status'] = true; 364 | $results['realm'] = !empty($results['realm']) ? $results['realm'] : $sip_server; 365 | $results['username'] = !empty($results['username']) ? $results['username'] : $dev['id']; 366 | $results['sipuri'] = !empty($results['sipuri']) ? $results['sipuri'] : 'sip:'.$results['username'].'@'.$sip_server; 367 | $results['password'] = !empty($results['password']) ? $results['password'] : $dev['secret']; 368 | $prefix = $this->FreePBX->Config->get('HTTPPREFIX'); 369 | $suffix = !empty($prefix) ? "/".$prefix."/ws" : "/ws"; 370 | 371 | if($secure && !$this->FreePBX->Config->get('HTTPTLSENABLE')) { 372 | return ["status" => false, "message" => _("HTTPS is not enabled for Asterisk")]; 373 | } 374 | 375 | $type = ($this->FreePBX->Config->get('HTTPTLSENABLE') && $secure) ? 'wss' : 'ws'; 376 | $port = ($this->FreePBX->Config->get('HTTPTLSENABLE') && $secure) ? $this->FreePBX->Config->get('HTTPTLSBINDPORT') : $this->FreePBX->Config->get('HTTPBINDPORT'); 377 | $results['websocket'] = !empty($results['websocket']) ? $results['websocket'] : $type.'://'.$sip_server.':'.$port.$suffix; 378 | try { 379 | $stunaddr = $this->FreePBX->Sipsettings->getConfig("webrtcstunaddr"); 380 | $stunaddr = !empty($stunaddr) ? $stunaddr : $this->FreePBX->Sipsettings->getConfig("stunaddr"); 381 | $results['stunaddr'] = $stunaddr; 382 | } catch(Exception) {} 383 | $results['stunaddr'] = !empty($results['stunaddr']) ? "stun:".$results['stunaddr'] : "stun:stun.l.google.com:19302"; 384 | return $results; 385 | } 386 | 387 | public function removeClientSettingsByUser($user,$prefix) { 388 | try { 389 | $sql = "DELETE FROM webrtc_clients WHERE `user` = ? AND prefix=?"; 390 | $sth = $this->Database->prepare($sql); 391 | return $sth->execute([$user, $prefix]); 392 | } catch(Exception) { 393 | return true; 394 | } 395 | } 396 | 397 | /* Updates the extension settings from Core */ 398 | 399 | public function updatefromcore($ext,$settings=[]){ 400 | //update accountcode from primary extension to all its devices of webrtc 401 | $sql = "SELECT `device` FROM webrtc_clients WHERE `user` = ? "; 402 | $sth = $this->Database->prepare($sql); 403 | $sth->execute([$ext]); 404 | $results = $sth->fetchAll(PDO::FETCH_ASSOC); 405 | if(is_array($results)){ 406 | foreach($results as $res) { 407 | $device = $res['device']; 408 | if(isset($settings['devinfo_accountcode']) && strlen(trim((string) $settings['devinfo_accountcode'])) > 0){ 409 | $data = $settings['devinfo_accountcode']; 410 | $query = "Update sip SET `data`=? Where `id`=? AND `keyword`=?"; 411 | $sth1 = $this->Database->prepare($query); 412 | $sth1->execute([$data, $device, 'accountcode']); 413 | } 414 | // update pickup groups 415 | if(isset($settings['devinfo_namedcallgroup']) && strlen(trim((string) $settings['devinfo_namedcallgroup'])) > 0){ 416 | $data = $settings['devinfo_namedcallgroup']; 417 | $query = "REPLACE INTO sip (`id`, `data`,`keyword`) VALUES(?,?,?)"; 418 | $sth1 = $this->Database->prepare($query); 419 | $sth1->execute([$device, $data, 'namedcallgroup']); 420 | } 421 | //devinfo_namedpickupgroup 422 | if(isset($settings['devinfo_namedpickupgroup']) && strlen(trim((string) $settings['devinfo_namedpickupgroup'])) > 0){ 423 | $data = $settings['devinfo_namedpickupgroup']; 424 | $query = "REPLACE INTO sip (`id`, `data`,`keyword`) VALUES(?,?,?)"; 425 | $sth1 = $this->Database->prepare($query); 426 | $sth1->execute([$device, $data, 'namedpickupgroup']); 427 | } 428 | } 429 | } 430 | } 431 | 432 | public function createDevice($extension,$cid = '',$prefix='',$module='UCP') { 433 | if($prefix == ''){ 434 | $id = $this->prefix.$extension; 435 | $prefix = $this->prefix; 436 | }else { 437 | $id = $prefix.$extension; 438 | } 439 | $previous = $this->FreePBX->Core->getDevice($id); 440 | if(!empty($previous)) { 441 | $this->FreePBX->Core->delDevice($id); 442 | } 443 | $version = $this->FreePBX->Config->get('ASTVERSION'); 444 | $user = $this->FreePBX->Core->getUser($extension); 445 | $user['name'] = (array_key_exists('name', $user)) ? $user['name'] : ''; 446 | $dev = $this->FreePBX->Core->getDevice($extension); 447 | $socket = $this->getSocketMode(); 448 | $settings = $this->FreePBX->Core->generateDefaultDeviceSettings($socket,$id,'WebRTC '.$user['name']); 449 | if(!empty($previous['secret'])) { 450 | $settings['secret']['value'] = $previous['secret']; 451 | } 452 | if($module == 'SangomaConnect'){ 453 | $settings['force_callerid']['value'] = 'yes'; 454 | $settings['callerid']['value'] = $user['name'].' <'.$extension.'>'; 455 | } 456 | if (isset($dev['accountcode'])) { 457 | $settings['accountcode']['value'] = $dev['accountcode']; 458 | } else { 459 | $accountcode = $this->FreePBX->astman->database_get("AMPUSER",$extension."/accountcode"); 460 | $settings['accountcode']['value'] = $accountcode; 461 | } 462 | 463 | $settings['namedcallgroup']['value'] = $dev['namedcallgroup'] ?? ''; 464 | $settings['namedpickupgroup']['value'] = $dev['namedpickupgroup'] ?? ''; 465 | $settings['devicetype']['value'] = 'fixed'; 466 | $settings['context']['value'] = !empty($dev['context']) ? $dev['context'] : "from-internal"; 467 | $settings['user']['value'] = $extension; 468 | $settings['webrtc']['value'] = !empty($dev['webrtc']) ? $dev['webrtc'] : "yes"; 469 | $defaultCert = $this->FreePBX->Certman->getDefaultCertDetails(); 470 | if(empty($defaultCert)) { 471 | return false; 472 | } 473 | if($cid == ''){ 474 | $cid = $defaultCert['cid']; 475 | } 476 | $cert = ["certificate" => $cid, "verify" => "fingerprint", "setup" => "actpass", "rekey" => "0"]; 477 | switch($socket) { 478 | case 'sip': 479 | $settings['avpf']['value'] = 'yes'; 480 | $settings['force_avp']['value'] = 'yes'; 481 | $settings['transport']['value'] = 'wss,ws'; 482 | $settings['icesupport']['value'] = 'yes'; 483 | $settings['encryption']['value'] = 'yes'; 484 | $settings['sessiontimers']['value'] = 'refuse'; 485 | $settings['videosupport']['value'] = 'no'; 486 | if((version_compare($version,'13.15.0','ge') && version_compare($version,'14.0','lt')) || version_compare($version,'14.4.0','ge')) { 487 | $settings['rtcp_mux']['value'] = 'yes'; 488 | } 489 | $this->FreePBX->Core->addDevice($id,'sip',$settings); 490 | break; 491 | case 'pjsip': 492 | $settings['avpf']['value'] = 'yes'; 493 | $settings['icesupport']['value'] = 'yes'; 494 | $settings['media_use_received_transport']['value'] = 'yes'; 495 | $settings['timers']['value'] = 'no'; 496 | $settings['media_encryption']['value'] = 'dtls'; 497 | $settings['webrtc']['value'] = 'yes'; 498 | if((version_compare($version,'13.15.0','ge') && version_compare($version,'14.0','lt')) || version_compare($version,'14.4.0','ge')) { 499 | $settings['rtcp_mux']['value'] = 'yes'; 500 | } 501 | $this->FreePBX->Core->addDevice($id,'pjsip',$settings); 502 | break; 503 | default: 504 | return false; 505 | break; 506 | } 507 | $this->FreePBX->Certman->addDTLSOptions($id, $cert); 508 | $this->setClientSettings($extension,$id,$prefix,$module); 509 | return true; 510 | } 511 | 512 | public function getSocketMode() { 513 | $websocketMode = null; 514 | if($this->FreePBX->astman->mod_loaded("res_pjsip_transport_websocket")) { 515 | $type = $this->FreePBX->astman->Command("module show like res_pjsip_transport_websocket"); 516 | if(preg_match("/Not Running/",(string) $type['data'])) { 517 | $websocketMode = 'sip'; 518 | } else { 519 | $websocketMode = 'pjsip'; 520 | } 521 | } else { 522 | $websocketMode = 'sip'; 523 | } 524 | return $websocketMode; 525 | } 526 | 527 | public function removeDevice($extension,$prefix = '') { 528 | if($prefix == ''){ 529 | $prefix = $this->prefix; 530 | } 531 | $id = $prefix.$extension; 532 | $this->removeClientSettingsByUser($extension,$prefix); 533 | $this->deleteDevice($id); 534 | } 535 | 536 | private function deleteDevice($device) { 537 | try { 538 | return $this->FreePBX->Core->delDevice($device); 539 | } catch(Exception) { 540 | return false; 541 | } 542 | } 543 | public function dashboardIgnoreExt(){ 544 | return [['length' => 2, 'value' => '99']]; 545 | } 546 | public function delUser($extension, $editmode=false) { 547 | if(!$editmode) { 548 | $this->removeDevice($extension); 549 | } 550 | } 551 | 552 | public function setDatabase($pdo){ 553 | $this->Database = $pdo; 554 | return $this; 555 | } 556 | 557 | public function resetDatabase(){ 558 | $this->Database = $this->FreePBX->Database; 559 | return $this; 560 | } 561 | } 562 | -------------------------------------------------------------------------------- /functions.inc.php: -------------------------------------------------------------------------------- 1 | getInfo('fw_ari'); 21 | if(!empty($fw_ari['fw_ari']) && $fw_ari['fw_ari']['status'] == MODULE_STATUS_ENABLED) { 22 | if ($tech_hardware != null || $extdisplay != '' || $action == 'add') { 23 | $currentcomponent->addoptlistitem('webrtc_enable', 'no', 'No'); 24 | $currentcomponent->addoptlistitem('webrtc_enable', 'yes', 'Yes'); 25 | $currentcomponent->setoptlistopts('webrtc_enable', 'sort', false); 26 | 27 | $currentcomponent->addguifunc("webrtc_{$pagename}_configpageload"); 28 | 29 | if (!empty($action)) { 30 | $currentcomponent->addprocessfunc("webrtc_{$pagename}_configprocess"); 31 | } 32 | } 33 | } 34 | } 35 | 36 | function webrtc_extensions_configpageload() { 37 | webrtc_configpageload('extension'); 38 | } 39 | 40 | function webrtc_configpageload($mode) { 41 | global $currentcomponent; 42 | $webrtc = FreePBX::Webrtc(); 43 | $certman = FreePBX::Certman(); 44 | 45 | 46 | $extdisplay = $_REQUEST['extdisplay'] ?? null; 47 | 48 | $webrtc_select = $currentcomponent->getoptlist('webrtc_enable'); 49 | 50 | $webrtc_value = $webrtc->checkEnabled($extdisplay) ? 'yes' : 'no'; 51 | $mcerts = $certman->getAllManagedCertificates(); 52 | 53 | $status = $webrtc->validVersion(); 54 | $mode = $webrtc->getSocketMode(); 55 | if($status === true) { 56 | if(!empty($mcerts)) { 57 | if($mode == 'pjsip') { 58 | $currentcomponent->addguielem('WebRTC Phone', new gui_label('webrtc_message',_('The WebSockets Interface is running through PJSIP, PJSIP is not supported at this time. Please enable the chan_sip driver (along with pjsip or by itself) or alert the FreePBX Developers'))); 59 | $currentcomponent->addguielem('Device Options', new gui_hidden('webrtc_enable', $webrtc_value)); 60 | } else { 61 | $currentcomponent->addguielem('WebRTC Phone', new gui_selectbox( 'webrtc_enable', $webrtc_select, $webrtc_value, 62 | _('Enable WebRTC Old ARI Phone'), sprintf(_('Enables WebRTC for this %s in the Asterisk Recording Interface (ARI). Note: ARI is depreciated in favor of UCP'),$mode), false)); 63 | $certs = []; 64 | foreach($mcerts as $cert) { 65 | $certs[] = ["text" => $cert['basename'], "value" => $cert['cid']]; 66 | } 67 | $settings = $webrtc->getClientSettingsByUser($extdisplay); 68 | $cert = !empty($settings['certid']) ? $settings['certid'] : ''; 69 | $currentcomponent->addguielem('WebRTC Phone', new gui_selectbox( 70 | 'webrtc_dtls_certificate', 71 | $certs, 72 | $cert, 73 | _('Use Certificate'), 74 | _("The Certificate to use from Certificate Manager"), 75 | false) 76 | ); 77 | } 78 | } else { 79 | $currentcomponent->addguielem('WebRTC Phone', new gui_label('webrtc_message',sprintf(_('To utilize WebRTC in ARI you must add at least one certificate %s through Certificate Manager'),$mode))); 80 | $currentcomponent->addguielem('Device Options', new gui_hidden('webrtc_enable', $webrtc_value)); 81 | } 82 | } else { 83 | $currentcomponent->addguielem('Device Options', new gui_hidden('webrtc_enable', $webrtc_value)); 84 | } 85 | } 86 | 87 | function webrtc_extensions_configprocess() { 88 | $action = $_REQUEST['action'] ?? null; 89 | $extension = $_REQUEST['extdisplay'] ?? null; 90 | $webrtc = FreePBX::Webrtc(); 91 | switch ($action) { 92 | case 'add': 93 | $extension = $_REQUEST['extension'] ?? null; 94 | case 'edit': 95 | $prev = ($action != 'add') ? $webrtc->checkEnabled($extension) : false; 96 | if($_REQUEST['webrtc_enable'] == 'yes' && !$prev) { 97 | $webrtc->createDevice($extension,$_REQUEST['webrtc_dtls_certificate']); 98 | } elseif($_REQUEST['webrtc_enable'] == 'no' && $prev) { 99 | $webrtc->removeDevice($extension); 100 | } 101 | break; 102 | case 'del': 103 | $webrtc->removeDevice($extension); 104 | break; 105 | } 106 | } 107 | -------------------------------------------------------------------------------- /i18n/cs/LC_MESSAGES/webrtc.mo: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/FreePBX/webrtc/92ce6acc8eced21055a5b63267b764a8609e4cae/i18n/cs/LC_MESSAGES/webrtc.mo -------------------------------------------------------------------------------- /i18n/cs/LC_MESSAGES/webrtc.po: -------------------------------------------------------------------------------- 1 | # This file is part of FreePBX. 2 | # 3 | # For licensing information, please see the file named LICENSE located in the module directory 4 | # 5 | # FreePBX language template for webrtc 6 | # Copyright (C) 2008-2018 Sangoma Technologies, Inc. 7 | # 8 | msgid "" 9 | msgstr "" 10 | "Project-Id-Version: PACKAGE VERSION\n" 11 | "Report-Msgid-Bugs-To: \n" 12 | "POT-Creation-Date: 2024-12-03 06:07-0500\n" 13 | "PO-Revision-Date: 2018-09-04 08:42+0000\n" 14 | "Last-Translator: Petr Varvarovsky \n" 15 | "Language-Team: Czech \n" 16 | "Language: cs\n" 17 | "MIME-Version: 1.0\n" 18 | "Content-Type: text/plain; charset=UTF-8\n" 19 | "Content-Transfer-Encoding: 8bit\n" 20 | "Plural-Forms: nplurals=3; plural=(n==1) ? 0 : (n>=2 && n<=4) ? 1 : 2;\n" 21 | "X-Generator: Weblate 3.0.1\n" 22 | 23 | #: ucp/views/phone.php:24 24 | msgid "ABC" 25 | msgstr "ABC" 26 | 27 | #: ucp/views/phone.php:78 28 | msgid "Call" 29 | msgstr "Volat" 30 | 31 | #: Webrtc.class.php:268 32 | msgid "Certificate Manager" 33 | msgstr "Správce certifikátů" 34 | 35 | #: ucp/assets/js/global.js:267 36 | msgid "Click this window to answer or close this window to ignore" 37 | msgstr "" 38 | "Kliknutím na toto okno můžete na tuto zprávu odpovědět, ignorovat ji nebo ji " 39 | "zavřít" 40 | 41 | #: ucp/assets/js/global.js:542 42 | msgid "Connect Phone" 43 | msgstr "Připojit telefon" 44 | 45 | #: ucp/assets/js/global.js:563 46 | msgid "Connecting to socket..." 47 | msgstr "Připojení k soketu ..." 48 | 49 | #: webrtc.i18n.php:6 50 | msgid "Connectivity" 51 | msgstr "Připojení" 52 | 53 | #: ucp/views/phone.php:28 54 | msgid "DEF" 55 | msgstr "DEF" 56 | 57 | #: ucp/assets/js/global.js:69 ucp/assets/js/global.js:77 58 | msgid "Disable" 59 | msgstr "Zakázat" 60 | 61 | #: ucp/views/settings.php:7 62 | msgid "Disconnect Phone" 63 | msgstr "Odpojit telefon" 64 | 65 | #: ucp/assets/js/global.js:554 66 | msgid "Disconnected" 67 | msgstr "Odpojeno" 68 | 69 | #: ucp/assets/js/global.js:68 ucp/assets/js/global.js:76 70 | msgid "Enable" 71 | msgstr "Povolit" 72 | 73 | #: views/ucp_config.php:19 74 | msgid "Enable Phone" 75 | msgstr "Povolit telefon" 76 | 77 | #: functions.inc/hook_core.php:62 78 | msgid "Enable WebRTC Old ARI Phone" 79 | msgstr "Povolit službu WebRTC pro starý telefon ARI" 80 | 81 | #: views/ucp_config.php:38 82 | msgid "Enable the phone in UCP for this user" 83 | msgstr "Povolit telefon v UCP pro tohoto uživatele" 84 | 85 | #: functions.inc/hook_core.php:62 86 | #, php-format 87 | msgid "" 88 | "Enables WebRTC for this %s in the Asterisk Recording Interface (ARI). Note: " 89 | "ARI is depreciated in favor of UCP" 90 | msgstr "" 91 | "Povolí WebRTC pro %s v nahrávacím rozhraní Asterisku (ARI). Poznámka: ARI je " 92 | "zastaralé oproti UCP" 93 | 94 | #: ucp/views/phone.php:34 95 | msgid "GHI" 96 | msgstr "GHI" 97 | 98 | #: Webrtc.class.php:372 99 | msgid "HTTPS is not enabled for Asterisk" 100 | msgstr "HTTPS není pro Asterisk povoleno" 101 | 102 | #: ucp/views/phone.php:81 103 | msgid "Hold" 104 | msgstr "Přidržet" 105 | 106 | #: ucp/assets/js/global.js:266 107 | #, perl-format 108 | msgid "Incoming call from %s" 109 | msgstr "Příchozí hovor od %s" 110 | 111 | #: views/ucp_config.php:29 112 | msgid "Inherit" 113 | msgstr "Zdědit" 114 | 115 | #: ucp/views/phone.php:38 116 | msgid "JKL" 117 | msgstr "JKL" 118 | 119 | #: ucp/views/phone.php:42 120 | msgid "MNO" 121 | msgstr "MNO" 122 | 123 | #: views/ucp_config.php:26 124 | msgid "No" 125 | msgstr "Ne" 126 | 127 | #: ucp/assets/js/global.js:633 128 | msgid "Not supported in this browser" 129 | msgstr "V tomto prohlížeči není podporováno" 130 | 131 | #: ucp/views/phone.php:88 132 | msgid "Note: An Active call will not be dropped when this window is closed" 133 | msgstr "Poznámka: Po uzavření tohoto okna nebude aktivní hovor zrušen" 134 | 135 | #: ucp/assets/js/global.js:640 136 | msgid "Only supported over HTTPS" 137 | msgstr "Podporováno pouze přes protokol HTTPS" 138 | 139 | #: ucp/views/phone.php:48 140 | msgid "PQRS" 141 | msgstr "PQRS" 142 | 143 | #: Webrtc.class.php:264 ucp/Webrtc.class.php:25 ucp/Webrtc.class.php:66 144 | msgid "Phone" 145 | msgstr "Telefon" 146 | 147 | #: ucp/views/phone.php:86 148 | msgid "Phone Status" 149 | msgstr "Stav telefonu" 150 | 151 | #: ucp/assets/js/global.js:522 152 | msgid "Registered" 153 | msgstr "Registrován" 154 | 155 | #: ucp/assets/js/global.js:569 156 | msgid "Registering..." 157 | msgstr "Registrace ..." 