luci-app-pbx(-voicemail): remove from repo

Remove the luci-app-pbx(-voicemail) packages, as they have been
marked BROKEN since August 2016.

The pbx packages depend on asterisk 1.8 that is EOL upstream
and has been moved from the telephony feed to the abandoned feed.

If LuCI pbx packages are still needed, they should be refreshed
to depend on a current asterisk version from the telephony feed,
and to match the features of that asterisk version.

Signed-off-by: Hannu Nyman <hannu.nyman@iki.fi>

132 files changed, 0 insertions, 21339 deletions

diff --git a/applications/luci-app-pbx-voicemail/COPYING b/applications/luci-app-pbx-voicemail/COPYING
deleted file mode 100644
index 94a9ed024d..0000000000
--- a/applications/luci-app-pbx-voicemail/COPYING
+++ /dev/null
@@ -1,674 +0,0 @@
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-  THERE IS NO WARRANTY FOR THE PROGRAM, TO THE EXTENT PERMITTED BY
-APPLICABLE LAW.  EXCEPT WHEN OTHERWISE STATED IN WRITING THE COPYRIGHT
-HOLDERS AND/OR OTHER PARTIES PROVIDE THE PROGRAM "AS IS" WITHOUT WARRANTY
-OF ANY KIND, EITHER EXPRESSED OR IMPLIED, INCLUDING, BUT NOT LIMITED TO,
-THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
-PURPOSE.  THE ENTIRE RISK AS TO THE QUALITY AND PERFORMANCE OF THE PROGRAM
-IS WITH YOU.  SHOULD THE PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF
-ALL NECESSARY SERVICING, REPAIR OR CORRECTION.
-
-  16. Limitation of Liability.
-
-  IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING
-WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MODIFIES AND/OR CONVEYS
-THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES, INCLUDING ANY
-GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING OUT OF THE
-USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED TO LOSS OF
-DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY YOU OR THIRD
-PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER PROGRAMS),
-EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF
-SUCH DAMAGES.
-
-  17. Interpretation of Sections 15 and 16.
-
-  If the disclaimer of warranty and limitation of liability provided
-above cannot be given local legal effect according to their terms,
-reviewing courts shall apply local law that most closely approximates
-an absolute waiver of all civil liability in connection with the
-Program, unless a warranty or assumption of liability accompanies a
-copy of the Program in return for a fee.
-
-                     END OF TERMS AND CONDITIONS
-
-            How to Apply These Terms to Your New Programs
-
-  If you develop a new program, and you want it to be of the greatest
-possible use to the public, the best way to achieve this is to make it
-free software which everyone can redistribute and change under these terms.
-
-  To do so, attach the following notices to the program.  It is safest
-to attach them to the start of each source file to most effectively
-state the exclusion of warranty; and each file should have at least
-the "copyright" line and a pointer to where the full notice is found.
-
-    <one line to give the program's name and a brief idea of what it does.>
-    Copyright (C) <year>  <name of author>
-
-    This program is free software: you can redistribute it and/or modify
-    it under the terms of the GNU General Public License as published by
-    the Free Software Foundation, either version 3 of the License, or
-    (at your option) any later version.
-
-    This program is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-    GNU General Public License for more details.
-
-    You should have received a copy of the GNU General Public License
-    along with this program.  If not, see <http://www.gnu.org/licenses/>.
-
-Also add information on how to contact you by electronic and paper mail.
-
-  If the program does terminal interaction, make it output a short
-notice like this when it starts in an interactive mode:
-
-    <program>  Copyright (C) <year>  <name of author>
-    This program comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
-    This is free software, and you are welcome to redistribute it
-    under certain conditions; type `show c' for details.
-
-The hypothetical commands `show w' and `show c' should show the appropriate
-parts of the General Public License.  Of course, your program's commands
-might be different; for a GUI interface, you would use an "about box".
-
-  You should also get your employer (if you work as a programmer) or school,
-if any, to sign a "copyright disclaimer" for the program, if necessary.
-For more information on this, and how to apply and follow the GNU GPL, see
-<http://www.gnu.org/licenses/>.
-
-  The GNU General Public License does not permit incorporating your program
-into proprietary programs.  If your program is a subroutine library, you
-may consider it more useful to permit linking proprietary applications with
-the library.  If this is what you want to do, use the GNU Lesser General
-Public License instead of this License.  But first, please read
-<http://www.gnu.org/philosophy/why-not-lgpl.html>.
diff --git a/applications/luci-app-pbx-voicemail/Makefile b/applications/luci-app-pbx-voicemail/Makefile
deleted file mode 100644
index eefe0fd249..0000000000
--- a/applications/luci-app-pbx-voicemail/Makefile
+++ /dev/null
@@ -1,14 +0,0 @@
-#
-# Copyright (C) 2008-2014 The LuCI Team <luci@lists.subsignal.org>
-#
-# This is free software, licensed under the Apache License, Version 2.0 .