158 | 159 | #: ucp/assets/js/global.js:540 160 | msgid "Registration Failed" 161 | msgstr "Registrace se nezdařila" 162 | 163 | #: Webrtc.class.php:281 164 | #, php-format 165 | msgid "" 166 | "Running an unsupported version of Asterisk. %s Detected Asterisk version: %s " 167 | msgstr "" 168 | 169 | #: ucp/views/settings.php:3 170 | msgid "Silence Ringer" 171 | msgstr "Tiché vyzvánění" 172 | 173 | #: ucp/views/phone.php:52 174 | msgid "TUV" 175 | msgstr "TUV" 176 | 177 | #: functions.inc/hook_core.php:74 178 | msgid "The Certificate to use from Certificate Manager" 179 | msgstr "Certifikát k použítí od správce certifikátů" 180 | 181 | #: webrtc.i18n.php:8 182 | msgid "The WebRTC Module allows an Administrator to enable a " 183 | msgstr "Modul WebRTC umožňuje správci povolit " 184 | 185 | #: functions.inc/hook_core.php:58 186 | msgid "" 187 | "The WebSockets Interface is running through PJSIP, PJSIP is not supported at " 188 | "this time. Please enable the chan_sip driver (along with pjsip or by itself) " 189 | "or alert the FreePBX Developers" 190 | msgstr "" 191 | "Rozhraní WebSockets běží přes PJSIP, PJSIP momentálně není podporován. " 192 | "Prosím, povolte ovladač chan_sip (spolu s pjsip nebo samostatně), nebo " 193 | "upozorněte vývojáře FreePBX" 194 | 195 | #: functions.inc/hook_core.php:79 196 | #, php-format 197 | msgid "" 198 | "To utilize WebRTC in ARI you must add at least one certificate %s through " 199 | "Certificate Manager" 200 | msgstr "" 201 | "Chcete-li využít technologii WebRTC v systému ARI, musíte přidat certifikát " 202 | "%s prostřednictvím Správce certifikátů" 203 | 204 | #: ucp/assets/js/global.js:532 ucp/assets/js/global.js:549 205 | msgid "Unregistered" 206 | msgstr "Neregistrován" 207 | 208 | #: functions.inc/hook_core.php:73 209 | msgid "Use Certificate" 210 | msgstr "Použít certifikát" 211 | 212 | #: ucp/views/phone.php:56 213 | msgid "WXYZ" 214 | msgstr "WXYZ" 215 | 216 | #: ucp/Webrtc.class.php:74 217 | msgid "WebRTC" 218 | msgstr "WebRTC" 219 | 220 | #: webrtc.i18n.php:4 221 | msgid "WebRTC Phone" 222 | msgstr "WebRTC telefon" 223 | 224 | #: ucp/Webrtc.class.php:54 225 | msgid "Webrtc is not enabled for this extension." 226 | msgstr "" 227 | 228 | #: views/ucp_config.php:24 229 | msgid "Yes" 230 | msgstr "Ano" 231 | 232 | #: Webrtc.class.php:268 233 | #, php-format 234 | msgid "You have no default certificates setup in %s" 235 | msgstr "Nemáte žádné výchozí nastavení certifikátů v %s" 236 | 237 | #~ msgid "Unsupported Version of Asterisk, You need at least %s you have %s" 238 | #~ msgstr "Nepodporovaná verze Asterisku, potřebujete alespoň %s, vy máte %s" 239 | -------------------------------------------------------------------------------- /i18n/de_DE/LC_MESSAGES/webrtc.mo: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/FreePBX/webrtc/92ce6acc8eced21055a5b63267b764a8609e4cae/i18n/de_DE/LC_MESSAGES/webrtc.mo -------------------------------------------------------------------------------- /i18n/de_DE/LC_MESSAGES/webrtc.po: -------------------------------------------------------------------------------- 1 | # This file is part of FreePBX. 2 | # 3 | # For licensing information, please see the file named LICENSE located in the module directory 4 | # 5 | # FreePBX language template for webrtc 6 | # Copyright (C) 2008-2017 Sangoma Technologies, Inc. 7 | # 8 | msgid "" 9 | msgstr "" 10 | "Project-Id-Version: PACKAGE VERSION\n" 11 | "Report-Msgid-Bugs-To: \n" 12 | "POT-Creation-Date: 2024-12-03 06:07-0500\n" 13 | "PO-Revision-Date: 2018-06-01 14:24+0000\n" 14 | "Last-Translator: Nils Heckenkamp \n" 15 | "Language-Team: German \n" 16 | "Language: de_DE\n" 17 | "MIME-Version: 1.0\n" 18 | "Content-Type: text/plain; charset=UTF-8\n" 19 | "Content-Transfer-Encoding: 8bit\n" 20 | "Plural-Forms: nplurals=2; plural=n != 1;\n" 21 | "X-Generator: Weblate 2.19.1\n" 22 | 23 | #: ucp/views/phone.php:24 24 | msgid "ABC" 25 | msgstr "" 26 | 27 | #: ucp/views/phone.php:78 28 | msgid "Call" 29 | msgstr "" 30 | 31 | #: Webrtc.class.php:268 32 | msgid "Certificate Manager" 33 | msgstr "" 34 | 35 | #: ucp/assets/js/global.js:267 36 | msgid "Click this window to answer or close this window to ignore" 37 | msgstr "" 38 | "Klicken Sie zum Antworten auf dieses Fenster oder schließen Sie dieses " 39 | "Fenster, um es zu ignorieren" 40 | 41 | #: ucp/assets/js/global.js:542 42 | msgid "Connect Phone" 43 | msgstr "Telefon verbinden" 44 | 45 | #: ucp/assets/js/global.js:563 46 | msgid "Connecting to socket..." 47 | msgstr "" 48 | 49 | #: webrtc.i18n.php:6 50 | msgid "Connectivity" 51 | msgstr "Verbindungen" 52 | 53 | #: ucp/views/phone.php:28 54 | msgid "DEF" 55 | msgstr "" 56 | 57 | #: ucp/assets/js/global.js:69 ucp/assets/js/global.js:77 58 | msgid "Disable" 59 | msgstr "" 60 | 61 | #: ucp/views/settings.php:7 62 | msgid "Disconnect Phone" 63 | msgstr "Telefonverbindung trennen" 64 | 65 | #: ucp/assets/js/global.js:554 66 | msgid "Disconnected" 67 | msgstr "" 68 | 69 | #: ucp/assets/js/global.js:68 ucp/assets/js/global.js:76 70 | msgid "Enable" 71 | msgstr "" 72 | 73 | #: views/ucp_config.php:19 74 | msgid "Enable Phone" 75 | msgstr "" 76 | 77 | #: functions.inc/hook_core.php:62 78 | msgid "Enable WebRTC Old ARI Phone" 79 | msgstr "WebRTC Old ARI-Telefon aktivieren" 80 | 81 | #: views/ucp_config.php:38 82 | msgid "Enable the phone in UCP for this user" 83 | msgstr "" 84 | 85 | #: functions.inc/hook_core.php:62 86 | #, php-format 87 | msgid "" 88 | "Enables WebRTC for this %s in the Asterisk Recording Interface (ARI). Note: " 89 | "ARI is depreciated in favor of UCP" 90 | msgstr "" 91 | "Aktiviert WebRTC für diesen %s in der Asterisk-Aufnahme-Schnittstelle (ARI). " 92 | "Hinweis: ARI ist UCP untergeordnet" 93 | 94 | #: ucp/views/phone.php:34 95 | msgid "GHI" 96 | msgstr "" 97 | 98 | #: Webrtc.class.php:372 99 | msgid "HTTPS is not enabled for Asterisk" 100 | msgstr "HTTPS ist für Asterisk nicht aktiviert" 101 | 102 | #: ucp/views/phone.php:81 103 | msgid "Hold" 104 | msgstr "" 105 | 106 | #: ucp/assets/js/global.js:266 107 | #, perl-format 108 | msgid "Incoming call from %s" 109 | msgstr "Eingehender Anruf von %s" 110 | 111 | #: views/ucp_config.php:29 112 | msgid "Inherit" 113 | msgstr "Übernehmen" 114 | 115 | #: ucp/views/phone.php:38 116 | msgid "JKL" 117 | msgstr "" 118 | 119 | #: ucp/views/phone.php:42 120 | msgid "MNO" 121 | msgstr "" 122 | 123 | #: views/ucp_config.php:26 124 | msgid "No" 125 | msgstr "Nein" 126 | 127 | #: ucp/assets/js/global.js:633 128 | msgid "Not supported in this browser" 129 | msgstr "" 130 | 131 | #: ucp/views/phone.php:88 132 | msgid "Note: An Active call will not be dropped when this window is closed" 133 | msgstr "" 134 | 135 | #: ucp/assets/js/global.js:640 136 | msgid "Only supported over HTTPS" 137 | msgstr "" 138 | 139 | #: ucp/views/phone.php:48 140 | msgid "PQRS" 141 | msgstr "" 142 | 143 | #: Webrtc.class.php:264 ucp/Webrtc.class.php:25 ucp/Webrtc.class.php:66 144 | msgid "Phone" 145 | msgstr "" 146 | 147 | #: ucp/views/phone.php:86 148 | msgid "Phone Status" 149 | msgstr "" 150 | 151 | #: ucp/assets/js/global.js:522 152 | msgid "Registered" 153 | msgstr "" 154 | 155 | #: ucp/assets/js/global.js:569 156 | msgid "Registering..." 157 | msgstr "" 158 | 159 | #: ucp/assets/js/global.js:540 160 | msgid "Registration Failed" 161 | msgstr "" 162 | 163 | #: Webrtc.class.php:281 164 | #, php-format 165 | msgid "" 166 | "Running an unsupported version of Asterisk. %s Detected Asterisk version: %s " 167 | msgstr "" 168 | 169 | #: ucp/views/settings.php:3 170 | msgid "Silence Ringer" 171 | msgstr "Stummschaltung" 172 | 173 | #: ucp/views/phone.php:52 174 | msgid "TUV" 175 | msgstr "" 176 | 177 | #: functions.inc/hook_core.php:74 178 | msgid "The Certificate to use from Certificate Manager" 179 | msgstr "Das zu verwendende Zertifikat des Zertifikatmanagers" 180 | 181 | #: webrtc.i18n.php:8 182 | msgid "The WebRTC Module allows an Administrator to enable a " 183 | msgstr "Das WebRTC-Modul ermöglicht einem Administrator die Aktivierung von " 184 | 185 | #: functions.inc/hook_core.php:58 186 | msgid "" 187 | "The WebSockets Interface is running through PJSIP, PJSIP is not supported at " 188 | "this time. Please enable the chan_sip driver (along with pjsip or by itself) " 189 | "or alert the FreePBX Developers" 190 | msgstr "" 191 | "Die WebSockets-Schnittstelle läuft mit PJSIP. PJSIP wird derzeit nicht " 192 | "unterstützt. Bitte aktivierenSie den chan_sip-Treiber (zusammen mit psjip " 193 | "oder einzeln) oder wenden Sie sich an die Entwickler von FreePBX" 194 | 195 | #: functions.inc/hook_core.php:79 196 | #, php-format 197 | msgid "" 198 | "To utilize WebRTC in ARI you must add at least one certificate %s through " 199 | "Certificate Manager" 200 | msgstr "" 201 | "Um WebRTC in ARI verwenden zu können, muss mindestens ein Zertifikat %s über " 202 | "den Zertifikatmanager hinzugefügt werden" 203 | 204 | #: ucp/assets/js/global.js:532 ucp/assets/js/global.js:549 205 | msgid "Unregistered" 206 | msgstr "" 207 | 208 | #: functions.inc/hook_core.php:73 209 | msgid "Use Certificate" 210 | msgstr "Zertifikat verwenden" 211 | 212 | #: ucp/views/phone.php:56 213 | msgid "WXYZ" 214 | msgstr "" 215 | 216 | #: ucp/Webrtc.class.php:74 217 | msgid "WebRTC" 218 | msgstr "WebRTC" 219 | 220 | #: webrtc.i18n.php:4 221 | msgid "WebRTC Phone" 222 | msgstr "WebRTC-Telefon" 223 | 224 | #: ucp/Webrtc.class.php:54 225 | msgid "Webrtc is not enabled for this extension." 226 | msgstr "" 227 | 228 | #: views/ucp_config.php:24 229 | msgid "Yes" 230 | msgstr "Ja" 231 | 232 | #: Webrtc.class.php:268 233 | #, php-format 234 | msgid "You have no default certificates setup in %s" 235 | msgstr "" 236 | 237 | #~ msgid "Unsupported Version of Asterisk, You need at least %s you have %s" 238 | #~ msgstr "" 239 | #~ "Nicht unterstützte Version von Asterisk. Sie benötigen mindestens %s. Sie " 240 | #~ "haben %s" 241 | 242 | #~ msgid "" 243 | #~ ")),-1===f?(this.headers[b].splice(c,1),void this.logger.warn('error " 244 | #~ "parsing " 245 | #~ msgstr "" 246 | #~ ")),-1===f?(this.headers[b].splice(c,1),void this.logger.warn('error " 247 | #~ "parsing " 248 | 249 | #~ msgid "Certificate WebRTC Phone should use for this user." 250 | #~ msgstr "" 251 | #~ "Für diesen Benutzer sollte ein Telefon mit WebRTC-Zertifikat verwendet " 252 | #~ "werden." 253 | 254 | #~ msgid "Enable WebRTC Phone" 255 | #~ msgstr "WebRTC-Telefon aktivieren" 256 | 257 | #~ msgid "Enable the WebRTC Phone in UCP for this user" 258 | #~ msgstr "WebRTC-Telefon in UCP für diesen Benutzer aktivieren" 259 | 260 | #~ msgid "New Web Phone Call" 261 | #~ msgstr "Neuer Online-Anruf" 262 | 263 | #~ msgid "Originate" 264 | #~ msgstr "Aufbauen" 265 | 266 | #~ msgid "Originate Call" 267 | #~ msgstr "Anruf aufbauen" 268 | 269 | #~ msgid "" 270 | #~ "Unable to start call. Please allow the WebRTC session in your browser and " 271 | #~ "refresh" 272 | #~ msgstr "" 273 | #~ "Anruf kann nicht gestartet werden. Erlauben Sie die WebRTC-Sitzung in " 274 | #~ "Ihrem Browser und aktualisieren Sie" 275 | 276 | #~ msgid "WebRTC Certificate" 277 | #~ msgstr "WebRTC-Zertifikat" 278 | 279 | #~ msgid "You have no certificates setup in Certificate Manager" 280 | #~ msgstr "Sie haben keine Zertifikate im Zertifikatmanager erstellt" 281 | -------------------------------------------------------------------------------- /i18n/es_ES/LC_MESSAGES/webrtc.mo: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/FreePBX/webrtc/92ce6acc8eced21055a5b63267b764a8609e4cae/i18n/es_ES/LC_MESSAGES/webrtc.mo -------------------------------------------------------------------------------- /i18n/es_ES/LC_MESSAGES/webrtc.po: -------------------------------------------------------------------------------- 1 | # This file is part of FreePBX. 2 | # 3 | # For licensing information, please see the file named LICENSE located in the module directory 4 | # 5 | # FreePBX language template for webrtc 6 | # Copyright (C) 2008-2015 Sangoma, Inc. 7 | # 8 | msgid "" 9 | msgstr "" 10 | "Project-Id-Version: PACKAGE VERSION\n" 11 | "Report-Msgid-Bugs-To: \n" 12 | "POT-Creation-Date: 2024-12-03 06:07-0500\n" 13 | "PO-Revision-Date: 2016-06-07 10:27+0200\n" 14 | "Last-Translator: Ernesto \n" 15 | "Language-Team: Spanish \n" 17 | "Language: es_ES\n" 18 | "MIME-Version: 1.0\n" 19 | "Content-Type: text/plain; charset=UTF-8\n" 20 | "Content-Transfer-Encoding: 8bit\n" 21 | "Plural-Forms: nplurals=2; plural=n != 1;\n" 22 | "X-Generator: Weblate 2.4\n" 23 | 24 | #: ucp/views/phone.php:24 25 | msgid "ABC" 26 | msgstr "" 27 | 28 | #: ucp/views/phone.php:78 29 | msgid "Call" 30 | msgstr "" 31 | 32 | #: Webrtc.class.php:268 33 | msgid "Certificate Manager" 34 | msgstr "" 35 | 36 | #: ucp/assets/js/global.js:267 37 | msgid "Click this window to answer or close this window to ignore" 38 | msgstr "" 39 | 40 | #: ucp/assets/js/global.js:542 41 | msgid "Connect Phone" 42 | msgstr "" 43 | 44 | #: ucp/assets/js/global.js:563 45 | msgid "Connecting to socket..." 46 | msgstr "" 47 | 48 | #: webrtc.i18n.php:6 49 | msgid "Connectivity" 50 | msgstr "Conectividad" 51 | 52 | #: ucp/views/phone.php:28 53 | msgid "DEF" 54 | msgstr "" 55 | 56 | #: ucp/assets/js/global.js:69 ucp/assets/js/global.js:77 57 | msgid "Disable" 58 | msgstr "" 59 | 60 | #: ucp/views/settings.php:7 61 | msgid "Disconnect Phone" 62 | msgstr "" 63 | 64 | #: ucp/assets/js/global.js:554 65 | msgid "Disconnected" 66 | msgstr "" 67 | 68 | #: ucp/assets/js/global.js:68 ucp/assets/js/global.js:76 69 | msgid "Enable" 70 | msgstr "" 71 | 72 | #: views/ucp_config.php:19 73 | msgid "Enable Phone" 74 | msgstr "" 75 | 76 | #: functions.inc/hook_core.php:62 77 | msgid "Enable WebRTC Old ARI Phone" 78 | msgstr "Habilitar WebRTC viejo ARI Phone" 79 | 80 | #: views/ucp_config.php:38 81 | msgid "Enable the phone in UCP for this user" 82 | msgstr "" 83 | 84 | #: functions.inc/hook_core.php:62 85 | #, php-format 86 | msgid "" 87 | "Enables WebRTC for this %s in the Asterisk Recording Interface (ARI). Note: " 88 | "ARI is depreciated in favor of UCP" 89 | msgstr "" 90 | "Habilita WebRTC para este %s en el Asterisk Recording Interface (ARI). Nota: " 91 | "ARI es obsoleto en favor de UCP" 92 | 93 | #: ucp/views/phone.php:34 94 | msgid "GHI" 95 | msgstr "" 96 | 97 | #: Webrtc.class.php:372 98 | msgid "HTTPS is not enabled for Asterisk" 99 | msgstr "" 100 | 101 | #: ucp/views/phone.php:81 102 | msgid "Hold" 103 | msgstr "" 104 | 105 | #: ucp/assets/js/global.js:266 106 | #, perl-format 107 | msgid "Incoming call from %s" 108 | msgstr "" 109 | 110 | #: views/ucp_config.php:29 111 | msgid "Inherit" 112 | msgstr "heredado" 113 | 114 | #: ucp/views/phone.php:38 115 | msgid "JKL" 116 | msgstr "" 117 | 118 | #: ucp/views/phone.php:42 119 | msgid "MNO" 120 | msgstr "" 121 | 122 | #: views/ucp_config.php:26 123 | msgid "No" 124 | msgstr "No" 125 | 126 | #: ucp/assets/js/global.js:633 127 | msgid "Not supported in this browser" 128 | msgstr "" 129 | 130 | #: ucp/views/phone.php:88 131 | msgid "Note: An Active call will not be dropped when this window is closed" 132 | msgstr "" 133 | 134 | #: ucp/assets/js/global.js:640 135 | msgid "Only supported over HTTPS" 136 | msgstr "" 137 | 138 | #: ucp/views/phone.php:48 139 | msgid "PQRS" 140 | msgstr "" 141 | 142 | #: Webrtc.class.php:264 ucp/Webrtc.class.php:25 ucp/Webrtc.class.php:66 143 | msgid "Phone" 144 | msgstr "" 145 | 146 | #: ucp/views/phone.php:86 147 | msgid "Phone Status" 148 | msgstr "" 149 | 150 | #: ucp/assets/js/global.js:522 151 | msgid "Registered" 152 | msgstr "" 153 | 154 | #: ucp/assets/js/global.js:569 155 | msgid "Registering..." 156 | msgstr "" 157 | 158 | #: ucp/assets/js/global.js:540 159 | msgid "Registration Failed" 160 | msgstr "" 161 | 162 | #: Webrtc.class.php:281 163 | #, php-format 164 | msgid "" 165 | "Running an unsupported version of Asterisk. %s Detected Asterisk version: %s " 166 | msgstr "" 167 | 168 | #: ucp/views/settings.php:3 169 | msgid "Silence Ringer" 170 | msgstr "" 171 | 172 | #: ucp/views/phone.php:52 173 | msgid "TUV" 174 | msgstr "" 175 | 176 | #: functions.inc/hook_core.php:74 177 | msgid "The Certificate to use from Certificate Manager" 178 | msgstr "Certificado a usar desde el Gestor de Certificados" 179 | 180 | #: webrtc.i18n.php:8 181 | msgid "The WebRTC Module allows an Administrator to enable a " 182 | msgstr "" 183 | 184 | #: functions.inc/hook_core.php:58 185 | msgid "" 186 | "The WebSockets Interface is running through PJSIP, PJSIP is not supported at " 187 | "this time. Please enable the chan_sip driver (along with pjsip or by itself) " 188 | "or alert the FreePBX Developers" 189 | msgstr "" 190 | "El WebSockets interfaz está ejecutando a través PJSIP, PJSIP no es soportado " 191 | "en este momento. Por favor, active el driver chan_sip (junto con PJSIP o por " 192 | "sí mismo) o alertar los desarrolladores de FreePBX" 193 | 194 | #: functions.inc/hook_core.php:79 195 | #, php-format 196 | msgid "" 197 | "To utilize WebRTC in ARI you must add at least one certificate %s through " 198 | "Certificate Manager" 199 | msgstr "" 200 | "Para utilizar WebRTC en ARI debe agregar al menos un certificado %s a través " 201 | "de Administrador de certificados" 202 | 203 | #: ucp/assets/js/global.js:532 ucp/assets/js/global.js:549 204 | msgid "Unregistered" 205 | msgstr "" 206 | 207 | #: functions.inc/hook_core.php:73 208 | msgid "Use Certificate" 209 | msgstr "Use Certificado" 210 | 211 | #: ucp/views/phone.php:56 212 | msgid "WXYZ" 213 | msgstr "" 214 | 215 | #: ucp/Webrtc.class.php:74 216 | msgid "WebRTC" 217 | msgstr "" 218 | 219 | #: webrtc.i18n.php:4 220 | msgid "WebRTC Phone" 221 | msgstr "Teléfono WebRTC" 222 | 223 | #: ucp/Webrtc.class.php:54 224 | msgid "Webrtc is not enabled for this extension." 225 | msgstr "" 226 | 227 | #: views/ucp_config.php:24 228 | msgid "Yes" 229 | msgstr "Si" 230 | 231 | #: Webrtc.class.php:268 232 | #, php-format 233 | msgid "You have no default certificates setup in %s" 234 | msgstr "" 235 | 236 | #~ msgid "Unsupported Version of Asterisk, You need at least %s you have %s" 237 | #~ msgstr "" 238 | #~ "versión de Asterisk no soportada. Usted necesita al menos %s y tiene %s" 239 | 240 | #~ msgid "Enable WebRTC Phone" 241 | #~ msgstr "Habilitar Teléfono WebRTC" 242 | 243 | #~ msgid "New Web Phone Call" 244 | #~ msgstr "Nueva llamada Web" 245 | 246 | #~ msgid "WebRTC Certificate" 247 | #~ msgstr "Certificado WebRTC" 248 | 249 | #~ msgid "Originate" 250 | #~ msgstr "Originar" 251 | 252 | #~ msgid "Originate Call" 253 | #~ msgstr "Originar Llamada" 254 | 255 | #~ msgid "" 256 | #~ "Unable to start call. Please allow the WebRTC session in your browser and " 257 | #~ "refresh" 258 | #~ msgstr "" 259 | #~ "no se pudo iniciar llamada. Por favor permita la sesión WebRTC en su " 260 | #~ "navegador y refresque" 261 | 262 | #~ msgid "" 263 | #~ "The STUN Server address is blank. In many cases this can cause issues. " 264 | #~ "Please define a valid server in the Asterisk SIP Settings module" 265 | #~ msgstr "" 266 | #~ "La dirección del servidor STUN esta en blanco. In muchos casos esto puede " 267 | #~ "causar problemas. Por favor defina un servidor valido en el modulo de " 268 | #~ "configuración de Asterisk" 269 | 270 | #~ msgid "Unsupported Version of Asterisk, You need at least %s you have &s" 271 | #~ msgstr "" 272 | #~ "Versión de Asterisk no soportada. Usted necesita al menos %s y tiene &s" 273 | 274 | #~ msgid "Allow Originating Calls" 275 | #~ msgstr "Permitir originar llamadas" 276 | 277 | #~ msgid "Allow calls to be originated from UCP" 278 | #~ msgstr "Permitir llamadas originadas desde UCP" 279 | 280 | #~ msgid "Nothing Entered" 281 | #~ msgstr "Nada ha sido entrado" 282 | 283 | #~ msgid "" 284 | #~ "The WebRTC Module allows an Administrator to enable a \"WebRTC phone\" " 285 | #~ "that can be attached to a user\\'s\n" 286 | #~ "\t\textension which they can connect to through FreePBX User Control " 287 | #~ "Panel, this WebRTC phone will then\n" 288 | #~ "\t\treceive phone calls at the same time as the users extension using " 289 | #~ "user and device mode behind the scenes.\n" 290 | #~ "\t\tIf you have User and Device Mode enabled any extension you enable the " 291 | #~ "WebRTC Phone a duplicate extension of 99XXXX\n" 292 | #~ "\t\twill be created (where XXXX is the original extension number), when " 293 | #~ "the user then views the web interface of the WebRTC\n" 294 | #~ "\t\tphone they will be connected to device 99XXXX which will receive " 295 | #~ "calls from the original extension" 296 | #~ msgstr "" 297 | #~ "El Módulo WebRTC permite un Administrador habilitar un \"teléfono " 298 | #~ "WebRTC\" que se puede asociar a un usuario↵\n" 299 | #~ "→→ extensión que se pueden conectar a través del Panel de Control de " 300 | #~ "usuario FreePBX, este teléfono WebRTC podrá entonces↵\n" 301 | #~ "→→ recibir llamadas de teléfono al mismo tiempo que la extensión de los " 302 | #~ "usuarios utilizando el modo de usuario y el dispositivo detrás de la " 303 | #~ "escenas.↵\n" 304 | #~ "→→ Si tiene el modo de usuario y dispositivo habilitado la extensión que " 305 | #~ "permita la WebRTC teléfono una extensión de duplicado de 99XXXX↵\n" 306 | #~ "→→ se crearán (donde XXXX es el número de la extensión original), cuando " 307 | #~ "el usuario se ve la interfaz web del WebRTC↵\n" 308 | #~ "→→ teléfono que estarán conectadas a un dispositivo 99XXXX que recibirá " 309 | #~ "las llamadas de la extensión original" 310 | 311 | #~ msgid "" 312 | #~ "Whether or not to enable the WebRTC Phone for this linked. Additionally " 313 | #~ "you must select a valid certificate to use." 314 | #~ msgstr "" 315 | #~ "Sea o no que permitir que el teléfono WebRTC para esto sea enlazado. " 316 | #~ "Además debe seleccionar un certificado válido para su uso." 317 | 318 | #~ msgid "Which certificate to use for the WebRTC Phone in UCP" 319 | #~ msgstr "Cual certificado a usar para el teléfono WebRTC en UCP" 320 | 321 | #~ msgid "" 322 | #~ "You have no certificates setup in Certificate Manager" 324 | #~ msgstr "" 325 | #~ "Usted no tiene certificados configurados en Certificate Manager" 327 | -------------------------------------------------------------------------------- /i18n/fa_IR/LC_MESSAGES/webrtc.mo: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/FreePBX/webrtc/92ce6acc8eced21055a5b63267b764a8609e4cae/i18n/fa_IR/LC_MESSAGES/webrtc.mo -------------------------------------------------------------------------------- /i18n/fa_IR/LC_MESSAGES/webrtc.po: -------------------------------------------------------------------------------- 1 | # This file is part of FreePBX. 2 | # 3 | # For licensing information, please see the file named LICENSE located in the module directory 4 | # 5 | # FreePBX language template for webrtc 6 | # Copyright (C) 2008-2015 Sangoma, Inc. 7 | # 8 | msgid "" 9 | msgstr "" 10 | "Project-Id-Version: PACKAGE VERSION\n" 11 | "Report-Msgid-Bugs-To: \n" 12 | "POT-Creation-Date: 2024-12-03 06:07-0500\n" 13 | "PO-Revision-Date: 2016-10-08 12:26+0200\n" 14 | "Last-Translator: psdk \n" 15 | "Language-Team: Persian (Iran) \n" 17 | "Language: fa_IR\n" 18 | "MIME-Version: 1.0\n" 19 | "Content-Type: text/plain; charset=UTF-8\n" 20 | "Content-Transfer-Encoding: 8bit\n" 21 | "Plural-Forms: nplurals=2; plural=n != 1;\n" 22 | "X-Generator: Weblate 2.4\n" 23 | 24 | #: ucp/views/phone.php:24 25 | msgid "ABC" 26 | msgstr "" 27 | 28 | #: ucp/views/phone.php:78 29 | msgid "Call" 30 | msgstr "" 31 | 32 | #: Webrtc.class.php:268 33 | msgid "Certificate Manager" 34 | msgstr "" 35 | 36 | #: ucp/assets/js/global.js:267 37 | msgid "Click this window to answer or close this window to ignore" 38 | msgstr "" 39 | "برای پاسخ به تماس روی این پنجره کلیک کنید یا برای رد تماس این پنجره را ببندید" 40 | 41 | #: ucp/assets/js/global.js:542 42 | msgid "Connect Phone" 43 | msgstr "اتصال تلفن" 44 | 45 | #: ucp/assets/js/global.js:563 46 | msgid "Connecting to socket..." 47 | msgstr "" 48 | 49 | #: webrtc.i18n.php:6 50 | msgid "Connectivity" 51 | msgstr "اتصال" 52 | 53 | #: ucp/views/phone.php:28 54 | msgid "DEF" 55 | msgstr "" 56 | 57 | #: ucp/assets/js/global.js:69 ucp/assets/js/global.js:77 58 | msgid "Disable" 59 | msgstr "" 60 | 61 | #: ucp/views/settings.php:7 62 | msgid "Disconnect Phone" 63 | msgstr "قطع کردن اتصال تلفن" 64 | 65 | #: ucp/assets/js/global.js:554 66 | msgid "Disconnected" 67 | msgstr "" 68 | 69 | #: ucp/assets/js/global.js:68 ucp/assets/js/global.js:76 70 | msgid "Enable" 71 | msgstr "" 72 | 73 | #: views/ucp_config.php:19 74 | msgid "Enable Phone" 75 | msgstr "" 76 | 77 | #: functions.inc/hook_core.php:62 78 | msgid "Enable WebRTC Old ARI Phone" 79 | msgstr "فعال سازی تلفن WebRTC قدیمی مبتنی بر ARI" 80 | 81 | #: views/ucp_config.php:38 82 | msgid "Enable the phone in UCP for this user" 83 | msgstr "" 84 | 85 | #: functions.inc/hook_core.php:62 86 | #, php-format 87 | msgid "" 88 | "Enables WebRTC for this %s in the Asterisk Recording Interface (ARI). Note: " 89 | "ARI is depreciated in favor of UCP" 90 | msgstr "" 91 | "فعال سازی WebRTC برای %s در ARI. توجه： با معرفی UCP، واسط ARI منسوخ شده است" 92 | 93 | #: ucp/views/phone.php:34 94 | msgid "GHI" 95 | msgstr "" 96 | 97 | #: Webrtc.class.php:372 98 | msgid "HTTPS is not enabled for Asterisk" 99 | msgstr "سرویس HTTPS برای استریسک فعال نشده است" 100 | 101 | #: ucp/views/phone.php:81 102 | msgid "Hold" 103 | msgstr "" 104 | 105 | #: ucp/assets/js/global.js:266 106 | #, perl-format 107 | msgid "Incoming call from %s" 108 | msgstr "تماس ورودی از %s" 109 | 110 | #: views/ucp_config.php:29 111 | msgid "Inherit" 112 | msgstr "موروثی" 113 | 114 | #: ucp/views/phone.php:38 115 | msgid "JKL" 116 | msgstr "" 117 | 118 | #: ucp/views/phone.php:42 119 | msgid "MNO" 120 | msgstr "" 121 | 122 | #: views/ucp_config.php:26 123 | msgid "No" 124 | msgstr "خیر" 125 | 126 | #: ucp/assets/js/global.js:633 127 | msgid "Not supported in this browser" 128 | msgstr "" 129 | 130 | #: ucp/views/phone.php:88 131 | msgid "Note: An Active call will not be dropped when this window is closed" 132 | msgstr "" 133 | 134 | #: ucp/assets/js/global.js:640 135 | msgid "Only supported over HTTPS" 136 | msgstr "" 137 | 138 | #: ucp/views/phone.php:48 139 | msgid "PQRS" 140 | msgstr "" 141 | 142 | #: Webrtc.class.php:264 ucp/Webrtc.class.php:25 ucp/Webrtc.class.php:66 143 | msgid "Phone" 144 | msgstr "" 145 | 146 | #: ucp/views/phone.php:86 147 | msgid "Phone Status" 148 | msgstr "" 149 | 150 | #: ucp/assets/js/global.js:522 151 | msgid "Registered" 152 | msgstr "" 153 | 154 | #: ucp/assets/js/global.js:569 155 | msgid "Registering..." 156 | msgstr "" 157 | 158 | #: ucp/assets/js/global.js:540 159 | msgid "Registration Failed" 160 | msgstr "" 161 | 162 | #: Webrtc.class.php:281 163 | #, php-format 164 | msgid "" 165 | "Running an unsupported version of Asterisk. %s Detected Asterisk version: %s " 166 | msgstr "" 167 | 168 | #: ucp/views/settings.php:3 169 | msgid "Silence Ringer" 170 | msgstr "سکوت زنگ" 171 | 172 | #: ucp/views/phone.php:52 173 | msgid "TUV" 174 | msgstr "" 175 | 176 | #: functions.inc/hook_core.php:74 177 | msgid "The Certificate to use from Certificate Manager" 178 | msgstr "مجوزی که از طریق مدیریت مجوزها، مورد استفاده قرار می گیرد" 179 | 180 | #: webrtc.i18n.php:8 181 | msgid "The WebRTC Module allows an Administrator to enable a " 182 | msgstr "ماژول WebRTC به یک مدیر اجازه می دهد که فعال کنید " 183 | 184 | #: functions.inc/hook_core.php:58 185 | msgid "" 186 | "The WebSockets Interface is running through PJSIP, PJSIP is not supported at " 187 | "this time. Please enable the chan_sip driver (along with pjsip or by itself) " 188 | "or alert the FreePBX Developers" 189 | msgstr "" 190 | "واسط وب سوکت در حال حاضر از طریق PJSIP اجرا می شود؛ PJSIP در حال حاضر " 191 | "پشتیبانی نمی شود. لطفا ماژول chan_sip را فعال کرده و از آن استفاده کنید" 192 | 193 | #: functions.inc/hook_core.php:79 194 | #, php-format 195 | msgid "" 196 | "To utilize WebRTC in ARI you must add at least one certificate %s through " 197 | "Certificate Manager" 198 | msgstr "" 199 | "برای استفاده از WebRTC در ARI می بایست حداقل یک مجوز %s از طریق ماژول مدیریت " 200 | "مجوز تعریف کنید" 201 | 202 | #: ucp/assets/js/global.js:532 ucp/assets/js/global.js:549 203 | msgid "Unregistered" 204 | msgstr "" 205 | 206 | #: functions.inc/hook_core.php:73 207 | msgid "Use Certificate" 208 | msgstr "از مجوز استفاده نمایید" 209 | 210 | #: ucp/views/phone.php:56 211 | msgid "WXYZ" 212 | msgstr "" 213 | 214 | #: ucp/Webrtc.class.php:74 215 | msgid "WebRTC" 216 | msgstr "ارتباط WebRTC" 217 | 218 | #: webrtc.i18n.php:4 219 | msgid "WebRTC Phone" 220 | msgstr "تلفن WebRTC" 221 | 222 | #: ucp/Webrtc.class.php:54 223 | msgid "Webrtc is not enabled for this extension." 224 | msgstr "" 225 | 226 | #: views/ucp_config.php:24 227 | msgid "Yes" 228 | msgstr "بله" 229 | 230 | #: Webrtc.class.php:268 231 | #, php-format 232 | msgid "You have no default certificates setup in %s" 233 | msgstr "" 234 | 235 | #~ msgid "Unsupported Version of Asterisk, You need at least %s you have %s" 236 | #~ msgstr "" 237 | #~ "عدم پشتیبانی از نسخه استریسک، شما باید حداقل نسخه %s را داشته باشید. در " 238 | #~ "حال حاضر نسخه شما %s است" 239 | 240 | #~ msgid "Certificate WebRTC Phone should use for this user." 241 | #~ msgstr "مجوزی که تلفن WebRTC برای این کاربر می بایست از آن استفاده کند." 242 | 243 | #~ msgid "Enable WebRTC Phone" 244 | #~ msgstr "فعال سازی تلفن WebRTC" 245 | 246 | #~ msgid "Enable the WebRTC Phone in UCP for this user" 247 | #~ msgstr "فعال سازی تلفن WebRTC در UCP برای این کاربر" 248 | 249 | #~ msgid "New Web Phone Call" 250 | #~ msgstr "تماس تحت تلفن وب جدید" 251 | 252 | #~ msgid "WebRTC Certificate" 253 | #~ msgstr "مجوز WebRTC" 254 | 255 | #~ msgid "You have no certificates setup in Certificate Manager" 256 | #~ msgstr "شما مجوزی در ماژول مدیریت مجوزها ایجاد نکرده اید" 257 | 258 | #~ msgid "Originate" 259 | #~ msgstr "شروع تماس" 260 | 261 | #~ msgid "Originate Call" 262 | #~ msgstr "ایجاد یک تماس" 263 | 264 | #~ msgid "" 265 | #~ ")),-1===f?(this.headers[b].splice(c,1),void this.logger.warn('error " 266 | #~ "parsing " 267 | #~ msgstr "" 268 | #~ ")),-1===f?(this.headers[b].splice(c,1),void this.logger.warn('خطا در " 269 | #~ "تجزیه " 270 | 271 | #~ msgid "" 272 | #~ "Unable to start call. Please allow the WebRTC session in your browser and " 273 | #~ "refresh" 274 | #~ msgstr "" 275 | #~ "عدم امکان برقراری تماس. لطفا به اجرای WebRTC در مرورگر خود مجوز داده و " 276 | #~ "مجددا صفحه را بارگذاری کنید" 277 | 278 | #~ msgid "" 279 | #~ "The STUN Server address is blank. In many cases this can cause issues. " 280 | #~ "Please define a valid server in the Asterisk SIP Settings module" 281 | #~ msgstr "" 282 | #~ "آدرس سرور STUN خالی است. در بسیاری از موارد این موضوع ممکن است مشکلاتی را " 283 | #~ "به وجود آورد. لطفا یک آدرس معتبر در ماژول تنظیمات SIP استریسک وارد کنید" 284 | 285 | #~ msgid "Unsupported Version of Asterisk, You need at least %s you have &s" 286 | #~ msgstr "" 287 | #~ "از نسخه استریسک پشتیبانی نمیکند, شما باید نسخه %s را داشته باشین نسخه شما " 288 | #~ "&s" 289 | -------------------------------------------------------------------------------- /i18n/fr_FR/LC_MESSAGES/webrtc.mo: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/FreePBX/webrtc/92ce6acc8eced21055a5b63267b764a8609e4cae/i18n/fr_FR/LC_MESSAGES/webrtc.mo -------------------------------------------------------------------------------- /i18n/fr_FR/LC_MESSAGES/webrtc.po: -------------------------------------------------------------------------------- 1 | # This file is part of FreePBX. 2 | # 3 | # For licensing information, please see the file named LICENSE located in the module directory 4 | # 5 | # FreePBX language template for webrtc 6 | # Copyright (C) 2008-2016 Sangoma, Inc. 7 | # 8 | msgid "" 9 | msgstr "" 10 | "Project-Id-Version: PACKAGE VERSION\n" 11 | "Report-Msgid-Bugs-To: \n" 12 | "POT-Creation-Date: 2024-12-03 06:07-0500\n" 13 | "PO-Revision-Date: 2016-01-31 05:57+0200\n" 14 | "Last-Translator: Nicolas Riendeau \n" 15 | "Language-Team: French \n" 17 | "Language: fr_FR\n" 18 | "MIME-Version: 1.0\n" 19 | "Content-Type: text/plain; charset=UTF-8\n" 20 | "Content-Transfer-Encoding: 8bit\n" 21 | "Plural-Forms: nplurals=2; plural=n > 1;\n" 22 | "X-Generator: Weblate 2.2-dev\n" 23 | 24 | #: ucp/views/phone.php:24 25 | msgid "ABC" 26 | msgstr "" 27 | 28 | #: ucp/views/phone.php:78 29 | msgid "Call" 30 | msgstr "" 31 | 32 | #: Webrtc.class.php:268 33 | msgid "Certificate Manager" 34 | msgstr "" 35 | 36 | #: ucp/assets/js/global.js:267 37 | msgid "Click this window to answer or close this window to ignore" 38 | msgstr "" 39 | 40 | #: ucp/assets/js/global.js:542 41 | msgid "Connect Phone" 42 | msgstr "" 43 | 44 | #: ucp/assets/js/global.js:563 45 | msgid "Connecting to socket..." 46 | msgstr "" 47 | 48 | #: webrtc.i18n.php:6 49 | msgid "Connectivity" 50 | msgstr "Connectivité" 51 | 52 | #: ucp/views/phone.php:28 53 | msgid "DEF" 54 | msgstr "" 55 | 56 | #: ucp/assets/js/global.js:69 ucp/assets/js/global.js:77 57 | msgid "Disable" 58 | msgstr "" 59 | 60 | #: ucp/views/settings.php:7 61 | msgid "Disconnect Phone" 62 | msgstr "" 63 | 64 | #: ucp/assets/js/global.js:554 65 | msgid "Disconnected" 66 | msgstr "" 67 | 68 | #: ucp/assets/js/global.js:68 ucp/assets/js/global.js:76 69 | msgid "Enable" 70 | msgstr "" 71 | 72 | #: views/ucp_config.php:19 73 | msgid "Enable Phone" 74 | msgstr "" 75 | 76 | #: functions.inc/hook_core.php:62 77 | msgid "Enable WebRTC Old ARI Phone" 78 | msgstr "Activer le téléphone WebRTC ARI (vieux)" 79 | 80 | #: views/ucp_config.php:38 81 | msgid "Enable the phone in UCP for this user" 82 | msgstr "" 83 | 84 | #: functions.inc/hook_core.php:62 85 | #, php-format 86 | msgid "" 87 | "Enables WebRTC for this %s in the Asterisk Recording Interface (ARI). Note: " 88 | "ARI is depreciated in favor of UCP" 89 | msgstr "" 90 | "Active WebRTC pour ce %s dans le 'Asterisk Recording Interface (ARI)'. " 91 | "Note : ARI est déprécié en faveur d'UCP" 92 | 93 | #: ucp/views/phone.php:34 94 | msgid "GHI" 95 | msgstr "" 96 | 97 | #: Webrtc.class.php:372 98 | msgid "HTTPS is not enabled for Asterisk" 99 | msgstr "" 100 | 101 | #: ucp/views/phone.php:81 102 | msgid "Hold" 103 | msgstr "" 104 | 105 | #: ucp/assets/js/global.js:266 106 | #, perl-format 107 | msgid "Incoming call from %s" 108 | msgstr "" 109 | 110 | #: views/ucp_config.php:29 111 | msgid "Inherit" 112 | msgstr "Hérite" 113 | 114 | #: ucp/views/phone.php:38 115 | msgid "JKL" 116 | msgstr "" 117 | 118 | #: ucp/views/phone.php:42 119 | msgid "MNO" 120 | msgstr "" 121 | 122 | #: views/ucp_config.php:26 123 | msgid "No" 124 | msgstr "Non" 125 | 126 | #: ucp/assets/js/global.js:633 127 | msgid "Not supported in this browser" 128 | msgstr "" 129 | 130 | #: ucp/views/phone.php:88 131 | msgid "Note: An Active call will not be dropped when this window is closed" 132 | msgstr "" 133 | 134 | #: ucp/assets/js/global.js:640 135 | msgid "Only supported over HTTPS" 136 | msgstr "" 137 | 138 | #: ucp/views/phone.php:48 139 | msgid "PQRS" 140 | msgstr "" 141 | 142 | #: Webrtc.class.php:264 ucp/Webrtc.class.php:25 ucp/Webrtc.class.php:66 143 | msgid "Phone" 144 | msgstr "" 145 | 146 | #: ucp/views/phone.php:86 147 | msgid "Phone Status" 148 | msgstr "" 149 | 150 | #: ucp/assets/js/global.js:522 151 | msgid "Registered" 152 | msgstr "" 153 | 154 | #: ucp/assets/js/global.js:569 155 | msgid "Registering..." 