-#
-
-include $(TOPDIR)/rules.mk
-
-LUCI_TITLE:=LuCI PBX Administration Voicemail Support
-LUCI_DEPENDS:=+luci-app-pbx +asterisk18 +msmtp +coreutils-base64 @BROKEN
-
-include ../../luci.mk
-
-# call BuildPackage - OpenWrt buildroot signature
diff --git a/applications/luci-app-pbx-voicemail/luasrc/controller/pbx-voicemail.lua b/applications/luci-app-pbx-voicemail/luasrc/controller/pbx-voicemail.lua
deleted file mode 100644
index 6f3dfac05e..0000000000
--- a/applications/luci-app-pbx-voicemail/luasrc/controller/pbx-voicemail.lua
+++ /dev/null
@@ -1,24 +0,0 @@
---[[
-    Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com>
-
-    This file is part of luci-pbx-voicemail.
-
-    luci-pbx-voicemail is free software: you can redistribute it and/or modify
-    it under the terms of the GNU General Public License as published by
-    the Free Software Foundation, either version 3 of the License, or
-    (at your option) any later version.
-
-    luci-pbx-voicemail is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-    GNU General Public License for more details.
-
-    You should have received a copy of the GNU General Public License
-    along with luci-pbx-voicemail.  If not, see <http://www.gnu.org/licenses/>.
-]]--
-
-module("luci.controller.pbx-voicemail", package.seeall)
-
-function index()
-        entry({"admin", "services", "pbx", "pbx-voicemail"}, cbi("pbx-voicemail"), "Voicemail", 5)
-end
diff --git a/applications/luci-app-pbx-voicemail/luasrc/model/cbi/pbx-voicemail.lua b/applications/luci-app-pbx-voicemail/luasrc/model/cbi/pbx-voicemail.lua
deleted file mode 100644
index a6087e9aec..0000000000
--- a/applications/luci-app-pbx-voicemail/luasrc/model/cbi/pbx-voicemail.lua
+++ /dev/null
@@ -1,153 +0,0 @@
---[[
-    Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com>
-
-    This file is part of luci-pbx-voicemail.
-
-    luci-pbx-voicemail is free software: you can redistribute it and/or modify
-    it under the terms of the GNU General Public License as published by
-    the Free Software Foundation, either version 3 of the License, or
-    (at your option) any later version.
-
-    luci-pbx-voicemail is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-    GNU General Public License for more details.
-
-    You should have received a copy of the GNU General Public License
-    along with luci-pbx-voicemail.  If not, see <http://www.gnu.org/licenses/>.
-]]--
-
-if     nixio.fs.access("/etc/init.d/asterisk")   then
-   server = "asterisk"
-elseif nixio.fs.access("/etc/init.d/freeswitch") then
-   server = "freeswitch"
-else
-   server = ""
-end
-
-modulename = "pbx-voicemail"
-vmlogfile  = "/tmp/last_sent_voicemail.log"
-
-m = Map (modulename, translate("Voicemail Setup"),
-         translate("Here you can configure a global voicemail for this PBX. Since this system is \
-         intended to run on embedded systems like routers, there is no local storage of voicemail - \
-         it must be sent out by email. Therefore you need to configure an outgoing mail (SMTP) server \
-         (for example your ISP's, Google's, or Yahoo's SMTP server), and provide a list of \
-         addresses that receive recorded voicemail."))
-
--- Recreate the config, and restart services after changes are commited to the configuration.
-function m.on_after_commit(self)
-        luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null")
-        luci.sys.call("/etc/init.d/"     .. server .. " restart 1\>/dev/null 2\>/dev/null")
-end
-
-
-----------------------------------------------------------------------------------------------------
-s = m:section(NamedSection, "global_voicemail", "voicemail", translate("Global Voicemail Setup"),
-              translate("When you enable voicemail, you will have the opportunity to specify \
-              email addresses that receive recorded voicemail. You must also set up an SMTP server below."))
-s.anonymous = true
-
-enable = s:option(ListValue, "enabled", translate("Enable Voicemail"))
-enable:value("yes", translate("Yes"))
-enable:value("no",  translate("No"))
-enable.default = "no"
-
-emails = s:option(DynamicList, "global_email_addresses",
-                  translate("Email Addresses that Receive Voicemail"))
-emails:depends("enabled", "yes")
-
-savepath = s:option(Value, "global_save_path", translate("Local Storage Directory"),
-                    translate("You can also retain copies of voicemail messages on the device running \
-                              your PBX. The path specified here will be created if it doesn't exist. \
-                              Beware of limited space on embedded devices like routers, and enable this \
-                              option only if you know what you are doing."))
-savepath.optional = true
-
-if     nixio.fs.access("/etc/pbx-voicemail/recordings/greeting.gsm")   then
-   m1 = s:option(DummyValue, "_m1")
-   m1:depends("enabled", "yes")
-   m1.default = "NOTE: Found a voicemail greeting. To check or change your voicemail greeting, dial *789 \
-                 and the system will play back your current greeting. After that, a long beep will sound and \
-                 you can press * in order to record a new message. Hang up to avoid recording a message. \
-                 If you press *, a second long beep will sound, and you can record a new greeting. \
-                 Hang up or press # to stop recording. When # is pressed the system will play back the \
-                 new greeting."
-else
-   m1 = s:option(DummyValue, "_m1")
-   m1:depends("enabled", "yes")
-   m1.default = "WARNING: Could not find voicemail greeting. Callers will hear only a beep before \
-                 recording starts. To record a greeting, dial *789, and press * after the long beep. \
-                 If you press *, a second long beep will sound, and you can record a new greeting. \
-                 Hang up or press # to stop recording. When # is pressed the system will play back the \
-                 new greeting."