156 | msgstr "" 157 | 158 | #: ucp/assets/js/global.js:540 159 | msgid "Registration Failed" 160 | msgstr "" 161 | 162 | #: Webrtc.class.php:281 163 | #, php-format 164 | msgid "" 165 | "Running an unsupported version of Asterisk. %s Detected Asterisk version: %s " 166 | msgstr "" 167 | 168 | #: ucp/views/settings.php:3 169 | msgid "Silence Ringer" 170 | msgstr "" 171 | 172 | #: ucp/views/phone.php:52 173 | msgid "TUV" 174 | msgstr "" 175 | 176 | #: functions.inc/hook_core.php:74 177 | msgid "The Certificate to use from Certificate Manager" 178 | msgstr "Le certificat à utiliser du gestionnaire de certificat" 179 | 180 | #: webrtc.i18n.php:8 181 | msgid "The WebRTC Module allows an Administrator to enable a " 182 | msgstr "Le module WebRTC permet à un administrateur d'activer un " 183 | 184 | #: functions.inc/hook_core.php:58 185 | msgid "" 186 | "The WebSockets Interface is running through PJSIP, PJSIP is not supported at " 187 | "this time. Please enable the chan_sip driver (along with pjsip or by itself) " 188 | "or alert the FreePBX Developers" 189 | msgstr "" 190 | "L'interface WebSockets utilise actuellement PJSIP et celui-ci n'est " 191 | "actuellement pas supporté. S'il vous plaît, activez le pilote chan_sip (avec " 192 | "ou sans pjsip) ou alertez les développeurs de FreePBX" 193 | 194 | #: functions.inc/hook_core.php:79 195 | #, php-format 196 | msgid "" 197 | "To utilize WebRTC in ARI you must add at least one certificate %s through " 198 | "Certificate Manager" 199 | msgstr "" 200 | "Pour utiliser WebRTC avec ARI vous devez ajouter au moins un certificat %s " 201 | "par l'entremise du gestionnaire de certificat" 202 | 203 | #: ucp/assets/js/global.js:532 ucp/assets/js/global.js:549 204 | msgid "Unregistered" 205 | msgstr "" 206 | 207 | #: functions.inc/hook_core.php:73 208 | msgid "Use Certificate" 209 | msgstr "Utiliser le certificat" 210 | 211 | #: ucp/views/phone.php:56 212 | msgid "WXYZ" 213 | msgstr "" 214 | 215 | #: ucp/Webrtc.class.php:74 216 | msgid "WebRTC" 217 | msgstr "WebRTC" 218 | 219 | #: webrtc.i18n.php:4 220 | msgid "WebRTC Phone" 221 | msgstr "Téléphone WebRTC" 222 | 223 | #: ucp/Webrtc.class.php:54 224 | msgid "Webrtc is not enabled for this extension." 225 | msgstr "" 226 | 227 | #: views/ucp_config.php:24 228 | msgid "Yes" 229 | msgstr "Oui" 230 | 231 | #: Webrtc.class.php:268 232 | #, php-format 233 | msgid "You have no default certificates setup in %s" 234 | msgstr "" 235 | 236 | #~ msgid "Unsupported Version of Asterisk, You need at least %s you have %s" 237 | #~ msgstr "" 238 | #~ "Version non supportée d'Asterisk. Vous devez avoir au minimum %s et vous " 239 | #~ "avez %s" 240 | 241 | #~ msgid "Certificate WebRTC Phone should use for this user." 242 | #~ msgstr "" 243 | #~ "Le certificat que le téléphone WebRTC devrait utiliser pour cet " 244 | #~ "utilisateur." 245 | 246 | #~ msgid "Enable WebRTC Phone" 247 | #~ msgstr "Activer le téléphone WebRTC" 248 | 249 | #~ msgid "Enable the WebRTC Phone in UCP for this user" 250 | #~ msgstr "Activer le téléphone WebRTC dans l'UCP pour cet utilisateur" 251 | 252 | #~ msgid "New Web Phone Call" 253 | #~ msgstr "Nouvel appel téléphonique web" 254 | 255 | #~ msgid "WebRTC Certificate" 256 | #~ msgstr "Certificat WebRTC" 257 | 258 | #~ msgid "You have no certificates setup in Certificate Manager" 259 | #~ msgstr "" 260 | #~ "Vous n'avez pas de certificats configurés dans le gestionnaire de " 261 | #~ "certificat" 262 | 263 | #~ msgid "Originate" 264 | #~ msgstr "Placer un appel" 265 | 266 | #~ msgid "Originate Call" 267 | #~ msgstr "Placer un appel" 268 | 269 | #~ msgid "" 270 | #~ "Unable to start call. Please allow the WebRTC session in your browser and " 271 | #~ "refresh" 272 | #~ msgstr "" 273 | #~ "Incapable de démarrer l'appel. S'il vous plaît veuillez accepter la " 274 | #~ "sessions WebRTC dans votre navigateur et rafraîchir" 275 | 276 | #~ msgid "" 277 | #~ "The STUN Server address is blank. In many cases this can cause issues. " 278 | #~ "Please define a valid server in the Asterisk SIP Settings module" 279 | #~ msgstr "" 280 | #~ "L'adresse du serveur STUN est vide. Dans plusieurs cas, cela peut causer " 281 | #~ "des problèmes. S'il vous plait, veuillez définir un serveur valide dans " 282 | #~ "le module de réglages SIP d'Asterisk" 283 | 284 | #~ msgid "Unsupported Version of Asterisk, You need at least %s you have &s" 285 | #~ msgstr "" 286 | #~ "Version non supportée d'Asterisk. Vous devez avoir au minimum %s et vous " 287 | #~ "avez &s" 288 | -------------------------------------------------------------------------------- /i18n/pl/LC_MESSAGES/webrtc.mo: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/FreePBX/webrtc/92ce6acc8eced21055a5b63267b764a8609e4cae/i18n/pl/LC_MESSAGES/webrtc.mo -------------------------------------------------------------------------------- /i18n/pl/LC_MESSAGES/webrtc.po: -------------------------------------------------------------------------------- 1 | # This file is part of FreePBX. 2 | # 3 | # For licensing information, please see the file named LICENSE located in the module directory 4 | # 5 | # FreePBX language template for webrtc 6 | # Copyright (C) 2008-2021 Sangoma Technologies, Inc. 7 | # 8 | msgid "" 9 | msgstr "" 10 | "Project-Id-Version: PACKAGE VERSION\n" 11 | "Report-Msgid-Bugs-To: \n" 12 | "POT-Creation-Date: 2024-12-03 06:07-0500\n" 13 | "PO-Revision-Date: 2023-12-10 19:47+0000\n" 14 | "Last-Translator: Szymon Wysokinski \n" 15 | "Language-Team: Polish \n" 17 | "Language: pl\n" 18 | "MIME-Version: 1.0\n" 19 | "Content-Type: text/plain; charset=UTF-8\n" 20 | "Content-Transfer-Encoding: 8bit\n" 21 | "Plural-Forms: nplurals=3; plural=n==1 ? 0 : n%10>=2 && n%10<=4 && (n%100<10 " 22 | "|| n%100>=20) ? 1 : 2;\n" 23 | "X-Generator: Weblate 3.0.1\n" 24 | 25 | #: ucp/views/phone.php:24 26 | msgid "ABC" 27 | msgstr "ABC" 28 | 29 | #: ucp/views/phone.php:78 30 | msgid "Call" 31 | msgstr "Połączenie" 32 | 33 | #: Webrtc.class.php:268 34 | msgid "Certificate Manager" 35 | msgstr "Menedżer certyfikatów" 36 | 37 | #: ucp/assets/js/global.js:267 38 | msgid "Click this window to answer or close this window to ignore" 39 | msgstr "Kliknij to okno aby odpowiedzieć lub zamknij to okno aby zignorować" 40 | 41 | #: ucp/assets/js/global.js:542 42 | msgid "Connect Phone" 43 | msgstr "Podłącz telefon" 44 | 45 | #: ucp/assets/js/global.js:563 46 | msgid "Connecting to socket..." 47 | msgstr "Podłączanie do socketu..." 48 | 49 | #: webrtc.i18n.php:6 50 | msgid "Connectivity" 51 | msgstr "Łączność" 52 | 53 | #: ucp/views/phone.php:28 54 | msgid "DEF" 55 | msgstr "DEF" 56 | 57 | #: ucp/assets/js/global.js:69 ucp/assets/js/global.js:77 58 | msgid "Disable" 59 | msgstr "Wyłącz" 60 | 61 | #: ucp/views/settings.php:7 62 | msgid "Disconnect Phone" 63 | msgstr "Odłącz telefon" 64 | 65 | #: ucp/assets/js/global.js:554 66 | msgid "Disconnected" 67 | msgstr "Odłączony" 68 | 69 | #: ucp/assets/js/global.js:68 ucp/assets/js/global.js:76 70 | msgid "Enable" 71 | msgstr "Włącz" 72 | 73 | #: views/ucp_config.php:19 74 | msgid "Enable Phone" 75 | msgstr "Włącz telefon" 76 | 77 | #: functions.inc/hook_core.php:62 78 | msgid "Enable WebRTC Old ARI Phone" 79 | msgstr "Włącz stary telefon ARI WebRTC" 80 | 81 | #: views/ucp_config.php:38 82 | msgid "Enable the phone in UCP for this user" 83 | msgstr "Włącz telefon w UCP dla tego użytkownika" 84 | 85 | #: functions.inc/hook_core.php:62 86 | #, php-format 87 | msgid "" 88 | "Enables WebRTC for this %s in the Asterisk Recording Interface (ARI). Note: " 89 | "ARI is depreciated in favor of UCP" 90 | msgstr "" 91 | "Włącza WebRTC dla tego %s w interfejsie nagrywania Asterisk (ARI). Uwaga: " 92 | "ARI jest zastępowane przez UCP" 93 | 94 | #: ucp/views/phone.php:34 95 | msgid "GHI" 96 | msgstr "GHI" 97 | 98 | #: Webrtc.class.php:372 99 | msgid "HTTPS is not enabled for Asterisk" 100 | msgstr "HTTPS nie jest włączony w Asterisk" 101 | 102 | #: ucp/views/phone.php:81 103 | msgid "Hold" 104 | msgstr "Zawieś" 105 | 106 | #: ucp/assets/js/global.js:266 107 | #, perl-format 108 | msgid "Incoming call from %s" 109 | msgstr "Połączenie przychodzące od %s" 110 | 111 | #: views/ucp_config.php:29 112 | msgid "Inherit" 113 | msgstr "Odziedzicz" 114 | 115 | #: ucp/views/phone.php:38 116 | msgid "JKL" 117 | msgstr "JKL" 118 | 119 | #: ucp/views/phone.php:42 120 | msgid "MNO" 121 | msgstr "MNO" 122 | 123 | #: views/ucp_config.php:26 124 | msgid "No" 125 | msgstr "Nie" 126 | 127 | #: ucp/assets/js/global.js:633 128 | msgid "Not supported in this browser" 129 | msgstr "Nieobsługiwane w tej przeglądarce" 130 | 131 | #: ucp/views/phone.php:88 132 | msgid "Note: An Active call will not be dropped when this window is closed" 133 | msgstr "" 134 | "Uwaga: Po zamknięciu tego okna aktywne połączenie nie zostanie przerwane" 135 | 136 | #: ucp/assets/js/global.js:640 137 | msgid "Only supported over HTTPS" 138 | msgstr "Obsługiwane tylko przez HTTPS" 139 | 140 | #: ucp/views/phone.php:48 141 | msgid "PQRS" 142 | msgstr "PQRS" 143 | 144 | #: Webrtc.class.php:264 ucp/Webrtc.class.php:25 ucp/Webrtc.class.php:66 145 | msgid "Phone" 146 | msgstr "Telefon" 147 | 148 | #: ucp/views/phone.php:86 149 | msgid "Phone Status" 150 | msgstr "Stan telefonu" 151 | 152 | #: ucp/assets/js/global.js:522 153 | msgid "Registered" 154 | msgstr "Zarejestrowany" 155 | 156 | #: ucp/assets/js/global.js:569 157 | msgid "Registering..." 158 | msgstr "Rejestrowanie..." 159 | 160 | #: ucp/assets/js/global.js:540 161 | msgid "Registration Failed" 162 | msgstr "Rejestracja nie powiodła się" 163 | 164 | #: Webrtc.class.php:281 165 | #, php-format 166 | msgid "" 167 | "Running an unsupported version of Asterisk. %s Detected Asterisk version: %s " 168 | msgstr "" 169 | 170 | #: ucp/views/settings.php:3 171 | msgid "Silence Ringer" 172 | msgstr "Wycisz dzwonek" 173 | 174 | #: ucp/views/phone.php:52 175 | msgid "TUV" 176 | msgstr "TUV" 177 | 178 | #: functions.inc/hook_core.php:74 179 | msgid "The Certificate to use from Certificate Manager" 180 | msgstr "Certyfikat z Menedżera Certyfikatów który ma być używany" 181 | 182 | #: webrtc.i18n.php:8 183 | msgid "The WebRTC Module allows an Administrator to enable a " 184 | msgstr "Moduł WebRTC umożliwia administratorowi włączenie " 185 | 186 | #: functions.inc/hook_core.php:58 187 | msgid "" 188 | "The WebSockets Interface is running through PJSIP, PJSIP is not supported at " 189 | "this time. Please enable the chan_sip driver (along with pjsip or by itself) " 190 | "or alert the FreePBX Developers" 191 | msgstr "" 192 | "Interfejs WebSockets działa poprzez protokół PJSIP. PJSIP nie jest obecnie " 193 | "obsługiwany. Włącz sterownik chan_sip (wraz z pjsip lub sam) lub powiadom " 194 | "programistów FreePBX" 195 | 196 | #: functions.inc/hook_core.php:79 197 | #, php-format 198 | msgid "" 199 | "To utilize WebRTC in ARI you must add at least one certificate %s through " 200 | "Certificate Manager" 201 | msgstr "" 202 | "Aby korzystać z WebRTC w ARI, musisz dodać co najmniej jeden certyfikat %s " 203 | "poprzez Menedżera Certyfikatów" 204 | 205 | #: ucp/assets/js/global.js:532 ucp/assets/js/global.js:549 206 | msgid "Unregistered" 207 | msgstr "Niezarejestrowany" 208 | 209 | #: functions.inc/hook_core.php:73 210 | msgid "Use Certificate" 211 | msgstr "Użyj certyfikatu" 212 | 213 | #: ucp/views/phone.php:56 214 | msgid "WXYZ" 215 | msgstr "WXYZ" 216 | 217 | #: ucp/Webrtc.class.php:74 218 | msgid "WebRTC" 219 | msgstr "Web RTC" 220 | 221 | #: webrtc.i18n.php:4 222 | msgid "WebRTC Phone" 223 | msgstr "Telefon WebRTC" 224 | 225 | #: ucp/Webrtc.class.php:54 226 | msgid "Webrtc is not enabled for this extension." 227 | msgstr "" 228 | 229 | #: views/ucp_config.php:24 230 | msgid "Yes" 231 | msgstr "Tak" 232 | 233 | #: Webrtc.class.php:268 234 | #, php-format 235 | msgid "You have no default certificates setup in %s" 236 | msgstr "Nie masz domyślnych ustawień certyfikatów w %s" 237 | 238 | #, php-format 239 | #~ msgid "Unsupported Version of Asterisk, You need at least %s you have %s" 240 | #~ msgstr "Nieobsługiwana wersja Asterisk. Potrzebujesz co najmniej %s masz %s" 241 | -------------------------------------------------------------------------------- /i18n/pt_BR/LC_MESSAGES/webrtc.mo: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/FreePBX/webrtc/92ce6acc8eced21055a5b63267b764a8609e4cae/i18n/pt_BR/LC_MESSAGES/webrtc.mo -------------------------------------------------------------------------------- /i18n/pt_BR/LC_MESSAGES/webrtc.po: -------------------------------------------------------------------------------- 1 | # This file is part of FreePBX. 2 | # 3 | # For licensing information, please see the file named LICENSE located in the module directory 4 | # 5 | # FreePBX language template for webrtc 6 | # Copyright (C) 2008-2016 Sangoma, Inc. 7 | # 8 | msgid "" 9 | msgstr "" 10 | "Project-Id-Version: PACKAGE VERSION\n" 11 | "Report-Msgid-Bugs-To: \n" 12 | "POT-Creation-Date: 2024-12-03 06:07-0500\n" 13 | "PO-Revision-Date: 2016-12-20 23:26+0200\n" 14 | "Last-Translator: Alexander \n" 15 | "Language-Team: Portuguese (Brazil) \n" 17 | "Language: pt_BR\n" 18 | "MIME-Version: 1.0\n" 19 | "Content-Type: text/plain; charset=UTF-8\n" 20 | "Content-Transfer-Encoding: 8bit\n" 21 | "Plural-Forms: nplurals=2; plural=n != 1;\n" 22 | "X-Generator: Weblate 2.4\n" 23 | 24 | #: ucp/views/phone.php:24 25 | msgid "ABC" 26 | msgstr "" 27 | 28 | #: ucp/views/phone.php:78 29 | msgid "Call" 30 | msgstr "" 31 | 32 | #: Webrtc.class.php:268 33 | msgid "Certificate Manager" 34 | msgstr "" 35 | 36 | #: ucp/assets/js/global.js:267 37 | msgid "Click this window to answer or close this window to ignore" 38 | msgstr "Clique nesta janela para responder ou feche esta janela para ignorar" 39 | 40 | #: ucp/assets/js/global.js:542 41 | msgid "Connect Phone" 42 | msgstr "Conectar Telefone" 43 | 44 | #: ucp/assets/js/global.js:563 45 | msgid "Connecting to socket..." 46 | msgstr "" 47 | 48 | #: webrtc.i18n.php:6 49 | msgid "Connectivity" 50 | msgstr "Conectividade" 51 | 52 | #: ucp/views/phone.php:28 53 | msgid "DEF" 54 | msgstr "" 55 | 56 | #: ucp/assets/js/global.js:69 ucp/assets/js/global.js:77 57 | msgid "Disable" 58 | msgstr "" 59 | 60 | #: ucp/views/settings.php:7 61 | msgid "Disconnect Phone" 62 | msgstr "Desconectar Telefone" 63 | 64 | #: ucp/assets/js/global.js:554 65 | msgid "Disconnected" 66 | msgstr "" 67 | 68 | #: ucp/assets/js/global.js:68 ucp/assets/js/global.js:76 69 | msgid "Enable" 70 | msgstr "" 71 | 72 | #: views/ucp_config.php:19 73 | msgid "Enable Phone" 74 | msgstr "" 75 | 76 | #: functions.inc/hook_core.php:62 77 | msgid "Enable WebRTC Old ARI Phone" 78 | msgstr "Habilitar WebRTC antigo Telefone ARI" 79 | 80 | #: views/ucp_config.php:38 81 | msgid "Enable the phone in UCP for this user" 82 | msgstr "" 83 | 84 | #: functions.inc/hook_core.php:62 85 | #, php-format 86 | msgid "" 87 | "Enables WebRTC for this %s in the Asterisk Recording Interface (ARI). Note: " 88 | "ARI is depreciated in favor of UCP" 89 | msgstr "" 90 | "Habilita o WebRTC para este %s na Interface de Gravação Asterisk (ARI). " 91 | "Nota: ARI obsoleto em favor do PCU" 92 | 93 | #: ucp/views/phone.php:34 94 | msgid "GHI" 95 | msgstr "" 96 | 97 | #: Webrtc.class.php:372 98 | msgid "HTTPS is not enabled for Asterisk" 99 | msgstr "HTTPS não está habilitado para o Asterisk" 100 | 101 | #: ucp/views/phone.php:81 102 | msgid "Hold" 103 | msgstr "" 104 | 105 | #: ucp/assets/js/global.js:266 106 | #, perl-format 107 | msgid "Incoming call from %s" 108 | msgstr "Chamada recebida de %s" 109 | 110 | #: views/ucp_config.php:29 111 | msgid "Inherit" 112 | msgstr "Herdar" 113 | 114 | #: ucp/views/phone.php:38 115 | msgid "JKL" 116 | msgstr "" 117 | 118 | #: ucp/views/phone.php:42 119 | msgid "MNO" 120 | msgstr "" 121 | 122 | #: views/ucp_config.php:26 123 | msgid "No" 124 | msgstr "Não" 125 | 126 | #: ucp/assets/js/global.js:633 127 | msgid "Not supported in this browser" 128 | msgstr "" 129 | 130 | #: ucp/views/phone.php:88 131 | msgid "Note: An Active call will not be dropped when this window is closed" 132 | msgstr "" 133 | 134 | #: ucp/assets/js/global.js:640 135 | msgid "Only supported over HTTPS" 136 | msgstr "" 137 | 138 | #: ucp/views/phone.php:48 139 | msgid "PQRS" 140 | msgstr "" 141 | 142 | #: Webrtc.class.php:264 ucp/Webrtc.class.php:25 ucp/Webrtc.class.php:66 143 | msgid "Phone" 144 | msgstr "" 145 | 146 | #: ucp/views/phone.php:86 147 | msgid "Phone Status" 148 | msgstr "" 149 | 150 | #: ucp/assets/js/global.js:522 151 | msgid "Registered" 152 | msgstr "" 153 | 154 | #: ucp/assets/js/global.js:569 155 | msgid "Registering..." 156 | msgstr "" 157 | 158 | #: ucp/assets/js/global.js:540 159 | msgid "Registration Failed" 160 | msgstr "" 161 | 162 | #: Webrtc.class.php:281 163 | #, php-format 164 | msgid "" 165 | "Running an unsupported version of Asterisk. %s Detected Asterisk version: %s " 166 | msgstr "" 167 | 168 | #: ucp/views/settings.php:3 169 | msgid "Silence Ringer" 170 | msgstr "Silenciar Toque" 171 | 172 | #: ucp/views/phone.php:52 173 | msgid "TUV" 174 | msgstr "" 175 | 176 | #: functions.inc/hook_core.php:74 177 | msgid "The Certificate to use from Certificate Manager" 178 | msgstr "Certificado a ser usado pelo Gerenciador de Certificados" 179 | 180 | #: webrtc.i18n.php:8 181 | msgid "The WebRTC Module allows an Administrator to enable a " 182 | msgstr "O Módulo WebRTC permite que um Administrador habilite um " 183 | 184 | #: functions.inc/hook_core.php:58 185 | msgid "" 186 | "The WebSockets Interface is running through PJSIP, PJSIP is not supported at " 187 | "this time. Please enable the chan_sip driver (along with pjsip or by itself) " 188 | "or alert the FreePBX Developers" 189 | msgstr "" 190 | "A Interface do WebSockets está sendo executada através de PJSIP, PJSIP não é " 191 | "suportado neste momento. Habilite o driver chan_sip (juntamente com o pjsip " 192 | "ou sozinho) ou avise os Desenvolvedores FreePBX" 193 | 194 | #: functions.inc/hook_core.php:79 195 | #, php-format 196 | msgid "" 197 | "To utilize WebRTC in ARI you must add at least one certificate %s through " 198 | "Certificate Manager" 199 | msgstr "" 200 | "Para utilizar o WebRTC no ARI, você deve adicionar pelo menos um certificado " 201 | "%s através do Gerenciador de Certificados" 202 | 203 | #: ucp/assets/js/global.js:532 ucp/assets/js/global.js:549 204 | msgid "Unregistered" 205 | msgstr "" 206 | 207 | #: functions.inc/hook_core.php:73 208 | msgid "Use Certificate" 209 | msgstr "Usar Certificado" 210 | 211 | #: ucp/views/phone.php:56 212 | msgid "WXYZ" 213 | msgstr "" 214 | 215 | #: ucp/Webrtc.class.php:74 216 | msgid "WebRTC" 217 | msgstr "WebRTC" 218 | 219 | #: webrtc.i18n.php:4 220 | msgid "WebRTC Phone" 221 | msgstr "Telefone WebRTC" 222 | 223 | #: ucp/Webrtc.class.php:54 224 | msgid "Webrtc is not enabled for this extension." 225 | msgstr "" 226 | 227 | #: views/ucp_config.php:24 228 | msgid "Yes" 229 | msgstr "Sim" 230 | 231 | #: Webrtc.class.php:268 232 | #, php-format 233 | msgid "You have no default certificates setup in %s" 234 | msgstr "" 235 | 236 | #~ msgid "Unsupported Version of Asterisk, You need at least %s you have %s" 237 | #~ msgstr "" 238 | #~ "Versão não suportada do Asterisk, você precisa ao menos %s você tem %s" 239 | 240 | #~ msgid "Certificate WebRTC Phone should use for this user." 241 | #~ msgstr "" 242 | #~ "Certificado de Telefone WebRTC deve ser utilizado para este usuário." 243 | 244 | #~ msgid "Enable WebRTC Phone" 245 | #~ msgstr "Habilitar Telefone WebRTC" 246 | 247 | #~ msgid "Enable the WebRTC Phone in UCP for this user" 248 | #~ msgstr "Habilitar telefone WebRTC no PCU para este usuário" 249 | 250 | #~ msgid "New Web Phone Call" 251 | #~ msgstr "Nova Chamada Telefônica Web" 252 | 253 | #~ msgid "WebRTC Certificate" 254 | #~ msgstr "Certificado WebRTC" 255 | 256 | #~ msgid "You have no certificates setup in Certificate Manager" 257 | #~ msgstr "" 258 | #~ "Você não tem nenhuma configuração de certificados no Gerenciador de " 259 | #~ "Certificados" 260 | 261 | #~ msgid "" 262 | #~ ")),-1===f?(this.headers[b].splice(c,1),void this.logger.warn('error " 263 | #~ "parsing " 264 | #~ msgstr "" 265 | #~ ")),-1===f?(this.headers[b].splice(c,1),void this.logger.warn('error " 266 | #~ "parsing " 267 | 268 | #~ msgid "Originate" 269 | #~ msgstr "Originar" 270 | 271 | #~ msgid "Originate Call" 272 | #~ msgstr "Originar Chamada" 273 | 274 | #~ msgid "" 275 | #~ "Unable to start call. Please allow the WebRTC session in your browser and " 276 | #~ "refresh" 277 | #~ msgstr "" 278 | #~ "Não é possível iniciar a chamada. Permita a sessão WebRTC no seu " 279 | #~ "navegador e atualize" 280 | -------------------------------------------------------------------------------- /i18n/ru/LC_MESSAGES/webrtc.mo: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/FreePBX/webrtc/92ce6acc8eced21055a5b63267b764a8609e4cae/i18n/ru/LC_MESSAGES/webrtc.mo -------------------------------------------------------------------------------- /i18n/ru/LC_MESSAGES/webrtc.po: -------------------------------------------------------------------------------- 1 | # This file is part of FreePBX. 2 | # 3 | # For licensing information, please see the file named LICENSE located in the module directory 4 | # 5 | # FreePBX language template for webrtc 6 | # Copyright (C) 2008-2021 Sangoma Technologies, Inc. 7 | # 8 | msgid "" 9 | msgstr "" 10 | "Project-Id-Version: PACKAGE VERSION\n" 11 | "Report-Msgid-Bugs-To: \n" 12 | "POT-Creation-Date: 2024-12-03 06:07-0500\n" 13 | "PO-Revision-Date: 2022-07-20 07:50+0000\n" 14 | "Last-Translator: ded \n" 15 | "Language-Team: Russian \n" 17 | "Language: ru\n" 18 | "MIME-Version: 1.0\n" 19 | "Content-Type: text/plain; charset=UTF-8\n" 20 | "Content-Transfer-Encoding: 8bit\n" 21 | "Plural-Forms: nplurals=3; plural=n%10==1 && n%100!=11 ? 0 : n%10>=2 && " 22 | "n%10<=4 && (n%100<10 || n%100>=20) ? 1 : 2;\n" 23 | "X-Generator: Weblate 3.0.1\n" 24 | 25 | #: ucp/views/phone.php:24 26 | msgid "ABC" 27 | msgstr "ABC" 28 | 29 | #: ucp/views/phone.php:78 30 | msgid "Call" 31 | msgstr "Вызов" 32 | 33 | #: Webrtc.class.php:268 34 | msgid "Certificate Manager" 35 | msgstr "Менеджер сертификатов" 36 | 37 | #: ucp/assets/js/global.js:267 38 | msgid "Click this window to answer or close this window to ignore" 39 | msgstr "Щелкните это окно, чтобы ответить, или закройте чтобы проигнорировать" 40 | 41 | #: ucp/assets/js/global.js:542 42 | msgid "Connect Phone" 43 | msgstr "Подключить телефон" 44 | 45 | #: ucp/assets/js/global.js:563 46 | msgid "Connecting to socket..." 47 | msgstr "Подключение к сокету..." 48 | 49 | #: webrtc.i18n.php:6 50 | msgid "Connectivity" 51 | msgstr "Подключения" 52 | 53 | #: ucp/views/phone.php:28 54 | msgid "DEF" 55 | msgstr "DEF" 56 | 57 | #: ucp/assets/js/global.js:69 ucp/assets/js/global.js:77 58 | msgid "Disable" 59 | msgstr "Выключить" 60 | 61 | #: ucp/views/settings.php:7 62 | msgid "Disconnect Phone" 63 | msgstr "ОТключить телефон" 64 | 65 | #: ucp/assets/js/global.js:554 66 | msgid "Disconnected" 67 | msgstr "Отключено" 68 | 69 | #: ucp/assets/js/global.js:68 ucp/assets/js/global.js:76 70 | msgid "Enable" 71 | msgstr "Включить" 72 | 73 | #: views/ucp_config.php:19 74 | msgid "Enable Phone" 75 | msgstr "Включить телефон" 76 | 77 | #: functions.inc/hook_core.php:62 78 | msgid "Enable WebRTC Old ARI Phone" 79 | msgstr "Использовать телефон со старым WebRTC ARI" 80 | 81 | #: views/ucp_config.php:38 82 | msgid "Enable the phone in UCP for this user" 83 | msgstr "Использовать телефон в Панели пользователя этого пользователя" 84 | 85 | #: functions.inc/hook_core.php:62 86 | #, php-format 87 | msgid "" 88 | "Enables WebRTC for this %s in the Asterisk Recording Interface (ARI). Note: " 89 | "ARI is depreciated in favor of UCP" 90 | msgstr "" 91 | "Мспользовать WebRTC для %s в Asterisk Recording Interface (ARI). " 92 | "Примечание: использование ARI устарело в пользу UCP" 93 | 94 | #: ucp/views/phone.php:34 95 | msgid "GHI" 96 | msgstr "GHI" 97 | 98 | #: Webrtc.class.php:372 99 | msgid "HTTPS is not enabled for Asterisk" 100 | msgstr "HTTPS не задействовано в Asterisk" 101 | 102 | #: ucp/views/phone.php:81 103 | msgid "Hold" 104 | msgstr "удержание" 105 | 106 | #: ucp/assets/js/global.js:266 107 | #, perl-format 108 | msgid "Incoming call from %s" 109 | msgstr "Входящий вызов от %s" 110 | 111 | #: views/ucp_config.php:29 112 | msgid "Inherit" 113 | msgstr "Наследовать" 114 | 115 | #: ucp/views/phone.php:38 116 | msgid "JKL" 117 | msgstr "JKL" 118 | 119 | #: ucp/views/phone.php:42 120 | msgid "MNO" 121 | msgstr "MNO" 122 | 123 | #: views/ucp_config.php:26 124 | msgid "No" 125 | msgstr "Нет" 126 | 127 | #: ucp/assets/js/global.js:633 128 | msgid "Not supported in this browser" 129 | msgstr "Не поддерживается данным браузером" 130 | 131 | #: ucp/views/phone.php:88 132 | msgid "Note: An Active call will not be dropped when this window is closed" 133 | msgstr "Примечание: активный вызов не сбрасывается призакрытиии этого окна" 134 | 135 | #: ucp/assets/js/global.js:640 136 | msgid "Only supported over HTTPS" 137 | msgstr "Поддерживается только с использованием HTTPS" 138 | 139 | #: ucp/views/phone.php:48 140 | msgid "PQRS" 141 | msgstr "PQRS" 142 | 143 | #: Webrtc.class.php:264 ucp/Webrtc.class.php:25 ucp/Webrtc.class.php:66 144 | msgid "Phone" 145 | msgstr "Телефон" 146 | 147 | #: ucp/views/phone.php:86 148 | msgid "Phone Status" 149 | msgstr "Статус телефона" 150 | 151 | #: ucp/assets/js/global.js:522 152 | msgid "Registered" 153 | msgstr "Зарегистрирован" 154 | 155 | #: ucp/assets/js/global.js:569 156 | msgid "Registering..." 157 | msgstr "Регистрируется..." 158 | 159 | #: ucp/assets/js/global.js:540 160 | msgid "Registration Failed" 161 | msgstr "Регистрация отклонена" 162 | 163 | #: Webrtc.class.php:281 164 | #, php-format 165 | msgid "" 166 | "Running an unsupported version of Asterisk. %s Detected Asterisk version: %s " 167 | msgstr "" 168 | 169 | #: ucp/views/settings.php:3 170 | msgid "Silence Ringer" 171 | msgstr "Тихий звонок" 172 | 173 | #: ucp/views/phone.php:52 174 | msgid "TUV" 175 | msgstr "TUV" 176 | 177 | #: functions.inc/hook_core.php:74 178 | msgid "The Certificate to use from Certificate Manager" 179 | msgstr "Сертификат для использования из Менеджера сертификатов" 180 | 181 | #: webrtc.i18n.php:8 182 | msgid "The WebRTC Module allows an Administrator to enable a " 183 | msgstr "Модуль WebRTC делает возможным задействовать Администратору " 184 | 185 | #: functions.inc/hook_core.php:58 186 | msgid "" 187 | "The WebSockets Interface is running through PJSIP, PJSIP is not supported at " 188 | "this time. Please enable the chan_sip driver (along with pjsip or by itself) " 189 | "or alert the FreePBX Developers" 190 | msgstr "" 191 | "Интерфейс вэбсокета работает через PJSIP, Но PJSIP не поддерживается в " 192 | "текущей версии. Задействуйте драйвер канала chan_sip (может работать вместе " 193 | "с pjsip или самостоятельно) или сообщите об этом разработчикам FreePBX" 194 | 195 | #: functions.inc/hook_core.php:79 196 | #, php-format 197 | msgid "" 198 | "To utilize WebRTC in ARI you must add at least one certificate %s through " 199 | "Certificate Manager" 200 | msgstr "" 201 | "Для реализации WebRTC в составе с ARI нужно добавить по крайней мере один " 202 | "сертификат %s через модуль управления сертификатами" 203 | 204 | #: ucp/assets/js/global.js:532 ucp/assets/js/global.js:549 205 | msgid "Unregistered" 206 | msgstr "Не зарегистрирован" 207 | 208 | #: functions.inc/hook_core.php:73 209 | msgid "Use Certificate" 210 | msgstr "Используемый сертификат" 211 | 212 | #: ucp/views/phone.php:56 213 | msgid "WXYZ" 214 | msgstr "WXYZ" 215 | 216 | #: ucp/Webrtc.class.php:74 217 | msgid "WebRTC" 218 | msgstr "WebRTC" 219 | 220 | #: webrtc.i18n.php:4 221 | msgid "WebRTC Phone" 222 | msgstr "Телефон WebRTC" 223 | 224 | #: ucp/Webrtc.class.php:54 225 | msgid "Webrtc is not enabled for this extension." 226 | msgstr "" 227 | 228 | #: views/ucp_config.php:24 229 | msgid "Yes" 230 | msgstr "Да" 231 | 232 | #: Webrtc.class.php:268 233 | #, php-format 234 | msgid "You have no default certificates setup in %s" 235 | msgstr "Нет ни одного сертификата по умолчанию в %s" 236 | 237 | #~ msgid "Unsupported Version of Asterisk, You need at least %s you have %s" 238 | #~ msgstr "" 239 | #~ "Неподдерживаемая версия Астериск. Требуется по крайней мере %s, у вас же " 240 | #~ "имеется %s" 241 | -------------------------------------------------------------------------------- /i18n/uk_UA/LC_MESSAGES/webrtc.mo: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/FreePBX/webrtc/92ce6acc8eced21055a5b63267b764a8609e4cae/i18n/uk_UA/LC_MESSAGES/webrtc.mo -------------------------------------------------------------------------------- /i18n/uk_UA/LC_MESSAGES/webrtc.po: -------------------------------------------------------------------------------- 1 | # This file is part of FreePBX. 2 | # 3 | # For licensing information, please see the file named LICENSE located in the module directory 4 | # 5 | # FreePBX language template for webrtc 6 | # Copyright (C) 2008-2016 Sangoma, Inc. 7 | # 8 | msgid "" 9 | msgstr "" 10 | "Project-Id-Version: PACKAGE VERSION\n" 11 | "Report-Msgid-Bugs-To: \n" 12 | "POT-Creation-Date: 2024-12-03 06:07-0500\n" 13 | "PO-Revision-Date: 2016-10-05 20:09+0200\n" 14 | "Last-Translator: Maxim \n" 15 | "Language-Team: Ukrainian \n" 17 | "Language: uk_UA\n" 18 | "MIME-Version: 1.0\n" 19 | "Content-Type: text/plain; charset=UTF-8\n" 20 | "Content-Transfer-Encoding: 8bit\n" 21 | "Plural-Forms: nplurals=3; plural=n%10==1 && n%100!=11 ? 0 : n%10>=2 && " 22 | "n%10<=4 && (n%100<10 || n%100>=20) ? 1 : 2;\n" 23 | "X-Generator: Weblate 2.4\n" 24 | 25 | #: ucp/views/phone.php:24 26 | msgid "ABC" 27 | msgstr "" 28 | 29 | #: ucp/views/phone.php:78 30 | msgid "Call" 31 | msgstr "" 32 | 33 | #: Webrtc.class.php:268 34 | msgid "Certificate Manager" 35 | msgstr "" 36 | 37 | #: ucp/assets/js/global.js:267 38 | msgid "Click this window to answer or close this window to ignore" 39 | msgstr "" 40 | "Натисніть на це вікно, щоб відповісти або зачинить його, щоб ігнорувати" 41 | 42 | #: ucp/assets/js/global.js:542 43 | msgid "Connect Phone" 44 | msgstr "Підключення телефону" 45 | 46 | #: ucp/assets/js/global.js:563 47 | msgid "Connecting to socket..." 48 | msgstr "" 49 | 50 | #: webrtc.i18n.php:6 51 | msgid "Connectivity" 52 | msgstr "Зв'язок" 53 | 54 | #: ucp/views/phone.php:28 55 | msgid "DEF" 56 | msgstr "" 57 | 58 | #: ucp/assets/js/global.js:69 ucp/assets/js/global.js:77 59 | msgid "Disable" 60 | msgstr "" 61 | 62 | #: ucp/views/settings.php:7 63 | msgid "Disconnect Phone" 64 | msgstr "Від'єднайте телефон" 65 | 66 | #: ucp/assets/js/global.js:554 67 | msgid "Disconnected" 68 | msgstr "" 69 | 70 | #: ucp/assets/js/global.js:68 ucp/assets/js/global.js:76 71 | msgid "Enable" 72 | msgstr "" 73 | 74 | #: views/ucp_config.php:19 75 | msgid "Enable Phone" 76 | msgstr "" 77 | 78 | #: functions.inc/hook_core.php:62 79 | msgid "Enable WebRTC Old ARI Phone" 80 | msgstr "" 81 | 82 | #: views/ucp_config.php:38 83 | msgid "Enable the phone in UCP for this user" 84 | msgstr "" 85 | 86 | #: functions.inc/hook_core.php:62 87 | #, php-format 88 | msgid "" 89 | "Enables WebRTC for this %s in the Asterisk Recording Interface (ARI). Note: " 90 | "ARI is depreciated in favor of UCP" 91 | msgstr "" 92 | 93 | #: ucp/views/phone.php:34 94 | msgid "GHI" 95 | msgstr "" 96 | 97 | #: Webrtc.class.php:372 98 | msgid "HTTPS is not enabled for Asterisk" 99 | msgstr "HTTPS не ввімкнуто для Астеріску" 100 | 101 | #: ucp/views/phone.php:81 102 | msgid "Hold" 103 | msgstr "" 104 | 105 | #: ucp/assets/js/global.js:266 106 | #, perl-format 107 | msgid "Incoming call from %s" 108 | msgstr "Вхідний дзвінок від %s" 109 | 110 | #: views/ucp_config.php:29 111 | msgid "Inherit" 112 | msgstr "Успадкувати" 113 | 114 | #: ucp/views/phone.php:38 115 | msgid "JKL" 116 | msgstr "" 117 | 118 | #: ucp/views/phone.php:42 119 | msgid "MNO" 120 | msgstr "" 121 | 122 | #: views/ucp_config.php:26 123 | msgid "No" 124 | msgstr "Ні" 125 | 126 | #: ucp/assets/js/global.js:633 127 | msgid "Not supported in this browser" 128 | msgstr "" 129 | 130 | #: ucp/views/phone.php:88 131 | msgid "Note: An Active call will not be dropped when this window is closed" 132 | msgstr "" 133 | 134 | #: ucp/assets/js/global.js:640 135 | msgid "Only supported over HTTPS" 136 | msgstr "" 137 | 138 | #: ucp/views/phone.php:48 139 | msgid "PQRS" 140 | msgstr "" 141 | 142 | #: Webrtc.class.php:264 ucp/Webrtc.class.php:25 ucp/Webrtc.class.php:66 143 | msgid "Phone" 144 | msgstr "" 145 | 146 | #: ucp/views/phone.php:86 147 | msgid "Phone Status" 148 | msgstr "" 149 | 150 | #: ucp/assets/js/global.js:522 151 | msgid "Registered" 152 | msgstr "" 153 | 154 | #: ucp/assets/js/global.js:569 155 | msgid "Registering..." 156 | msgstr "" 157 | 158 | #: ucp/assets/js/global.js:540 159 | msgid "Registration Failed" 160 | msgstr "" 161 | 162 | #: Webrtc.class.php:281 163 | #, php-format 164 | msgid "" 165 | "Running an unsupported version of Asterisk. %s Detected Asterisk version: %s " 166 | msgstr "" 167 | 168 | #: ucp/views/settings.php:3 169 | msgid "Silence Ringer" 170 | msgstr "" 171 | 172 | #: ucp/views/phone.php:52 173 | msgid "TUV" 174 | msgstr "" 175 | 176 | #: functions.inc/hook_core.php:74 177 | msgid "The Certificate to use from Certificate Manager" 178 | msgstr "" 179 | 180 | #: webrtc.i18n.php:8 181 | msgid "The WebRTC Module allows an Administrator to enable a " 182 | msgstr "" 183 | 184 | #: functions.inc/hook_core.php:58 185 | msgid "" 186 | "The WebSockets Interface is running through PJSIP, PJSIP is not supported at " 187 | "this time. Please enable the chan_sip driver (along with pjsip or by itself) " 188 | "or alert the FreePBX Developers" 189 | msgstr "" 190 | "WebSockets Інтерфейс працює через PJSIP, PJSIP не підтримуються в даний час. " 191 | "Будь ласка, включите драйвер chan_sip (разом з pjsip або сам по собі) або " 192 | "попередьте розробників FreePBX" 193 | 194 | #: functions.inc/hook_core.php:79 195 | #, php-format 196 | msgid "" 197 | "To utilize WebRTC in ARI you must add at least one certificate %s through " 198 | "Certificate Manager" 199 | msgstr "" 200 | 201 | #: ucp/assets/js/global.js:532 ucp/assets/js/global.js:549 202 | msgid "Unregistered" 203 | msgstr "" 204 | 205 | #: functions.inc/hook_core.php:73 206 | msgid "Use Certificate" 207 | msgstr "Використовувати сертифікат" 208 | 209 | #: ucp/views/phone.php:56 210 | msgid "WXYZ" 211 | msgstr "" 212 | 213 | #: ucp/Webrtc.class.php:74 214 | #, fuzzy 215 | msgid "WebRTC" 216 | msgstr "WebRTC" 217 | 218 | #: webrtc.i18n.php:4 219 | msgid "WebRTC Phone" 220 | msgstr "WebRTC телефон" 221 | 222 | #: ucp/Webrtc.class.php:54 223 | msgid "Webrtc is not enabled for this extension." 224 | msgstr "" 225 | 226 | #: views/ucp_config.php:24 227 | msgid "Yes" 228 | msgstr "Так" 229 | 230 | #: Webrtc.class.php:268 231 | #, php-format 232 | msgid "You have no default certificates setup in %s" 233 | msgstr "" 234 | 235 | #~ msgid "Unsupported Version of Asterisk, You need at least %s you have %s" 236 | #~ msgstr "" 237 | #~ "Непідтримувана версія Астеріск, Вам необхідно мінімум %s, встановлено %s" 238 | 239 | #~ msgid "Certificate WebRTC Phone should use for this user." 240 | #~ msgstr "" 241 | #~ "Сертифікат WebRTC телефону повинен використовувати для цього користувача." 242 | 243 | #~ msgid "Enable WebRTC Phone" 244 | #~ msgstr "Активувати WebRTC телефон" 245 | 246 | #~ msgid "Enable the WebRTC Phone in UCP for this user" 247 | #~ msgstr "Активує WebRTC телефон для цього користувача в UCP" 248 | 249 | #~ msgid "New Web Phone Call" 250 | #~ msgstr "Новий Веб-дзвінок" 251 | 252 | #~ msgid "WebRTC Certificate" 253 | #~ msgstr "Сертифікат WebRTC" 254 | 255 | #~ msgid "Originate" 256 | #~ msgstr "Ініціювати" 257 | 258 | #~ msgid "Originate Call" 259 | #~ msgstr "Вихідний виклик" 260 | 261 | #~ msgid "" 262 | #~ "Unable to start call. Please allow the WebRTC session in your browser and " 263 | #~ "refresh" 264 | #~ msgstr "" 265 | #~ "Неможливо розпочати виклик. Будь ласка, дозвольте сеанс WebRTC в вашому " 266 | #~ "браузері і оновить сторінку" 267 | 268 | #~ msgid "" 269 | #~ "The STUN Server address is blank. In many cases this can cause issues. " 270 | #~ "Please define a valid server in the Asterisk SIP Settings module" 271 | #~ msgstr "" 272 | #~ "Адреса STUN сервера порожня. У багатьох випадках це може викликати " 273 | #~ "проблеми. Будь ласка, визначте допустимий сервер в модулі налаштування " 274 | #~ "SIP Астеріску" 275 | -------------------------------------------------------------------------------- /i18n/vi/LC_MESSAGES/webrtc.mo: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/FreePBX/webrtc/92ce6acc8eced21055a5b63267b764a8609e4cae/i18n/vi/LC_MESSAGES/webrtc.mo -------------------------------------------------------------------------------- /i18n/vi/LC_MESSAGES/webrtc.po: -------------------------------------------------------------------------------- 1 | # This file is part of FreePBX. 