-end
-
-
-----------------------------------------------------------------------------------------------------
-s = m:section(NamedSection, "voicemail_smtp", "voicemail", translate("Outgoing mail (SMTP) Server"),
-              translate("In order for this PBX to send emails containing voicemail recordings, you need to \
-              set up an SMTP server here. Your ISP usually provides an SMTP server for that purpose. \
-              You can also set up a third party SMTP server such as the one provided by Google or Yahoo."))
-s.anonymous = true
-
-serv = s:option(Value, "smtp_server", translate("SMTP Server Hostname or IP Address"))
-serv.datatype = "host(0)"
-
-port = s:option(Value, "smtp_port", translate("SMTP Port Number"))
-port.datatype = "port"
-port.default = "25"
-
-tls = s:option(ListValue, "smtp_tls", translate("Secure Connection Using TLS"))
-tls:value("on",  translate("Yes"))
-tls:value("off", translate("No"))
-tls.default = "on"
-
-auth = s:option(ListValue, "smtp_auth", translate("SMTP Server Authentication"))
-auth:value("on",  translate("Yes"))
-auth:value("off", translate("No"))
-auth.default = "off"
-
-user = s:option(Value, "smtp_user", translate("SMTP User Name"))
-user:depends("smtp_auth", "on")
-
-pwd = s:option(Value, "smtp_password", translate("SMTP Password"),
-               translate("Your real SMTP password is not shown for your protection. It will be changed \
-                         only when you change the value in this box."))
-pwd.password = true
-pwd:depends("smtp_auth", "on")
-                            
--- We skip reading off the saved value and return nothing.
-function pwd.cfgvalue(self, section)
-   return "Password Not Displayed"
-end
-   
--- We check the entered value against the saved one, and only write if the entered value is
--- something other than the empty string, and it differes from the saved value.
-function pwd.write(self, section, value)
-   local orig_pwd = m:get(section, self.option)
-   if value == "Password Not Displayed" then value = "" end
-   if value and #value > 0 and orig_pwd ~= value then
-      Value.write(self, section, value)
-   end
-end
-
-----------------------------------------------------------------------------------------------------
-s = m:section(NamedSection, "voicemail_log", "voicemail", translate("Last Sent Voicemail Log"))
-s.anonymous = true
-
-s:option (DummyValue, "vmlog")
-
-sts = s:option(DummyValue, "_sts") 
-sts.template = "cbi/tvalue"
-sts.rows = 5
-
-function sts.cfgvalue(self, section)
-   log = nixio.fs.readfile(vmlogfile)
-   if log == nil or log == "" then
-      log = "No errors or messages reported."
-   end
-   return log
-end
-
-return m
diff --git a/applications/luci-app-pbx-voicemail/po/ca/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/ca/pbx-voicemail.po
deleted file mode 100644
index c0119f7735..0000000000
--- a/applications/luci-app-pbx-voicemail/po/ca/pbx-voicemail.po
+++ /dev/null
@@ -1,100 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2014-07-01 05:47+0200\n"
-"Last-Translator: Alex <alexhenrie24@gmail.com>\n"
-"Language-Team: none\n"
-"Language: ca\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr "Adreces de correu electrònic que reben correu de veu"
-
-msgid "Enable Voicemail"
-msgstr "Habilita el correu de veu"
-
-msgid "Global Voicemail Setup"
-msgstr ""
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-
-msgid "Last Sent Voicemail Log"
-msgstr "Registre del últim correu de veu enviat"
-
-msgid "Local Storage Directory"
-msgstr "Directori d'emmagatzematge local"
-
-msgid "No"
-msgstr "No"
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr "Servidor de correu sortint (SMTP)"
-
-msgid "SMTP Password"
-msgstr "Contrasenya SMTP"
-
-msgid "SMTP Port Number"
-msgstr "Nombre de port SMTP"
-
-msgid "SMTP Server Authentication"
-msgstr "Autenticació del servidor SMTP"
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr "Adreça IP o nom de host del servidor SMTP"
-
-msgid "SMTP User Name"
-msgstr "Nom d'usuari SMTP"
-
-msgid "Secure Connection Using TLS"
-msgstr "Assegura la connexió mitjançant TLS"
-
-msgid "Voicemail Setup"
-msgstr ""
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-"Quan habiliteu el correu de veu, tindreu l'oportunitat d'especificar adreces "
-"de correu electrònic que reben correu de veu gravat. Heu d'establir també "
-"un servidor SMTP a baix."
-
-msgid "Yes"
-msgstr "Sí"
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-"Podeu també retenir còpies de missatges de veu en el dispositiu executant el "
-"vostre PBX. La ruta especificat aquí es crearà si no existeix. Teniu compte "
-"d'espai limitat en dispositius incrustats com els encaminadors, i habiliteu "
-"aquesta opció només si coneixeu ho que feu."