2 | # 3 | # For licensing information, please see the file named LICENSE located in the module directory 4 | # 5 | # FreePBX language template for webrtc 6 | # Copyright (C) 2008-2017 Sangoma Technologies, Inc. 7 | # 8 | msgid "" 9 | msgstr "" 10 | "Project-Id-Version: PACKAGE VERSION\n" 11 | "Report-Msgid-Bugs-To: \n" 12 | "POT-Creation-Date: 2024-12-03 06:07-0500\n" 13 | "PO-Revision-Date: 2017-07-18 10:23+0200\n" 14 | "Last-Translator: PETER \n" 15 | "Language-Team: Vietnamese \n" 17 | "Language: vi\n" 18 | "MIME-Version: 1.0\n" 19 | "Content-Type: text/plain; charset=UTF-8\n" 20 | "Content-Transfer-Encoding: 8bit\n" 21 | "Plural-Forms: nplurals=1; plural=0;\n" 22 | "X-Generator: Weblate 2.4\n" 23 | 24 | #: ucp/views/phone.php:24 25 | msgid "ABC" 26 | msgstr "" 27 | 28 | #: ucp/views/phone.php:78 29 | msgid "Call" 30 | msgstr "" 31 | 32 | #: Webrtc.class.php:268 33 | msgid "Certificate Manager" 34 | msgstr "" 35 | 36 | #: ucp/assets/js/global.js:267 37 | msgid "Click this window to answer or close this window to ignore" 38 | msgstr "Kích vào cửa sổ này để trả lời hoặc đóng cửa sổ để bỏ qua" 39 | 40 | #: ucp/assets/js/global.js:542 41 | msgid "Connect Phone" 42 | msgstr "Kết nối điện thoại" 43 | 44 | #: ucp/assets/js/global.js:563 45 | msgid "Connecting to socket..." 46 | msgstr "" 47 | 48 | #: webrtc.i18n.php:6 49 | msgid "Connectivity" 50 | msgstr "Khả năng kết nối" 51 | 52 | #: ucp/views/phone.php:28 53 | msgid "DEF" 54 | msgstr "" 55 | 56 | #: ucp/assets/js/global.js:69 ucp/assets/js/global.js:77 57 | msgid "Disable" 58 | msgstr "" 59 | 60 | #: ucp/views/settings.php:7 61 | msgid "Disconnect Phone" 62 | msgstr "Ngắt kết nối điện thoại" 63 | 64 | #: ucp/assets/js/global.js:554 65 | msgid "Disconnected" 66 | msgstr "" 67 | 68 | #: ucp/assets/js/global.js:68 ucp/assets/js/global.js:76 69 | msgid "Enable" 70 | msgstr "" 71 | 72 | #: views/ucp_config.php:19 73 | msgid "Enable Phone" 74 | msgstr "" 75 | 76 | #: functions.inc/hook_core.php:62 77 | msgid "Enable WebRTC Old ARI Phone" 78 | msgstr "Kích hoạt WebRTC Old ARI Phone" 79 | 80 | #: views/ucp_config.php:38 81 | msgid "Enable the phone in UCP for this user" 82 | msgstr "" 83 | 84 | #: functions.inc/hook_core.php:62 85 | #, php-format 86 | msgid "" 87 | "Enables WebRTC for this %s in the Asterisk Recording Interface (ARI). Note: " 88 | "ARI is depreciated in favor of UCP" 89 | msgstr "" 90 | "Kích hoạt WebRTC cho %s này tại giao diện ghi âm Asterisk ( ẢI). Lưu ý: ARI " 91 | "bị sụt giảm để có lợi cho Bảng điều khiển" 92 | 93 | #: ucp/views/phone.php:34 94 | msgid "GHI" 95 | msgstr "" 96 | 97 | #: Webrtc.class.php:372 98 | msgid "HTTPS is not enabled for Asterisk" 99 | msgstr "HTTPS không được kích hoạt cho Asterisk" 100 | 101 | #: ucp/views/phone.php:81 102 | msgid "Hold" 103 | msgstr "" 104 | 105 | #: ucp/assets/js/global.js:266 106 | #, perl-format 107 | msgid "Incoming call from %s" 108 | msgstr "Cuộc gọi đến từ %s" 109 | 110 | #: views/ucp_config.php:29 111 | msgid "Inherit" 112 | msgstr "Kế thừa" 113 | 114 | #: ucp/views/phone.php:38 115 | msgid "JKL" 116 | msgstr "" 117 | 118 | #: ucp/views/phone.php:42 119 | msgid "MNO" 120 | msgstr "" 121 | 122 | #: views/ucp_config.php:26 123 | msgid "No" 124 | msgstr "Không" 125 | 126 | #: ucp/assets/js/global.js:633 127 | msgid "Not supported in this browser" 128 | msgstr "" 129 | 130 | #: ucp/views/phone.php:88 131 | msgid "Note: An Active call will not be dropped when this window is closed" 132 | msgstr "" 133 | 134 | #: ucp/assets/js/global.js:640 135 | msgid "Only supported over HTTPS" 136 | msgstr "" 137 | 138 | #: ucp/views/phone.php:48 139 | msgid "PQRS" 140 | msgstr "" 141 | 142 | #: Webrtc.class.php:264 ucp/Webrtc.class.php:25 ucp/Webrtc.class.php:66 143 | msgid "Phone" 144 | msgstr "" 145 | 146 | #: ucp/views/phone.php:86 147 | msgid "Phone Status" 148 | msgstr "" 149 | 150 | #: ucp/assets/js/global.js:522 151 | msgid "Registered" 152 | msgstr "" 153 | 154 | #: ucp/assets/js/global.js:569 155 | msgid "Registering..." 156 | msgstr "" 157 | 158 | #: ucp/assets/js/global.js:540 159 | msgid "Registration Failed" 160 | msgstr "" 161 | 162 | #: Webrtc.class.php:281 163 | #, php-format 164 | msgid "" 165 | "Running an unsupported version of Asterisk. %s Detected Asterisk version: %s " 166 | msgstr "" 167 | 168 | #: ucp/views/settings.php:3 169 | msgid "Silence Ringer" 170 | msgstr "Để chuông chế độ im lặng" 171 | 172 | #: ucp/views/phone.php:52 173 | msgid "TUV" 174 | msgstr "" 175 | 176 | #: functions.inc/hook_core.php:74 177 | msgid "The Certificate to use from Certificate Manager" 178 | msgstr "Chứng thực sử dụng từ trình quản lý chứng thực" 179 | 180 | #: webrtc.i18n.php:8 181 | msgid "The WebRTC Module allows an Administrator to enable a " 182 | msgstr "Mô đun WebRTC cho phép một quản trị viên kích hoạt một " 183 | 184 | #: functions.inc/hook_core.php:58 185 | msgid "" 186 | "The WebSockets Interface is running through PJSIP, PJSIP is not supported at " 187 | "this time. Please enable the chan_sip driver (along with pjsip or by itself) " 188 | "or alert the FreePBX Developers" 189 | msgstr "" 190 | "Giao diện WebSocket đang chạy qua PJSIP, PJSIP không được hỗ trợ tại thời " 191 | "điểm này. Vui lòng kích hoạt driver chan_sip ( kèm với pjsip hoặc chính nó) " 192 | "hoặc cảnh báo những nhà phát triển FreePBX" 193 | 194 | #: functions.inc/hook_core.php:79 195 | #, php-format 196 | msgid "" 197 | "To utilize WebRTC in ARI you must add at least one certificate %s through " 198 | "Certificate Manager" 199 | msgstr "" 200 | "Để sử dụng WebRTC trong ARI, bạn phải thêm ít nhất một chứng thực %s qua " 201 | "Trình Quản lý Chứng thực" 202 | 203 | #: ucp/assets/js/global.js:532 ucp/assets/js/global.js:549 204 | msgid "Unregistered" 205 | msgstr "" 206 | 207 | #: functions.inc/hook_core.php:73 208 | msgid "Use Certificate" 209 | msgstr "Sử dụng chứng thực" 210 | 211 | #: ucp/views/phone.php:56 212 | msgid "WXYZ" 213 | msgstr "" 214 | 215 | #: ucp/Webrtc.class.php:74 216 | msgid "WebRTC" 217 | msgstr "WebRTC (web API được phát triển bởi World Wide Web Consortium)" 218 | 219 | #: webrtc.i18n.php:4 220 | msgid "WebRTC Phone" 221 | msgstr "Điện thoại WebRTC" 222 | 223 | #: ucp/Webrtc.class.php:54 224 | msgid "Webrtc is not enabled for this extension." 225 | msgstr "" 226 | 227 | #: views/ucp_config.php:24 228 | msgid "Yes" 229 | msgstr "Có" 230 | 231 | #: Webrtc.class.php:268 232 | #, php-format 233 | msgid "You have no default certificates setup in %s" 234 | msgstr "" 235 | 236 | #~ msgid "Unsupported Version of Asterisk, You need at least %s you have %s" 237 | #~ msgstr "" 238 | #~ "Phiên bản không được hỗ trợ của Asterisk. Bạn cần ít nhất %s để bạn có %s" 239 | 240 | #~ msgid "" 241 | #~ ")),-1===f?(this.headers[b].splice(c,1),void this.logger.warn('error " 242 | #~ "parsing " 243 | #~ msgstr "" 244 | #~ ")),-1===f?(this.headers[b].splice(c,1),void this.logger.warn('error " 245 | #~ "parsing " 246 | 247 | #~ msgid "Certificate WebRTC Phone should use for this user." 248 | #~ msgstr "Giấy chứng nhận điện thoại WebRTC nên sử dụng cho người dùng này." 249 | 250 | #~ msgid "Enable WebRTC Phone" 251 | #~ msgstr "Kích hoạt WebRTC Phone" 252 | 253 | #~ msgid "Enable the WebRTC Phone in UCP for this user" 254 | #~ msgstr "" 255 | #~ "Kích hoạt WebRTC Phone tại bảng điều khiển người dùng cho người dùng này" 256 | 257 | #~ msgid "New Web Phone Call" 258 | #~ msgstr "Cuộc gọi Web Phone mới" 259 | 260 | #~ msgid "Originate" 261 | #~ msgstr "Khởi tạo" 262 | 263 | #~ msgid "Originate Call" 264 | #~ msgstr "Khởi tạo cuộc gọi" 265 | 266 | #~ msgid "" 267 | #~ "Unable to start call. Please allow the WebRTC session in your browser and " 268 | #~ "refresh" 269 | #~ msgstr "" 270 | #~ "Không thể bắt đầu cuộc gọi. Vui lòng cho phép phiên WebRTC trên trình " 271 | #~ "duyệt của bạn và làm mới nó" 272 | 273 | #~ msgid "WebRTC Certificate" 274 | #~ msgstr "Chứng thực WebRTC" 275 | 276 | #~ msgid "You have no certificates setup in Certificate Manager" 277 | #~ msgstr "Bạn không có chứng thực được cài đặt trong trình quản lý chứng thực" 278 | -------------------------------------------------------------------------------- /i18n/webrtc.pot: -------------------------------------------------------------------------------- 1 | # This file is part of FreePBX. 2 | # 3 | # For licensing information, please see the file named LICENSE located in the module directory 4 | # 5 | # FreePBX language template for webrtc 6 | # Copyright (C) 2008-2024 Sangoma Technologies, Inc. 7 | # 8 | msgid "" 9 | msgstr "" 10 | "Project-Id-Version: PACKAGE VERSION\n" 11 | "Report-Msgid-Bugs-To: \n" 12 | "POT-Creation-Date: 2024-12-04 02:17-0500\n" 13 | "PO-Revision-Date: YEAR-MO-DA HO:MI+ZONE\n" 14 | "Last-Translator: FULL NAME \n" 15 | "Language-Team: LANGUAGE \n" 16 | "Language: \n" 17 | "MIME-Version: 1.0\n" 18 | "Content-Type: text/plain; charset=utf-8\n" 19 | "Content-Transfer-Encoding: 8bit\n" 20 | 21 | #: ucp/views/phone.php:24 22 | msgid "ABC" 23 | msgstr "" 24 | 25 | #: ucp/views/phone.php:78 26 | msgid "Call" 27 | msgstr "" 28 | 29 | #: Webrtc.class.php:268 30 | msgid "Certificate Manager" 31 | msgstr "" 32 | 33 | #: ucp/assets/js/global.js:267 34 | msgid "Click this window to answer or close this window to ignore" 35 | msgstr "" 36 | 37 | #: ucp/assets/js/global.js:542 38 | msgid "Connect Phone" 39 | msgstr "" 40 | 41 | #: ucp/assets/js/global.js:563 42 | msgid "Connecting to socket..." 43 | msgstr "" 44 | 45 | #: webrtc.i18n.php:6 46 | msgid "Connectivity" 47 | msgstr "" 48 | 49 | #: ucp/views/phone.php:28 50 | msgid "DEF" 51 | msgstr "" 52 | 53 | #: ucp/assets/js/global.js:69 54 | #: ucp/assets/js/global.js:77 55 | msgid "Disable" 56 | msgstr "" 57 | 58 | #: ucp/views/settings.php:7 59 | msgid "Disconnect Phone" 60 | msgstr "" 61 | 62 | #: ucp/assets/js/global.js:554 63 | msgid "Disconnected" 64 | msgstr "" 65 | 66 | #: ucp/assets/js/global.js:68 67 | #: ucp/assets/js/global.js:76 68 | msgid "Enable" 69 | msgstr "" 70 | 71 | #: views/ucp_config.php:19 72 | msgid "Enable Phone" 73 | msgstr "" 74 | 75 | #: functions.inc/hook_core.php:62 76 | msgid "Enable WebRTC Old ARI Phone" 77 | msgstr "" 78 | 79 | #: views/ucp_config.php:38 80 | msgid "Enable the phone in UCP for this user" 81 | msgstr "" 82 | 83 | #: functions.inc/hook_core.php:62 84 | #, php-format 85 | msgid "" 86 | "Enables WebRTC for this %s in the Asterisk Recording Interface (ARI). Note: " 87 | "ARI is depreciated in favor of UCP" 88 | msgstr "" 89 | 90 | #: ucp/views/phone.php:34 91 | msgid "GHI" 92 | msgstr "" 93 | 94 | #: Webrtc.class.php:372 95 | msgid "HTTPS is not enabled for Asterisk" 96 | msgstr "" 97 | 98 | #: ucp/views/phone.php:81 99 | msgid "Hold" 100 | msgstr "" 101 | 102 | #: ucp/assets/js/global.js:266 103 | #, perl-format 104 | msgid "Incoming call from %s" 105 | msgstr "" 106 | 107 | #: views/ucp_config.php:29 108 | msgid "Inherit" 109 | msgstr "" 110 | 111 | #: ucp/views/phone.php:38 112 | msgid "JKL" 113 | msgstr "" 114 | 115 | #: ucp/views/phone.php:42 116 | msgid "MNO" 117 | msgstr "" 118 | 119 | #: views/ucp_config.php:26 120 | msgid "No" 121 | msgstr "" 122 | 123 | #: ucp/assets/js/global.js:633 124 | msgid "Not supported in this browser" 125 | msgstr "" 126 | 127 | #: ucp/views/phone.php:88 128 | msgid "Note: An Active call will not be dropped when this window is closed" 129 | msgstr "" 130 | 131 | #: ucp/assets/js/global.js:640 132 | msgid "Only supported over HTTPS" 133 | msgstr "" 134 | 135 | #: ucp/views/phone.php:48 136 | msgid "PQRS" 137 | msgstr "" 138 | 139 | #: Webrtc.class.php:264 140 | #: ucp/Webrtc.class.php:25 141 | #: ucp/Webrtc.class.php:66 142 | msgid "Phone" 143 | msgstr "" 144 | 145 | #: ucp/views/phone.php:86 146 | msgid "Phone Status" 147 | msgstr "" 148 | 149 | #: ucp/assets/js/global.js:522 150 | msgid "Registered" 151 | msgstr "" 152 | 153 | #: ucp/assets/js/global.js:569 154 | msgid "Registering..." 155 | msgstr "" 156 | 157 | #: ucp/assets/js/global.js:540 158 | msgid "Registration Failed" 159 | msgstr "" 160 | 161 | #: Webrtc.class.php:281 162 | #, php-format 163 | msgid "" 164 | "Running an unsupported version of Asterisk. %s Detected Asterisk version: %s " 165 | msgstr "" 166 | 167 | #: ucp/views/settings.php:3 168 | msgid "Silence Ringer" 169 | msgstr "" 170 | 171 | #: ucp/views/phone.php:52 172 | msgid "TUV" 173 | msgstr "" 174 | 175 | #: functions.inc/hook_core.php:74 176 | msgid "The Certificate to use from Certificate Manager" 177 | msgstr "" 178 | 179 | #: webrtc.i18n.php:8 180 | msgid "The WebRTC Module allows an Administrator to enable a " 181 | msgstr "" 182 | 183 | #: functions.inc/hook_core.php:58 184 | msgid "" 185 | "The WebSockets Interface is running through PJSIP, PJSIP is not supported at " 186 | "this time. Please enable the chan_sip driver (along with pjsip or by itself) " 187 | "or alert the FreePBX Developers" 188 | msgstr "" 189 | 190 | #: functions.inc/hook_core.php:79 191 | #, php-format 192 | msgid "" 193 | "To utilize WebRTC in ARI you must add at least one certificate %s through " 194 | "Certificate Manager" 195 | msgstr "" 196 | 197 | #: ucp/assets/js/global.js:532 198 | #: ucp/assets/js/global.js:549 199 | msgid "Unregistered" 200 | msgstr "" 201 | 202 | #: functions.inc/hook_core.php:73 203 | msgid "Use Certificate" 204 | msgstr "" 205 | 206 | #: ucp/views/phone.php:56 207 | msgid "WXYZ" 208 | msgstr "" 209 | 210 | #: ucp/Webrtc.class.php:74 211 | msgid "WebRTC" 212 | msgstr "" 213 | 214 | #: webrtc.i18n.php:4 215 | msgid "WebRTC Phone" 216 | msgstr "" 217 | 218 | #: ucp/Webrtc.class.php:54 219 | msgid "Webrtc is not enabled for this extension." 220 | msgstr "" 221 | 222 | #: views/ucp_config.php:24 223 | msgid "Yes" 224 | msgstr "" 225 | 226 | #: Webrtc.class.php:268 227 | #, php-format 228 | msgid "You have no default certificates setup in %s" 229 | msgstr "" 230 | -------------------------------------------------------------------------------- /i18n/zh_CN/LC_MESSAGES/webrtc.mo: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/FreePBX/webrtc/92ce6acc8eced21055a5b63267b764a8609e4cae/i18n/zh_CN/LC_MESSAGES/webrtc.mo -------------------------------------------------------------------------------- /i18n/zh_CN/LC_MESSAGES/webrtc.po: -------------------------------------------------------------------------------- 1 | # This file is part of FreePBX. 2 | # 3 | # For licensing information, please see the file named LICENSE located in the module directory 4 | # 5 | # FreePBX language template for webrtc 6 | # Copyright (C) 2008-2015 Sangoma, Inc. 7 | # 8 | msgid "" 9 | msgstr "" 10 | "Project-Id-Version: FreePBX 2.5 Chinese Translation\n" 11 | "Report-Msgid-Bugs-To: \n" 12 | "POT-Creation-Date: 2024-12-03 06:07-0500\n" 13 | "PO-Revision-Date: 2018-03-14 14:52+0200\n" 14 | "Last-Translator: james \n" 15 | "Language-Team: Simplified Chinese \n" 17 | "Language: zh_CN\n" 18 | "MIME-Version: 1.0\n" 19 | "Content-Type: text/plain; charset=UTF-8\n" 20 | "Content-Transfer-Encoding: 8bit\n" 21 | "Plural-Forms: nplurals=1; plural=0;\n" 22 | "X-Generator: Weblate 2.4\n" 23 | 24 | #: ucp/views/phone.php:24 25 | msgid "ABC" 26 | msgstr "" 27 | 28 | #: ucp/views/phone.php:78 29 | msgid "Call" 30 | msgstr "" 31 | 32 | #: Webrtc.class.php:268 33 | msgid "Certificate Manager" 34 | msgstr "" 35 | 36 | #: ucp/assets/js/global.js:267 37 | msgid "Click this window to answer or close this window to ignore" 38 | msgstr "点击此窗口应答或忽略关闭窗口" 39 | 40 | #: ucp/assets/js/global.js:542 41 | msgid "Connect Phone" 42 | msgstr "连接电话终端" 43 | 44 | #: ucp/assets/js/global.js:563 45 | msgid "Connecting to socket..." 46 | msgstr "" 47 | 48 | #: webrtc.i18n.php:6 49 | msgid "Connectivity" 50 | msgstr "通信接口连接" 51 | 52 | #: ucp/views/phone.php:28 53 | msgid "DEF" 54 | msgstr "" 55 | 56 | #: ucp/assets/js/global.js:69 ucp/assets/js/global.js:77 57 | msgid "Disable" 58 | msgstr "" 59 | 60 | #: ucp/views/settings.php:7 61 | msgid "Disconnect Phone" 62 | msgstr "对电话挂机" 63 | 64 | #: ucp/assets/js/global.js:554 65 | msgid "Disconnected" 66 | msgstr "" 67 | 68 | #: ucp/assets/js/global.js:68 ucp/assets/js/global.js:76 69 | msgid "Enable" 70 | msgstr "" 71 | 72 | #: views/ucp_config.php:19 73 | msgid "Enable Phone" 74 | msgstr "" 75 | 76 | #: functions.inc/hook_core.php:62 77 | msgid "Enable WebRTC Old ARI Phone" 78 | msgstr "开启WebRTC 以前的 ARI 电话" 79 | 80 | #: views/ucp_config.php:38 81 | msgid "Enable the phone in UCP for this user" 82 | msgstr "" 83 | 84 | #: functions.inc/hook_core.php:62 85 | #, php-format 86 | msgid "" 87 | "Enables WebRTC for this %s in the Asterisk Recording Interface (ARI). Note: " 88 | "ARI is depreciated in favor of UCP" 89 | msgstr "在ARI中对%s 开启WebRTC。注意：UCP中已经停止使用ARI" 90 | 91 | #: ucp/views/phone.php:34 92 | msgid "GHI" 93 | msgstr "" 94 | 95 | #: Webrtc.class.php:372 96 | msgid "HTTPS is not enabled for Asterisk" 97 | msgstr "没有对Asterisk开启HTTPS" 98 | 99 | #: ucp/views/phone.php:81 100 | msgid "Hold" 101 | msgstr "" 102 | 103 | #: ucp/assets/js/global.js:266 104 | #, perl-format 105 | msgid "Incoming call from %s" 106 | msgstr "从 %s 进入的呼叫" 107 | 108 | #: views/ucp_config.php:29 109 | msgid "Inherit" 110 | msgstr "继承" 111 | 112 | #: ucp/views/phone.php:38 113 | msgid "JKL" 114 | msgstr "" 115 | 116 | #: ucp/views/phone.php:42 117 | msgid "MNO" 118 | msgstr "" 119 | 120 | #: views/ucp_config.php:26 121 | msgid "No" 122 | msgstr "否" 123 | 124 | #: ucp/assets/js/global.js:633 125 | msgid "Not supported in this browser" 126 | msgstr "" 127 | 128 | #: ucp/views/phone.php:88 129 | msgid "Note: An Active call will not be dropped when this window is closed" 130 | msgstr "" 131 | 132 | #: ucp/assets/js/global.js:640 133 | msgid "Only supported over HTTPS" 134 | msgstr "" 135 | 136 | #: ucp/views/phone.php:48 137 | msgid "PQRS" 138 | msgstr "" 139 | 140 | #: Webrtc.class.php:264 ucp/Webrtc.class.php:25 ucp/Webrtc.class.php:66 141 | msgid "Phone" 142 | msgstr "" 143 | 144 | #: ucp/views/phone.php:86 145 | msgid "Phone Status" 146 | msgstr "" 147 | 148 | #: ucp/assets/js/global.js:522 149 | msgid "Registered" 150 | msgstr "" 151 | 152 | #: ucp/assets/js/global.js:569 153 | msgid "Registering..." 154 | msgstr "" 155 | 156 | #: ucp/assets/js/global.js:540 157 | msgid "Registration Failed" 158 | msgstr "" 159 | 160 | #: Webrtc.class.php:281 161 | #, php-format 162 | msgid "" 163 | "Running an unsupported version of Asterisk. %s Detected Asterisk version: %s " 164 | msgstr "" 165 | 166 | #: ucp/views/settings.php:3 167 | msgid "Silence Ringer" 168 | msgstr "" 169 | 170 | #: ucp/views/phone.php:52 171 | msgid "TUV" 172 | msgstr "" 173 | 174 | #: functions.inc/hook_core.php:74 175 | msgid "The Certificate to use from Certificate Manager" 176 | msgstr "使用的证书来自于证书管理员" 177 | 178 | #: webrtc.i18n.php:8 179 | msgid "The WebRTC Module allows an Administrator to enable a " 180 | msgstr "" 181 | 182 | #: functions.inc/hook_core.php:58 183 | msgid "" 184 | "The WebSockets Interface is running through PJSIP, PJSIP is not supported at " 185 | "this time. Please enable the chan_sip driver (along with pjsip or by itself) " 186 | "or alert the FreePBX Developers" 187 | msgstr "" 188 | "WebSockets 接口通过PJSIP在运行，现在PJSIP不支持。请开启chan_sip 驱动或者提醒" 189 | "FreePBX 开发人员" 190 | 191 | #: functions.inc/hook_core.php:79 192 | #, php-format 193 | msgid "" 194 | "To utilize WebRTC in ARI you must add at least one certificate %s through " 195 | "Certificate Manager" 196 | msgstr "为了使用webRTC，必须通过证书管理员添加证书 %s" 197 | 198 | #: ucp/assets/js/global.js:532 ucp/assets/js/global.js:549 199 | msgid "Unregistered" 200 | msgstr "" 201 | 202 | #: functions.inc/hook_core.php:73 203 | msgid "Use Certificate" 204 | msgstr "使用证书" 205 | 206 | #: ucp/views/phone.php:56 207 | msgid "WXYZ" 208 | msgstr "" 209 | 210 | #: ucp/Webrtc.class.php:74 211 | msgid "WebRTC" 212 | msgstr "" 213 | 214 | #: webrtc.i18n.php:4 215 | msgid "WebRTC Phone" 216 | msgstr "WebRTC 电话" 217 | 218 | #: ucp/Webrtc.class.php:54 219 | msgid "Webrtc is not enabled for this extension." 220 | msgstr "" 221 | 222 | #: views/ucp_config.php:24 223 | msgid "Yes" 224 | msgstr "是" 225 | 226 | #: Webrtc.class.php:268 227 | #, php-format 228 | msgid "You have no default certificates setup in %s" 229 | msgstr "" 230 | 231 | #~ msgid "Unsupported Version of Asterisk, You need at least %s you have %s" 232 | #~ msgstr "不能支持Asterisk的版本，用户至少需要%s, 但是现在的版本是%s" 233 | 234 | #~ msgid "Certificate WebRTC Phone should use for this user." 235 | #~ msgstr "此用户使用带证书的WebRTC 电话。" 236 | 237 | #~ msgid "Enable WebRTC Phone" 238 | #~ msgstr "开启 WebRTC 电话" 239 | 240 | #~ msgid "Enable the WebRTC Phone in UCP for this user" 241 | #~ msgstr "对此用户开启WebRTC电话" 242 | 243 | #~ msgid "New Web Phone Call" 244 | #~ msgstr "新的页面电话呼叫" 245 | 246 | #~ msgid "Originate" 247 | #~ msgstr "发起" 248 | 249 | #~ msgid "Originate Call" 250 | #~ msgstr "发起呼叫" 251 | 252 | #~ msgid "" 253 | #~ "Unable to start call. Please allow the WebRTC session in your browser and " 254 | #~ "refresh" 255 | #~ msgstr "不能发起呼叫，请在浏览器设置支持WebRTC，或者重新刷新浏览器" 256 | 257 | #~ msgid "WebRTC Certificate" 258 | #~ msgstr "WebRTC证书" 259 | 260 | #~ msgid "You have no certificates setup in Certificate Manager" 261 | #~ msgstr "在证书管理员中没有证书设置" 262 | 263 | #~ msgid "" 264 | #~ "The STUN Server address is blank. In many cases this can cause issues. " 265 | #~ "Please define a valid server in the Asterisk SIP Settings module" 266 | #~ msgstr "" 267 | #~ "无STUN 服务器地址。很多情况下，就是因为这个问题导致。请在Asterisk SIP 设置" 268 | #~ "模块中定义有效的服务器地址" 269 | 270 | #~ msgid "Unsupported Version of Asterisk, You need at least %s you have &s" 271 | #~ msgstr "不能支持Asterisk，至少需要版本%s，现在是&s" 272 | 273 | #~ msgid "" 274 | #~ "Whether or not to enable the WebRTC Phone for this linked. Additionally " 275 | #~ "you must select a valid certificate to use." 276 | #~ msgstr "无论是否对这个链接开启WebRTC，用户还要添加有效的证书。" 277 | 278 | #~ msgid "Which certificate to use for the WebRTC Phone in UCP" 279 | #~ msgstr "WebRTC中使用了哪个证书" 280 | 281 | #~ msgid "" 282 | #~ "You have no certificates setup in Certificate Manager" 284 | #~ msgstr "" 285 | #~ "没有在证书管理员 中设置证书" 286 | 287 | #~ msgid "Allow Originating Calls" 288 | #~ msgstr "允许发起呼叫" 289 | 290 | #~ msgid "Allow calls to be originated from UCP" 291 | #~ msgstr "允许通过UCP发起呼叫" 292 | 293 | #~ msgid "Nothing Entered" 294 | #~ msgstr "没有任何输入" 295 | 296 | #~ msgid "" 297 | #~ "The WebRTC Module allows an Administrator to enable a \"WebRTC phone\" " 298 | #~ "that can be attached to a user\\'s\n" 299 | #~ "\t\textension which they can connect to through FreePBX User Control " 300 | #~ "Panel, this WebRTC phone will then\n" 301 | #~ "\t\treceive phone calls at the same time as the users extension using " 302 | #~ "user and device mode behind the scenes.\n" 303 | #~ "\t\tIf you have User and Device Mode enabled any extension you enable the " 304 | #~ "WebRTC Phone a duplicate extension of 99XXXX\n" 305 | #~ "\t\twill be created (where XXXX is the original extension number), when " 306 | #~ "the user then views the web interface of the WebRTC\n" 307 | #~ "\t\tphone they will be connected to device 99XXXX which will receive " 308 | #~ "calls from the original extension" 309 | #~ msgstr "" 310 | #~ "WebRTC 模块支持管理员开启 \"WebRTC 电话\" ，这个电话可以添加到用户\\'s↵\n" 311 | #~ "→→通过FreePBX用户面板可以添加到分机中，这个 WebRTC 电话将↵\n" 312 | #~ "→→可以同时接听和设备分机一样的电话。↵\n" 313 | #~ "→→如果用户已经开启了用户和设备模式，用户分机将覆盖99XXXX的分机，XXX是已经" 314 | #~ "创建的分机↵\n" 315 | #~ "→→用户可以通过WebRTC 界面看到电话状态↵\n" 316 | #~ "→→从原来的分机中可以接听设备99XXXX 电话" 317 | -------------------------------------------------------------------------------- /module.xml: -------------------------------------------------------------------------------- 1 | 2 | webrtc 3 | WebRTC Phone 4 | 17.0.3.1 5 | Sangoma Technologies Corporation 6 | GPLv3+ 7 | http://www.gnu.org/licenses/gpl-3.0.txt 8 | standard 9 | 10 | The WebRTC Module allows an Administrator to enable a "WebRTC phone" that can be attached to a user's 11 | extension which they can connect to through FreePBX User Control Panel, this WebRTC phone will then 12 | receive phone calls at the same time as the users extension using user and device mode behind the scenes. 13 | If you have User and Device Mode enabled any extension you enable the WebRTC Phone a duplicate extension of 99XXXX 14 | will be created (where XXXX is the original extension number), when the user then views the web interface of the WebRTC 15 | phone they will be connected to device 99XXXX which will receive calls from the original extension 16 | 17 | https://sangomakb.atlassian.net/wiki/spaces/PG/pages/38634428/WebRTC+Phone-UCP 18 | 19 | *17.0.3.1* Fixing FREEI-1507 Webrtc call not showing callerid issue 20 | *17.0.3* Fixing FREEI-1408 UCP Web Phone cannot Hangup call before it is answered issue. 21 | *17.0.2* 17.0.2 release 22 | 23 | 24 | Connectivity 25 | 26 | asterisk 11.5 27 | certman ge 17.0.1 28 | core ge 17.0.1 29 | ucp ge 17.0.1 30 | 17.0 31 | 32 | 33 | 34 | ucpConfigPage 35 | ucpAddUser 36 | ucpUpdateUser 37 | ucpDelUser 38 | ucpAddGroup 39 | ucpUpdateGroup 40 | ucpDelGroup 41 | 42 | 43 | dashboardIgnoreExt 44 | 45 | 46 | delUser 47 | updatefromcore 48 | 49 | 50 | 51 | 52 | 53 | 54 | 55 | 56 | 57 | 58 | 59 | 60 | 61 | 62 | 63 | 64 |
65 | 66 | 67 | 17.0 68 | 69 | 70 | -------------------------------------------------------------------------------- /ucp/Webrtc.class.php: -------------------------------------------------------------------------------- 1 | Modules = $Modules; 16 | $this->webrtc = $this->UCP->FreePBX->Webrtc; 17 | $this->ext = $this->Modules->getDefaultDevice(); 18 | $this->astman = $this->UCP->FreePBX->astman; 19 | $this->user = $this->UCP->User->getUser(); 20 | } 21 | 22 | public function getSimpleWidgetList() { 23 | $responseData = array( 24 | "rawname" => "webrtc", 25 | "display" => _("Phone"), 26 | "icon" => "fa fa-phone", 27 | "list" => [] 28 | ); 29 | $errors = $this->validate(); 30 | if ($errors['hasError']) { 31 | return array_merge($responseData, $errors); 32 | } 33 | 34 | $widgets['phone'] = [ 35 | "display" => "Phone", 36 | "hasSettings" => true 37 | ]; 38 | 39 | $responseData['list'] = $widgets; 40 | return $responseData; 41 | } 42 | 43 | /** 44 | * validate against rules 45 | */ 46 | private function validate() { 47 | $data = array( 48 | 'hasError' => false, 49 | 'errorMessages' => [] 50 | ); 51 | 52 | if (!$this->webrtc->checkEnabled($this->ext)) { 53 | $data['hasError'] = true; 54 | $data['errorMessages'][] = _('Webrtc is not enabled for this extension.'); 55 | } 56 | 57 | return $data; 58 | } 59 | 60 | public function getSimpleWidgetDisplay($id) { 61 | $errors = $this->validate(); 62 | if ($errors['hasError']) { 63 | return $errors; 64 | } 65 | 66 | return ['title' => _("Phone"), 'html' => load_view(__DIR__."/views/phone.php",[])]; 67 | } 68 | 69 | public function getSimpleWidgetSettingsDisplay($id) { 70 | if(!$this->webrtc->checkEnabled($this->ext)) { 71 | return []; 72 | } 73 | $displayvars = []; 74 | $display = ['title' => _("WebRTC"), 'html' => $this->load_view(__DIR__.'/views/settings.php',$displayvars)]; 75 | 76 | return $display; 77 | } 78 | 79 | 80 | /** 81 | * Determine what commands are allowed 82 | * 83 | * Used by Ajax Class to determine what commands are allowed by this class 84 | * 85 | * @param string $command The command something is trying to perform 86 | * @param string $settings The Settings being passed through $_POST or $_PUT 87 | * @return bool True if pass 88 | */ 89 | function ajaxRequest($command, $settings) { 90 | return match ($command) { 91 | 'cimage', 'contacts' => true, 92 | default => false, 93 | }; 94 | } 95 | 96 | public function ajaxCustomHandler() { 97 | switch($_REQUEST['command']) { 98 | case "cimage": 99 | if($this->UCP->Modules->moduleHasMethod('Contactmanager', 'userDetails')) { 100 | $did = $_REQUEST['did']; 101 | $result = $this->UCP->FreePBX->Contactmanager->lookupByUserID($this->user['id'], $did,"/\D/"); 102 | if($result['image']) { 103 | dbug($result); 104 | $data = $this->UCP->FreePBX->Contactmanager->getImageByID($result['id'],$result['email'], $result['type']); 105 | if(!empty($data)) { 106 | $finfo = new \finfo(FILEINFO_MIME); 107 | header("Cache-Control: no-cache, must-revalidate"); 108 | header("Expires: Sat, 26 Jul 1997 05:00:00 GMT"); 109 | header("Content-type: ".$finfo->buffer($data['image'])); 110 | echo $data['image']; 111 | return true; 112 | } 113 | } 114 | } 115 | $contents = file_get_contents(__DIR__."/assets/images/no_user_logo.png"); 116 | $finfo = new \finfo(FILEINFO_MIME); 117 | header("Cache-Control: no-cache, must-revalidate"); 118 | header("Expires: Sat, 26 Jul 1997 05:00:00 GMT"); 119 | header("Content-type: ".$finfo->buffer($contents)); 120 | echo $contents; 121 | return true; 122 | break; 123 | } 124 | } 125 | 126 | /** 127 | * The Handler for all ajax events releated to this class 128 | * 129 | * Used by Ajax Class to process commands 130 | * 131 | * @return mixed Output if success, otherwise false will generate a 500 error serverside 132 | */ 133 | function ajaxHandler() { 134 | $return = ["status" => false, "message" => ""]; 135 | switch($_REQUEST['command']) { 136 | case "contacts": 137 | $return = []; 138 | if($this->Modules->moduleHasMethod('Contactmanager','lookupMultiple')) { 139 | $search = !empty($_REQUEST['search']) ? $_REQUEST['search'] : ""; 140 | $results = $this->Modules->Contactmanager->lookupMultiple($search); 141 | if(!empty($results)) { 142 | foreach($results as $res) { 143 | foreach($res['numbers'] as $type => $num) { 144 | if(!empty($num)) { 145 | $return[] = ["value" => $num, "text" => $res['displayname'] . " (".$type.")"]; 146 | } 147 | } 148 | } 149 | } 150 | } 151 | break; 152 | default: 153 | return false; 154 | break; 155 | } 156 | return $return; 157 | } 158 | 159 | /** 160 | * Send settings to UCP upon initalization 161 | */ 162 | function getStaticSettings() { 163 | $settings = $this->webrtc->getClientSettingsByUser($this->ext); 164 | if(!empty($settings['status'])) { 165 | return ['enabled' => true, 'settings' => ['wsservers' => $settings['websocket'], 'uri' => $settings['sipuri'], 'password' => $settings['password'], 'log' => 3, 'iceServers' => [$settings['stunaddr']], 'gatheringTimeout' => 2000], 'extensions' => [$this->user['default_extension']]]; 166 | } else { 167 | if(!empty($settings['message'])) { 168 | return ['enabled' => false, "message" => $settings['message']]; 169 | } else { 170 | return ['enabled' => false]; 171 | } 172 | } 173 | } 174 | } 175 | -------------------------------------------------------------------------------- /ucp/assets/images/no_user_logo.png: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/FreePBX/webrtc/92ce6acc8eced21055a5b63267b764a8609e4cae/ucp/assets/images/no_user_logo.png -------------------------------------------------------------------------------- /ucp/assets/js/global.js: -------------------------------------------------------------------------------- 1 | var WebrtcC = UCPMC.extend({ 2 | init: function() { 3 | this.phone = null; 4 | this.activeCalls = {}; 5 | this.activeCallId = null; 6 | this.answering = false; 7 | this.userBlocked = false; 8 | this.silenced = false; 9 | this.autoRegister = false; 10 | this.displayState = null; 11 | this.state = null; 12 | this.timerObject = null; 13 | this.callBinds = [ 14 | "progress", 15 | "accepted", 16 | "rejected", 17 | "failed", 18 | "terminated", 19 | "cancel", 20 | "refer", 21 | "replaced", 22 | "dtmf", 23 | "muted", 24 | "unmuted", 25 | "bye", 26 | "addStream" 27 | ]; 28 | 29 | this.callOptions = { 30 | "media": { 31 | "constraints": { 32 | "audio": true, 33 | "video": false 34 | }, 35 | "render": { 36 | "remote": null 37 | } 38 | } 39 | }; 40 | 41 | this.notification = null; 42 | var st = Cookies.get("webrtc-silenced"); 43 | st = (st === "1") ? true : false; 44 | this.silence(st); 45 | 46 | var rg = Cookies.get("webrtc-register"); 47 | this.autoRegister = (typeof rg === "undefined" || rg === "1") ? true : false; 48 | }, 49 | settingsDisplay: function() { 50 | 51 | }, 52 | settingsHide: function() { 53 | 54 | }, 55 | addSimpleWidget: function(widget_id) { 56 | this.initiateLibrary(); 57 | }, 58 | displaySimpleWidgetSettings: function(widget_id) { 59 | var $this = this; 60 | 61 | var st = Cookies.get("webrtc-silenced"); 62 | st = (st === "1") ? true : false; 63 | 64 | $("#webrtc-silence-switch").prop("checked",st); 65 | 66 | $("#webrtc-silence-switch").bootstrapToggle('destroy'); 67 | $("#webrtc-silence-switch").bootstrapToggle({ 68 | on: _("Enable"), 69 | off: _("Disable") 70 | }); 71 | 72 | $("#webrtc-disconnect-switch").prop("checked",!this.autoRegister); 73 | 74 | $("#webrtc-disconnect-switch").bootstrapToggle('destroy'); 75 | $("#webrtc-disconnect-switch").bootstrapToggle({ 76 | on: _("Enable"), 77 | off: _("Disable") 78 | }); 79 | 80 | if(this.phone === null) { 81 | $("#webrtc-silence-switch").bootstrapToggle('disable'); 82 | $("#webrtc-disconnect-switch").bootstrapToggle('disable'); 83 | return; 84 | } 85 | 86 | $("#webrtc-silence-switch").change(function() { 87 | $this.silence(); 88 | }); 89 | $("#webrtc-disconnect-switch").change(function(e) { 90 | $this.toggleRegister(); 91 | }); 92 | }, 93 | displaySimpleWidget: function(widget_id) { 94 | var $this = this; 95 | $("#menu_webrtc_phone .status span").text(this.displayState); 96 | 97 | if(this.phone === null) { 98 | $("#menu_webrtc_phone input.dialpad").prop("disabled",true); 99 | return; 100 | } 101 | 102 | if(this.state == "hold") { 103 | this.switchState('accepted'); 104 | this.switchState('hold'); 105 | } else { 106 | this.switchState(this.state); 107 | } 108 | 109 | if(typeof this.phone === "object" && this.phone !== null && this.phone.isRegistered()) { 110 | $("#menu_webrtc_phone .action").prop("disable",false); 111 | } 112 | 113 | $("#menu_webrtc_phone .keypad td").click(function() { 114 | var text = $("#menu_webrtc_phone .dialpad").val() + $(this).data("num"), 115 | button = $("#menu_webrtc_phone button.action"); 116 | if ($this.state == "registered" || $this.state == "accepted") { 117 | if ($this.state == "registered") { 118 | $( "#menu_webrtc_phone .message").text("To: " + text); 119 | } 120 | $("#menu_webrtc_phone .dialpad").val(text); 121 | $this.DTMF($(this).data("num")); 122 | button.prop("disabled", false); 123 | $("#menu_webrtc_phone .message-container").textfill(); 124 | } 125 | }); 126 | 127 | $("#menu_webrtc_phone .clear-input").click(function() { 128 | var button = $("#menu_webrtc_phone button.action"); 129 | $("#menu_webrtc_phone .dialpad").val(""); 130 | if ($this.state == "registered") { 131 | $( "#menu_webrtc_phone .message").text(""); 132 | button.prop("disabled", true); 133 | } 134 | }); 135 | $("#menu_webrtc_phone .dialpad").on('keyup paste', function() { 136 | var button = $("#menu_webrtc_phone button.action"), 137 | text = $("#menu_webrtc_phone .dialpad").val(); 138 | if ($(this).val().length === 0 && ($this.state == "accepted" || $this.state == "registered")) { 139 | $( "#menu_webrtc_phone .message").text(""); 140 | button.prop("disabled", true); 141 | } else { 142 | $( "#menu_webrtc_phone .message").text("To: " + text); 143 | $this.DTMF(text.slice(-1)); 144 | button.prop("disabled", false); 145 | } 146 | $("#menu_webrtc_phone .message-container").textfill(); 147 | }); 148 | $("#menu_webrtc_phone button.action").click(function() { 149 | switch ($this.state) { 150 | case "registered": 151 | $this.call($("#menu_webrtc_phone .dialpad").val()); 152 | break; 153 | case "hold": 154 | case "accepted": 155 | $this.hangup(); 156 | break; 157 | case "progress": 158 | $this.abortCall(); 159 | break; 160 | case "invite": 161 | $this.answer(); 162 | break; 163 | } 164 | }); 165 | $("#menu_webrtc_phone button.secondaction").click(function() { 166 | switch ($this.state) { 167 | case "hold": 168 | case "accepted": 169 | $this.toggleHold(); 170 | break; 171 | case "invite": 172 | $this.hangup(); 173 | break; 174 | } 175 | }); 176 | $("#menu_webrtc_phone .message-container").textfill(); 177 | }, 178 | deleteSimpleWidget: function(widget_id) { 179 | if(this.phone !