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
-"La vostra contrasenya SMTP no es mostra per a la vostra protecció. Es "
-"canviarà només quan canvieu el valor en aquesta caixa."
diff --git a/applications/luci-app-pbx-voicemail/po/cs/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/cs/pbx-voicemail.po
deleted file mode 100644
index 6d94b3522c..0000000000
--- a/applications/luci-app-pbx-voicemail/po/cs/pbx-voicemail.po
+++ /dev/null
@@ -1,91 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2014-06-05 19:16+0200\n"
-"Last-Translator: koli <lukas.koluch@gmail.com>\n"
-"Language-Team: none\n"
-"Language: cs\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=3; plural=(n==1) ? 0 : (n>=2 && n<=4) ? 1 : 2;\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr ""
-
-msgid "Enable Voicemail"
-msgstr "Povolit Voicemail"
-
-msgid "Global Voicemail Setup"
-msgstr ""
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-
-msgid "Last Sent Voicemail Log"
-msgstr ""
-
-msgid "Local Storage Directory"
-msgstr ""
-
-msgid "No"
-msgstr "Ne"
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr ""
-
-msgid "SMTP Password"
-msgstr "SMTP heslo"
-
-msgid "SMTP Port Number"
-msgstr "SMTP číslo portu"
-
-msgid "SMTP Server Authentication"
-msgstr ""
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr ""
-
-msgid "SMTP User Name"
-msgstr "SMTP uživatelské jméno"
-
-msgid "Secure Connection Using TLS"
-msgstr ""
-
-msgid "Voicemail Setup"
-msgstr "Voicemail nastavení"
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-
-msgid "Yes"
-msgstr "Ano"
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
diff --git a/applications/luci-app-pbx-voicemail/po/de/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/de/pbx-voicemail.po
deleted file mode 100644
index 224b5a929b..0000000000
--- a/applications/luci-app-pbx-voicemail/po/de/pbx-voicemail.po
+++ /dev/null
@@ -1,141 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2013-01-28 22:30+0200\n"
-"Last-Translator: DAC324 <gerd_roethig@web.de>\n"
-"Language-Team: none\n"
-"Language: de\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr "E-Mail Adressen die Sprachnachrichten empfangen"
-
-msgid "Enable Voicemail"
-msgstr "Anrufbeantworter aktivieren"
-
-msgid "Global Voicemail Setup"
-msgstr "Allgemeine Einstellungen für Voicemail"
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-"Es kann ein systemweiter Anrufbeantworter für diese Telefonanlage (<abbr "
-"title=\"Private Branch eXchange\">PBX</abbr>) konfiguriert werden. Da dieses "
-"System für den Einsatz auf embedded Systemen wie Routern optimiert wurde, "
-"gibt es keine Möglichkeit die Sprachnachrichten lokal zu speichern. Sie "
-"müssen per E-Mail versendet werden. Daher muss ein ausgehender Mail-Server "
-"(<abbr title=\"Simple Mail Transfer Protocol\">SMTP</abbr>) konfiguriert "
-"werden. Hier kann zum Beispiel der SMTP-Server des Providers, aber auch ein "
-"Freemailer wie GMail eingetragen werden. Zusätzlich muss noch eine Liste von "
-"Adressen angegeben werden, an die aufgezeichnete Sprachnachrichten geschickt "
-"werden."
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-"Damit diese Telefonanlage (<abbr title=\"Private Branch eXchange\">PBX</abbr>) "
-"E-Mails mit Sprachaufnahmen senden kann, muss an dieser Stelle ein SMTP-"
-"Server eingetragen werden. Ihr Internet-Dienstanbieter (<abbr "
-"title=\"Internet Service Provider\">ISP</abbr>) stellt normalerweise einen "
-"SMTP-Server für diesen Zweck zur Verfügung. Sie können auch einen SMTP-"
-"Server eines Drittanbieters, wie z.B. Google oder Yahoo, hier einstellen."
-
-msgid "Last Sent Voicemail Log"
-msgstr "Log der zuletzt gesendeten Voicemails"
-
-msgid "Local Storage Directory"
-msgstr "Lokales Speicherverzeichnis"
-
-msgid "No"
-msgstr "Nein"
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr "Server für ausgehende Mails (SMTP)"
-
-msgid "SMTP Password"
-msgstr "SMTP-Passwort"
-
-msgid "SMTP Port Number"
-msgstr "SMTP-Portnummer"
-
-msgid "SMTP Server Authentication"
-msgstr "SMTP-Server-Anmeldung"
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr "SMTP-Servername oder IP-Adresse"
-
-msgid "SMTP User Name"
-msgstr "SMTP-Benutzername"
-
-msgid "Secure Connection Using TLS"
-msgstr "Sichere Verbindung über TLS"
-
-msgid "Voicemail Setup"
-msgstr "Voicemail-Einstellungen"
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-"Wenn Sie Voicemail aktivieren, können Sie E-Mail-Adressen angeben, die "
-"aufgenommene Voicemails erhalten sollen. Sie müssen ebenfalls einen SMTP-"
-"Server unten angeben."
-
-msgid "Yes"
-msgstr "Ja"
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-"Sie können auch Kopien Ihrer Voicemail-Nachrichten auf dem Gerät speichern. "
-"Der hier angegebene Pfad wird erstellt, falls er nicht existiert. Beachten "
-"Sie, dass Geräte wie Router nur begrenzten Speicherplatz haben, und "
-"aktivieren Sie diese Option nur, wenn Sie wissen, was Sie tun."
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
-"Ihr SMTP-Passwort wird zu Ihrem Schutz nicht angezeigt. Es wird nur "
-"geändert, wenn Sie dein Eintrag in diesem Kästchen verändern."