== null) { 180 | this.disconnect(); 181 | } 182 | }, 183 | engineEvent: function(type, event) { 184 | console.log("Engine " + type); 185 | switch (type){ 186 | case "invite": 187 | this.manageSession(event,"inbound"); 188 | this.switchState("invite"); 189 | break; 190 | case "registered": 191 | this.switchState("registered"); 192 | break; 193 | case "unregistered": 194 | this.switchState("unregistered"); 195 | break; 196 | case "registrationFailed": 197 | this.switchState("registrationfailed"); 198 | break; 199 | case "connected": 200 | this.switchState("connected"); 201 | break; 202 | case "disconnected": 203 | this.switchState("disconnected"); 204 | break; 205 | case "connecting": 206 | this.switchState("connecting"); 207 | break; 208 | case "registering": //custom event type 209 | this.switchState("registering"); 210 | break; 211 | } 212 | }, 213 | setDisplayState: function(state) { 214 | this.displayState = state; 215 | $("#menu_webrtc_phone .status span").text(this.displayState); 216 | }, 217 | playRing: function() { 218 | if(!this.silenced) { 219 | $("#ringtone").trigger("play"); 220 | } 221 | }, 222 | stopRing: function() { 223 | $("#ringtone").trigger("pause"); 224 | $("#ringtone").trigger("load"); 225 | }, 226 | playRingBack: function() { 227 | if(!this.silenced) { 228 | $("#ringback").trigger("play"); 229 | } 230 | }, 231 | stopRingBack: function() { 232 | $("#ringback").trigger("pause"); 233 | $("#ringback").trigger("load"); 234 | }, 235 | 236 | manageSession: function(session, direction) { 237 | var Webrtc = this, 238 | id, 239 | displayName, 240 | status, 241 | cnum, 242 | cnam, 243 | call = session; 244 | 245 | id = Math.floor((Math.random() * 100000) + 1); 246 | // If the session exists with active call reject it. 247 | // TODO this can be useful for call waiting 248 | if (this.activeCallId) { 249 | call.terminate(); 250 | return false; 251 | } 252 | 253 | // If this is a new session create it 254 | if (!this.activeCallId) { 255 | this.activeCallId = id; 256 | this.activeCalls[id] = call; 257 | } 258 | 259 | cnum = this.activeCalls[id].remoteIdentity.uri.user; 260 | cnam = this.activeCalls[this.activeCallId].remoteIdentity.displayName || ""; 261 | displayName = (cnam !== "") ? cnam + " <" + cnum + ">" : cnum; 262 | $("#menu_webrtc_phone .contactDisplay .contactImage").css("background-image",'url("?quietmode=1&module=Webrtc&command=cimage&did='+cnum+'")'); 263 | Webrtc.answering = false; 264 | if (direction === "inbound") { 265 | if (UCP.notify) { 266 | this.notification = new Notify(sprintf(_("Incoming call from %s"), displayName), { 267 | body: _("Click this window to answer or close this window to ignore"), 268 | icon: "modules/Webrtc/assets/images/no_user_logo.png", //TODO: get the user logo 269 | notifyClose: function() { 270 | if (Webrtc.answering) { 271 | Webrtc.answering = false; 272 | } else { 273 | Webrtc.hangup(); 274 | } 275 | }, 276 | notifyClick: function() { 277 | Webrtc.answering = true; 278 | Webrtc.answer(); 279 | $(".custom-widget[data-widget_rawname=webrtc]").click(); 280 | Webrtc.notification.close(); 281 | } 282 | }); 283 | this.notification.show(); 284 | } 285 | } 286 | 287 | $.each(this.callBinds, function(i, v) { 288 | Webrtc.activeCalls[Webrtc.activeCallId].on(v, function(data, cause) { 289 | Webrtc.sessionEvent(v, data, cause); 290 | }); 291 | }); 292 | }, 293 | sessionEvent: function(type, data, cause) { 294 | console.log("Session " + type); 295 | switch (type){ 296 | case "terminated": 297 | this.switchState("terminated"); 298 | this.endCall(data, cause); 299 | break; 300 | case "accepted": 301 | this.switchState("accepted"); 302 | this.startCall(data); 303 | break; 304 | case "progress": 305 | this.switchState("progress"); 306 | break; 307 | case "dtmf": 308 | this.switchState("dtmf"); 309 | break; 310 | case "muted": 311 | this.switchState("muted"); 312 | break; 313 | case "unmuted": 314 | this.switchState("unmuted"); 315 | break; 316 | } 317 | }, 318 | endCall: function(message, cause) { 319 | this.activeCalls[this.activeCallId] = null; 320 | this.activeCallId = null; 321 | if (this.notification !== null) { 322 | this.notification.close(); 323 | } 324 | if(typeof cause !== "undefined" && cause === SIP.C.causes.USER_DENIED_MEDIA_ACCESS) { 325 | this.userBlocked = true; 326 | } 327 | $("#menu_webrtc_phone .btn-primary").prop("disabled", false); 328 | this.stopRing(); 329 | this.stopRingBack(); 330 | }, 331 | startCall: function(event) { 332 | if (this.notification !== null) { 333 | this.notification.close(); 334 | } 335 | this.stopRing(); 336 | }, 337 | silence: function(state) { 338 | state = (typeof state !== "undefined") ? state : !this.silenced; 339 | if(!$("#webrtc-silence").length) { 340 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").after(''); 341 | } 342 | if(state) { 343 | this.stopRing(); 344 | $("#webrtc-silence").removeClass("hidden"); 345 | $("#webrtc-silence .fa-check").removeClass("hidden"); 346 | } else { 347 | $("#webrtc-silence").addClass("hidden"); 348 | $("#webrtc-silence .fa-check").addClass("hidden"); 349 | } 350 | Cookies.set("webrtc-silenced",(state ? "1" : "0")); 351 | this.silenced = state; 352 | }, 353 | call: function(number) { 354 | if (this.phone.isConnected() && !this.userBlocked) { 355 | $("#menu_webrtc_phone .btn-primary").prop("disabled", true); 356 | var session = this.phone.invite(number, this.callOptions); 357 | this.manageSession(session,"outbound"); 358 | } else if(this.phone.isConnected() && this.userBlocked) { 359 | alert(_("Unable to start call. Please allow the WebRTC session in your browser and refresh")); 360 | } 361 | }, 362 | answer: function() { 363 | if (this.activeCallId !== null) { 364 | this.answering = true; 365 | this.activeCalls[this.activeCallId].accept(this.callOptions); 366 | } 367 | }, 368 | toggleHold: function() { 369 | if (this.activeCallId !== null) { 370 | var call = this.activeCalls[this.activeCallId], 371 | holds = this.activeCalls[this.activeCallId].isOnHold(); 372 | if (!holds.local) { 373 | this.switchState("hold"); 374 | call.hold(); 375 | } else { 376 | this.switchState("unhold"); 377 | call.unhold(); 378 | } 379 | } 380 | }, 381 | DTMF: function(num) { 382 | if (this.state == "accepted" && this.activeCallId !== null) { 383 | this.activeCalls[this.activeCallId].dtmf(num); 384 | } 385 | }, 386 | hangup: function() { 387 | if ((this.state == "accepted" || this.state == "invite") && this.activeCallId !== null) { 388 | this.activeCalls[this.activeCallId].terminate(); 389 | } 390 | this.stopRing(); 391 | this.stopRingBack(); 392 | }, 393 | abortCall: function() { 394 | if (this.activeCallId !== null) { 395 | this.activeCalls[this.activeCallId].terminate(); 396 | } 397 | this.stopRing(); 398 | this.stopRingBack(); 399 | }, 400 | poll: function(data) { 401 | 402 | }, 403 | display: function(event) { 404 | 405 | }, 406 | hide: function(event) { 407 | 408 | }, 409 | switchState: function(t) { 410 | var button = $("#menu_webrtc_phone button.action"), 411 | secondbutton = $("#menu_webrtc_phone button.secondaction"), 412 | input = $("#menu_webrtc_phone input.dialpad"), 413 | type = (typeof t !== "undefined" && t !== null) ? t : "registered", 414 | $this = this; 415 | this.state = type; 416 | button.data("type", type); 417 | switch (type){ 418 | case "dtmf": 419 | this.state = "accepted"; 420 | break; 421 | case "invite": 422 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").addClass("shake"); 423 | this.playRing(); 424 | $("#menu_webrtc_phone .activeCallSession .keypad").hide(); 425 | $("#menu_webrtc_phone .activeCallSession .input-container").hide(); 426 | $("#menu_webrtc_phone .contactDisplay").show(); 427 | secondbutton.removeClass().addClass("btn btn-danger secondaction").text("Ignore"); 428 | $("#menu_webrtc_phone .actions .right").show(); 429 | button.removeClass().addClass("btn btn-success action").text("Answer"); 430 | button.prop("disabled", false); 431 | break; 432 | case "hold": 433 | secondbutton.removeClass().addClass("btn btn-success secondaction").text("Resume"); 434 | secondbutton.css("background-color","orange"); 435 | if(!$("#webrtc-hold").length) { 436 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").after(''); 437 | } 438 | $("#webrtc-hold").removeClass("hidden"); 439 | break; 440 | case "unhold": 441 | secondbutton.removeClass().addClass("btn btn-success secondaction").text("Hold"); 442 | secondbutton.css("background-color",""); 443 | if($("#webrtc-hold").length) { 444 | $("#webrtc-hold").addClass("hidden"); 445 | } 446 | 447 | this.state = "accepted"; 448 | break; 449 | case "progress": 450 | this.playRingBack(); 451 | button.prop("disabled", false); 452 | button.removeClass().addClass("btn btn-danger action").text("Cancel"); 453 | break; 454 | case "accepted": 455 | this.stopRingBack(); 456 | this.stopRing(); 457 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").removeClass("shake"); 458 | $("#menu_webrtc_phone .contactDisplay").hide(); 459 | $("#menu_webrtc_phone .activeCallSession .keypad").show(); 460 | $("#menu_webrtc_phone .activeCallSession .input-container").show(); 461 | secondbutton.removeClass().addClass("btn btn-success secondaction").text("Hold"); 462 | secondbutton.css("color",""); 463 | $("#menu_webrtc_phone .actions .right").show(); 464 | 465 | input.prop("disabled", false); 466 | button.prop("disabled", false); 467 | button.removeClass().addClass("btn btn-danger action").text("Hangup"); 468 | $("#menu_webrtc_phone .contact-info").removeClass("fade"); 469 | $("#webrtc-timer-container").remove(); 470 | clearInterval(this.timerObject); 471 | $('#webrtc-disconnect-switch').bootstrapToggle('disable'); 472 | var updateTimer = function() { 473 | if($this.activeCallId === null) { 474 | clearInterval($this.timerObject); 475 | $("#menu_webrtc_phone .contact-info").addClass("fade"); 476 | $('#webrtc-disconnect-switch').bootstrapToggle('enable'); 477 | return; 478 | } 479 | // 480 | var start = moment($this.activeCalls[$this.activeCallId].startTime); 481 | var end = moment(); 482 | var duration = moment.duration(end.diff(start)); 483 | 484 | var padLeft = function(nr){ 485 | return Array(2-String(nr).length+1).join('0')+nr; 486 | }; 487 | 488 | var time = padLeft(duration.hours())+":"+padLeft(duration.minutes())+":"+padLeft(duration.seconds()); 489 | 490 | if($("#menu_webrtc_phone .contact-info .timer").is(":visible")) { 491 | $("#menu_webrtc_phone .contact-info .timer").text(time); 492 | } else { 493 | if(!$("#webrtc-timer-container").length) { 494 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").after('
'+time+'
'); 495 | } else { 496 | $("#webrtc-timer-container .timer").text(time); 497 | } 498 | } 499 | }; 500 | updateTimer(); 501 | this.timerObject = setInterval(updateTimer,1000); 502 | 503 | var cnam = this.activeCalls[this.activeCallId].remoteIdentity.displayName || "", 504 | cnum = this.activeCalls[this.activeCallId].remoteIdentity.uri.user, 505 | displayName = (cnam !== "") ? cnam + " <" + cnum + ">" : cnum; 506 | $("#menu_webrtc_phone .contact-info .contact").text(displayName); 507 | break; 508 | case "terminated": 509 | this.stopRing(); 510 | this.stopRingBack(); 511 | $("#menu_webrtc_phone .actions .right").hide(); 512 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").removeClass("shake"); 513 | $("#menu_webrtc_phone .activeCallSession .keypad").show(); 514 | $("#menu_webrtc_phone .activeCallSession .input-container").show(); 515 | $("#menu_webrtc_phone .contactDisplay").hide(); 516 | button.removeClass().addClass("btn btn-primary action").text("Call"); 517 | $("#menu_webrtc_phone .contact-info .contact").text(""); 518 | this.state = "registered"; 519 | break; 520 | case "registered": 521 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").removeClass("registering"); 522 | this.setDisplayState(_("Registered")); 523 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").css("color", "green"); 524 | input.prop("disabled", false); 525 | input.val(""); 526 | $("#menu_webrtc_phone .keypad").removeClass("disable"); 527 | button.prop("disabled", true); 528 | $("#menu_webrtc_phone .actions .right").hide(); 529 | button.removeClass().addClass("btn btn-primary action").text("Call"); 530 | break; 531 | case "unregistered": 532 | this.setDisplayState(_("Unregistered")); 533 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").removeClass("registering"); 534 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").css("color", "yellow"); 535 | $("#menu_webrtc_phone .keypad").addClass("disable"); 536 | input.prop("disabled", true); 537 | input.val(""); 538 | break; 539 | case "registrationfailed": 540 | this.setDisplayState(_("Registration Failed")); 541 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").removeClass("registering"); 542 | $("#webrtc-dc a span").text(_("Connect Phone")); 543 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").css("color", "red"); 544 | $("#menu_webrtc_phone .keypad").addClass("disable"); 545 | input.prop("disabled", true); 546 | input.val(""); 547 | break; 548 | case "connected": 549 | this.setDisplayState(_("Unregistered")); 550 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").removeClass("connecting"); 551 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").css("color", "yellow"); 552 | break; 553 | case "disconnected": 554 | this.setDisplayState(_("Disconnected")); 555 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").removeClass("connecting"); 556 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").removeClass("registering"); 557 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").css("color", "red"); 558 | $("#menu_webrtc_phone .keypad").addClass("disable"); 559 | input.prop("disabled", true); 560 | input.val(""); 561 | break; 562 | case "connecting": 563 | this.setDisplayState(_("Connecting to socket...")); 564 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").css("color", "red"); 565 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").addClass("connecting"); 566 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").removeClass("registering"); 567 | break; 568 | case "registering": //custom event type 569 | this.setDisplayState(_("Registering...")); 570 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").addClass("registering"); 571 | break; 572 | } 573 | }, 574 | connect: function() { 575 | if ((typeof this.staticsettings !== "undefined") && 576 | this.staticsettings.enabled && 577 | Modernizr.getusermedia && 578 | this.phone !== null && 579 | !this.phone.isConnected()) { 580 | this.phone.start(); 581 | } 582 | }, 583 | disconnect: function() { 584 | if (this.phone !== null && 585 | this.phone.isConnected()) { 586 | this.phone.stop(); 587 | } 588 | }, 589 | register: function() { 590 | if(!this.phone.isConnected()) { 591 | this.connect(); 592 | } 593 | if (this.phone !== null && 594 | !this.phone.isRegistered()) { 595 | } 596 | this.phone.register(); 597 | }, 598 | unregister: function() { 599 | if(!this.phone.isConnected()) { 600 | throw "Phone is not connected, nothing to register"; 601 | } 602 | if (this.phone !== null && 603 | this.phone.isRegistered()) { 604 | } 605 | this.phone.unregister(); 606 | }, 607 | toggleRegister: function() { 608 | if(!this.phone.isConnected()) { 609 | return; //nope 610 | } 611 | if($(".custom-widget[data-widget_rawname=webrtc] .fa-phone").hasClass("registering")) { 612 | return; //we are already doing something 613 | } 614 | if(!this.phone.isRegistered()) { 615 | this.register(); 616 | Cookies.set("webrtc-register",1); 617 | } else { 618 | this.unregister(); 619 | Cookies.set("webrtc-register",0); 620 | } 621 | 622 | }, 623 | initiateLibrary: function() { 624 | var $this = this, 625 | ver = "0.7.7"; 626 | 627 | if(typeof SIP === "object") { 628 | return; 629 | } 630 | 631 | if(!$("html").hasClass("getusermedia")) { 632 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").css("color", "red"); 633 | this.setDisplayState(_("Not supported in this browser")); 634 | console.warn("WebRTC is not supported in this browser"); 635 | return; 636 | } 637 | 638 | if(document.location.protocol !== "https:") { 639 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").css("color", "red"); 640 | this.setDisplayState(_("Only supported over HTTPS")); 641 | console.warn("WebRTC is not supported in non-SSL mode"); 642 | return; 643 | } 644 | 645 | if(!$(".custom-widget[data-widget_rawname=webrtc]").length) { 646 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").css("color", "red"); 647 | console.warn("WebRTC Widget has not been added"); 648 | return; 649 | } 650 | 651 | if(typeof moduleSettings.Webrtc === "undefined") { 652 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").css("color", "red"); 653 | console.warn("WebRTC is not configured properly"); 654 | return; 655 | } 656 | 657 | if(!moduleSettings.Webrtc.enabled) { 658 | $(".custom-widget[data-widget_rawname=webrtc] .fa-phone").css("color", "red"); 659 | console.warn(moduleSettings.Webrtc.message); 660 | this.setDisplayState(moduleSettings.Webrtc.message); 661 | return; 662 | } 663 | 664 | $.getScript("modules/Webrtc/assets/jssiplibs/sip-" + ver + ".min.js") 665 | .done(function( script, textStatus ) { 666 | $("#footer").append("