-
-#~ msgid "Directory to save voicemail into"
-#~ msgstr "Verzeichnis für eingehende Sprachnachrichten"
-
-#~ msgid "Enabled"
-#~ msgstr "Aktiviert"
-
-#~ msgid ""
-#~ "Here you can configure a global voicemail for this PBX. Since this system "
-#~ "is intended to run on embedded systems like routers, there is no local "
-#~ "storage of voicemail - it must be sent out by email. Therefore you need "
-#~ "to configure an outgoing mail (SMTP) server (for example the SMTP server "
-#~ "your ISP provides, or GMail), and provide a list of addresses the "
-#~ "voicemail will be sent to."
-#~ msgstr ""
-#~ "Es kann ein systemweiter Anrufbeantworter für diese Telefonanlage (<abbr "
-#~ "title=\"Private Branch eXchange\">PBX</abbr>) konfiguriert werden. Da "
-#~ "dieses System für den Einsatz auf embedded Systemen wie Routern optimiert "
-#~ "wurde, gibt es keine Möglichkeit die Sprachnachrichten lokal zu "
-#~ "speichern. Sie müssen per E-Mail versendet werden. Daher muss ein "
-#~ "ausgehender Mail-Server (<abbr title=\"Simple Mail Transfer Protocol"
-#~ "\">SMTP</abbr>) konfiguriert werden. Hier kann zum Beispiel der SMTP-"
-#~ "Server des Providers, aber auch ein Freemailer wie GMail verwendet "
-#~ "werden. Zusätzlich muss noch eine Liste von Adressen angegeben werden, zu "
-#~ "denen die Sprachnachrichten geschickt werden."
diff --git a/applications/luci-app-pbx-voicemail/po/el/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/el/pbx-voicemail.po
deleted file mode 100644
index fc0a2e351f..0000000000
--- a/applications/luci-app-pbx-voicemail/po/el/pbx-voicemail.po
+++ /dev/null
@@ -1,94 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2012-03-18 15:26+0200\n"
-"Last-Translator: Vasilis <acinonyx@openwrt.gr>\n"
-"Language-Team: none\n"
-"Language: el\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-"X-Generator: Pootle 2.0.4\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr ""
-
-msgid "Enable Voicemail"
-msgstr ""
-
-msgid "Global Voicemail Setup"
-msgstr ""
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-
-msgid "Last Sent Voicemail Log"
-msgstr ""
-
-msgid "Local Storage Directory"
-msgstr ""
-
-msgid "No"
-msgstr "Όχι"
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr ""
-
-msgid "SMTP Password"
-msgstr ""
-
-msgid "SMTP Port Number"
-msgstr ""
-
-msgid "SMTP Server Authentication"
-msgstr ""
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr ""
-
-msgid "SMTP User Name"
-msgstr ""
-
-msgid "Secure Connection Using TLS"
-msgstr ""
-
-msgid "Voicemail Setup"
-msgstr ""
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
-
-#~ msgid "Enabled"
-#~ msgstr "Ενεργοποιημένο"
diff --git a/applications/luci-app-pbx-voicemail/po/en/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/en/pbx-voicemail.po
deleted file mode 100644
index 79c8848bdb..0000000000
--- a/applications/luci-app-pbx-voicemail/po/en/pbx-voicemail.po
+++ /dev/null
@@ -1,169 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr "Email Addresses that Receive Voicemail"
-
-msgid "Enable Voicemail"
-msgstr "Enable Voicemail"
-
-msgid "Global Voicemail Setup"
-msgstr "Global Voicemail Setup"
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-
-msgid "Last Sent Voicemail Log"
-msgstr "Last Sent Voicemail Log"
-
-msgid "Local Storage Directory"
-msgstr "Local Storage Directory"
-
-msgid "No"
-msgstr "No"
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr "Outgoing mail (SMTP) Server"
-
-msgid "SMTP Password"
-msgstr "SMTP Password"
-
-msgid "SMTP Port Number"
-msgstr "SMTP Port Number"
-
-msgid "SMTP Server Authentication"
-msgstr "SMTP Server Authentication"
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr "SMTP Server Hostname or IP Address"
-
-msgid "SMTP User Name"
-msgstr "SMTP User Name"
-
-msgid "Secure Connection Using TLS"
-msgstr "Secure Connection Using TLS"
-
-msgid "Voicemail Setup"
-msgstr "Voicemail Setup"
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-
-msgid "Yes"
-msgstr "Yes"
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-
-#~ msgid "Directory to save voicemail into"
-#~ msgstr "Directory to save voicemail into"
-
-#~ msgid "Email addresses to forward to"
-#~ msgstr "Email addresses to forward to"
-
-#~ msgid "Enabled"
-#~ msgstr "Enabled"
-
-#~ msgid ""
-#~ "Here you can configure a global voicemail for this PBX. Since this system "
-#~ "is intended to run on embedded systems like routers, there is no local "
-#~ "storage of voicemail - it must be sent out by email. Therefore you need "
-#~ "to configure an outgoing mail (SMTP) server (for example the SMTP server "
-#~ "your ISP provides, or GMail), and provide a list of addresses the "
-#~ "voicemail will be sent to."
-#~ msgstr ""
-#~ "Here you can configure a global voicemail for this PBX. Since this system "
-#~ "is intended to run on embedded systems like routers, there is no local "
-#~ "storage of voicemail - it must be sent out by email. Therefore you need "
-#~ "to configure an outgoing mail (SMTP) server (for example the SMTP server "
-#~ "your ISP provides, or GMail), and provide a list of addresses the "
-#~ "voicemail will be sent to."
-
-#~ msgid ""
-#~ "In order for this PBX to send emails containing voicemail recordings, you "
-#~ "need to set up an SMTP server here. Your ISP usually provides an SMTP "
-#~ "server for that purpose. You can also set up a GMail, Yahoo, or other 3rd "
-#~ "party SMTP server."
-#~ msgstr ""
-#~ "In order for this PBX to send emails containing voicemail recordings, you "
-#~ "need to set up an SMTP server here. Your ISP usually provides an SMTP "
-#~ "server for that purpose. You can also set up a GMail, Yahoo, or other 3rd "
-#~ "party SMTP server."
-
-#~ msgid "SMTP port number"
-#~ msgstr "SMTP port number"
-
-#~ msgid "SMTP server authentication"
-#~ msgstr "SMTP server authentication"
-
-#~ msgid "SMTP server hostname or IP"
-#~ msgstr "SMTP server hostname or IP"
-
-#~ msgid "SMTP user name"
-#~ msgstr "SMTP user name"
-
-#~ msgid "Timeout before sending callers to voicemail"
-#~ msgstr "Timeout before sending callers to voicemail"
-
-#~ msgid "Use TLS (secure connection)"
-#~ msgstr "Use TLS (secure connection)"
-
-#~ msgid ""
-#~ "When you enable voicemail, you will have the opportunity to specify email "
-#~ "addresses which receive the message. You must also set up an SMTP server "
-#~ "below."
-#~ msgstr ""
-#~ "When you enable voicemail, you will have the opportunity to specify email "
-#~ "addresses which receive the message. You must also set up an SMTP server "
-#~ "below."
diff --git a/applications/luci-app-pbx-voicemail/po/es/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/es/pbx-voicemail.po
deleted file mode 100644
index 384e169de1..0000000000
--- a/applications/luci-app-pbx-voicemail/po/es/pbx-voicemail.po
+++ /dev/null
@@ -1,106 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2012-08-23 22:44+0200\n"
-"Last-Translator: José Vicente <josevteg@gmail.com>\n"
-"Language-Team: none\n"
-"Language: es\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr "Dirección de correo electrónica que recibe Voicemail"
-
-msgid "Enable Voicemail"
-msgstr "Activar Voicemail"
-
-msgid "Global Voicemail Setup"
-msgstr "Configuración global de Voicemail"
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-"Configure un correo de voz global para esta PBX. Como el sistema está "
-"diseñado para sistemas integrados como routers, no tiene un almacenamiento "
-"local y debe enviarse por correo. Por este motivo debe configurar un "
-"servidor SMTP de correo saliente (como ISP, Google o el correo de Yahoo) y "
-"establecer una lista de direcciones que recibirán el correo de voz grabado."
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-"Para usar esta PBX para enviar correos con grabaciones de correos de voz "
-"tiene que configurar un servidor SMTP. Su ISP es posible que tenga uno. "
-"También puede usar otros como los de Google o Yahoo."
-
-msgid "Last Sent Voicemail Log"
-msgstr "Último registro de Voicemail enviado"
-
-msgid "Local Storage Directory"
-msgstr "Directorio local de almacenamiento"
-
-msgid "No"
-msgstr "No"
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr "Servidor de correo SMTP saliente"
-
-msgid "SMTP Password"
-msgstr "Contraseña SMTP"
-
-msgid "SMTP Port Number"
-msgstr "Número de puerto SMTP"
-
-msgid "SMTP Server Authentication"
-msgstr "Autentificación de servidor SMTP"
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr "Nombre del servidor SMTP o dirección IP"
-
-msgid "SMTP User Name"
-msgstr "Nombre de usuario SMTP"
-
-msgid "Secure Connection Using TLS"
-msgstr "Asegurar la conexión con TLS"
-
-msgid "Voicemail Setup"
-msgstr "Configuración de Voicemail"
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-"Cuando active Voicemail tendrá que especificar direcciones de correo que "
-"recibirán los correos grabados así como un servidor SMTP."
-
-msgid "Yes"
-msgstr "Sí"
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-"También puede guardar copias de los mensajes de Voicemail en el dispositivo. "
-"El camino se creará si no existe. Tenga cuidado de no sobrepasar el espacio "
-"disponible en dispositivos pequeños como routers."
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
-"Su contraseña SMTP real no se muestra. Se cambiará solo cuando cambie el "
-"valor en esta caja."
diff --git a/applications/luci-app-pbx-voicemail/po/fr/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/fr/pbx-voicemail.po
deleted file mode 100644
index 0ccaa502e6..0000000000
--- a/applications/luci-app-pbx-voicemail/po/fr/pbx-voicemail.po
+++ /dev/null
@@ -1,88 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n > 1);\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr ""
-
-msgid "Enable Voicemail"
-msgstr ""
-
-msgid "Global Voicemail Setup"
-msgstr ""
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-
-msgid "Last Sent Voicemail Log"
-msgstr ""
-
-msgid "Local Storage Directory"
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr ""
-
-msgid "SMTP Password"
-msgstr ""
-
-msgid "SMTP Port Number"
-msgstr ""
-
-msgid "SMTP Server Authentication"
-msgstr ""
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr ""
-
-msgid "SMTP User Name"
-msgstr ""
-
-msgid "Secure Connection Using TLS"
-msgstr ""
-
-msgid "Voicemail Setup"
-msgstr ""
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
diff --git a/applications/luci-app-pbx-voicemail/po/he/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/he/pbx-voicemail.po
deleted file mode 100644
index be25f0b591..0000000000
--- a/applications/luci-app-pbx-voicemail/po/he/pbx-voicemail.po
+++ /dev/null
@@ -1,88 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr ""
-
-msgid "Enable Voicemail"
-msgstr ""
-
-msgid "Global Voicemail Setup"
-msgstr ""
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-
-msgid "Last Sent Voicemail Log"
-msgstr ""
-
-msgid "Local Storage Directory"
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr ""
-
-msgid "SMTP Password"
-msgstr ""
-
-msgid "SMTP Port Number"
-msgstr ""
-
-msgid "SMTP Server Authentication"
-msgstr ""
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr ""
-
-msgid "SMTP User Name"
-msgstr ""
-
-msgid "Secure Connection Using TLS"
-msgstr ""
-
-msgid "Voicemail Setup"
-msgstr ""
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
diff --git a/applications/luci-app-pbx-voicemail/po/hu/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/hu/pbx-voicemail.po
deleted file mode 100644
index be25f0b591..0000000000
--- a/applications/luci-app-pbx-voicemail/po/hu/pbx-voicemail.po
+++ /dev/null
@@ -1,88 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr ""
-
-msgid "Enable Voicemail"
-msgstr ""
-
-msgid "Global Voicemail Setup"
-msgstr ""
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-
-msgid "Last Sent Voicemail Log"
-msgstr ""
-
-msgid "Local Storage Directory"
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr ""
-
-msgid "SMTP Password"
-msgstr ""
-
-msgid "SMTP Port Number"
-msgstr ""
-
-msgid "SMTP Server Authentication"
-msgstr ""
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr ""
-
-msgid "SMTP User Name"
-msgstr ""
-
-msgid "Secure Connection Using TLS"
-msgstr ""
-
-msgid "Voicemail Setup"
-msgstr ""
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
diff --git a/applications/luci-app-pbx-voicemail/po/it/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/it/pbx-voicemail.po
deleted file mode 100644
index fe8c4bf004..0000000000
--- a/applications/luci-app-pbx-voicemail/po/it/pbx-voicemail.po
+++ /dev/null
@@ -1,110 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2013-02-09 21:15+0200\n"
-"Last-Translator: Francesco <3gasas@gmail.com>\n"
-"Language-Team: none\n"
-"Language: it\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr "Indirizzi e-mail che ricevono Voicemail"
-
-msgid "Enable Voicemail"
-msgstr "Attiva Voicemail"
-
-msgid "Global Voicemail Setup"
-msgstr "Attiva Voicemail"
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-"Qui è possibile configurare un messaggio vocale globale per questo PBX. "
-"Poiché questo sistema è destinato a girare su sistemi embedded come router, "
-"non vi è alcuna memorizzazione locale di segreteria - deve essere inviato "
-"via e-mail. Pertanto, è necessario configurare un server di posta in uscita "
-"(SMTP) (ad esempio del vostro ISP, di Google, Server Yahoo  SMTP), e di "
-"fornire un elenco di indirizzi che ricevono posta vocale registrato."
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-"Affinché questo PBX possa inviare messaggi di posta elettronica contenenti "
-"le registrazioni vocali, è necessario impostare un server SMTP qui. Il tuo "
-"ISP in genere fornisce un server SMTP per tale scopo. È inoltre possibile "
-"impostare un terzo SMTP  come quello fornito da Google o Yahoo."
-
-msgid "Last Sent Voicemail Log"
-msgstr "Ultimo file di registro Voicemail inviato"
-
-msgid "Local Storage Directory"
-msgstr "Cartella di memorizzazione Locale"
-
-msgid "No"
-msgstr "No"
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr "Server posta in uscita (SMTP)"
-
-msgid "SMTP Password"
-msgstr "Password SMTP"
-
-msgid "SMTP Port Number"
-msgstr "Numero Porta SMTP"
-
-msgid "SMTP Server Authentication"
-msgstr "Autenticazione Server SMTP"
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr "Nome Host Server SMTP o Indirizzo IP"
-
-msgid "SMTP User Name"
-msgstr "Nome Utente SMTP"
-
-msgid "Secure Connection Using TLS"
-msgstr "Connessione Sicura utilizzando TLS"
-
-msgid "Voicemail Setup"
-msgstr "Impostazione Voicemail"
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-"Quando si attiva la segreteria telefonica, si avrà la possibilità di "
-"specificare gli indirizzi e-mail che ricevono i messaggi vocali registrati. "
-"È inoltre necessario impostare un server SMTP di seguito."
-
-msgid "Yes"
-msgstr "Sì"
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-"È inoltre possibile conservare copie dei messaggi vocali sul dispositivo che "
-"esegue il PBX. Il percorso specificato in questo campo viene creato se non "
-"esiste. Attenzione lo spazio è limitato sui dispositivi embedded come "
-"router, e abilitare questa opzione solo se si sa cosa si sta facendo."
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
-"La tua password SMTP reale non viene visualizzata per la vostra protezione. "
-"Verrà modificato solo quando si modifica il valore in questa casella."
diff --git a/applications/luci-app-pbx-voicemail/po/ja/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/ja/pbx-voicemail.po
deleted file mode 100644
index 19e26bffa8..0000000000
--- a/applications/luci-app-pbx-voicemail/po/ja/pbx-voicemail.po
+++ /dev/null
@@ -1,88 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=1; plural=0;\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr ""
-
-msgid "Enable Voicemail"
-msgstr ""
-
-msgid "Global Voicemail Setup"
-msgstr ""
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-
-msgid "Last Sent Voicemail Log"
-msgstr ""
-
-msgid "Local Storage Directory"
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr ""
-
-msgid "SMTP Password"
-msgstr ""
-
-msgid "SMTP Port Number"
-msgstr ""
-
-msgid "SMTP Server Authentication"
-msgstr ""
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr ""
-
-msgid "SMTP User Name"
-msgstr ""
-
-msgid "Secure Connection Using TLS"
-msgstr ""
-
-msgid "Voicemail Setup"
-msgstr ""
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
diff --git a/applications/luci-app-pbx-voicemail/po/ms/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/ms/pbx-voicemail.po
deleted file mode 100644
index bf633a6702..0000000000
--- a/applications/luci-app-pbx-voicemail/po/ms/pbx-voicemail.po
+++ /dev/null
@@ -1,87 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr ""
-
-msgid "Enable Voicemail"
-msgstr ""
-
-msgid "Global Voicemail Setup"
-msgstr ""
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-
-msgid "Last Sent Voicemail Log"
-msgstr ""
-
-msgid "Local Storage Directory"
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr ""
-
-msgid "SMTP Password"
-msgstr ""
-
-msgid "SMTP Port Number"
-msgstr ""
-
-msgid "SMTP Server Authentication"
-msgstr ""
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr ""
-
-msgid "SMTP User Name"
-msgstr ""
-
-msgid "Secure Connection Using TLS"
-msgstr ""
-
-msgid "Voicemail Setup"
-msgstr ""
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
diff --git a/applications/luci-app-pbx-voicemail/po/no/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/no/pbx-voicemail.po
deleted file mode 100644
index be25f0b591..0000000000
--- a/applications/luci-app-pbx-voicemail/po/no/pbx-voicemail.po
+++ /dev/null
@@ -1,88 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr ""
-
-msgid "Enable Voicemail"
-msgstr ""
-
-msgid "Global Voicemail Setup"
-msgstr ""
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-
-msgid "Last Sent Voicemail Log"
-msgstr ""
-
-msgid "Local Storage Directory"
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr ""
-
-msgid "SMTP Password"
-msgstr ""
-
-msgid "SMTP Port Number"
-msgstr ""
-
-msgid "SMTP Server Authentication"
-msgstr ""
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr ""
-
-msgid "SMTP User Name"
-msgstr ""
-
-msgid "Secure Connection Using TLS"
-msgstr ""
-
-msgid "Voicemail Setup"
-msgstr ""
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
diff --git a/applications/luci-app-pbx-voicemail/po/pl/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/pl/pbx-voicemail.po
deleted file mode 100644
index 3f07fe4ba4..0000000000
--- a/applications/luci-app-pbx-voicemail/po/pl/pbx-voicemail.po
+++ /dev/null
@@ -1,111 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2014-05-05 02:29+0200\n"
-"Last-Translator: piosl <sleczek.piotr@gmail.com>\n"
-"Language-Team: none\n"
-"Language: pl\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=3; plural=(n==1 ? 0 : n%10>=2 && n%10<=4 && (n%100<10 "
-"|| n%100>=20) ? 1 : 2);\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr "Adres e-mail do odbierania poczty głosowej"
-
-msgid "Enable Voicemail"
-msgstr "Włącz pocztę głosową"
-
-msgid "Global Voicemail Setup"
-msgstr "Ustawienia globalnej poczty głosowej"
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-"Tutaj można skonfigurować globalną pocztę głosową dla PBX. Ponieważ system "
-"ten jest przeznaczony do pracy w systemach wbudowanych, takich jak routery, "
-"nie ma możliwości lokalnego przechowywania poczty głosowej - wiadomości "
-"muszą być wysłane e-mailem. Z tego powodu należy skonfigurować serwer poczty "
-"wychodzącej (SMTP) i podać listę adresów, które będą otrzymywać nagrania z "
-"poczty głosowej"
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-"W celu wysyłało e-maili zawierających nagrania poczty głosowej przez PBX, "
-"należy skonfigurować serwer SMTP. Twój dostawca usług internetowych "
-"zazwyczaj dostarcza serwer SMTP. Można również skonfigurować serwer SMTP "
-"firm trzecich, takich jak Google lub Yahoo."
-
-msgid "Last Sent Voicemail Log"
-msgstr ""
-
-msgid "Local Storage Directory"
-msgstr "Lokalny katalog przechowywania"
-
-msgid "No"
-msgstr "Nie"
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr "Serwer poczty wychodzącej (SMTP)"
-
-msgid "SMTP Password"
-msgstr "Hasło SMTP"
-
-msgid "SMTP Port Number"
-msgstr "Numer portu SMTP"
-
-msgid "SMTP Server Authentication"
-msgstr "Uwierzytelnianie serwera SMTP"
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr "Nazwa hosta serwera SMTP lub adres IP"
-
-msgid "SMTP User Name"
-msgstr "Nazwa użytkownika SMTP"
-
-msgid "Secure Connection Using TLS"
-msgstr "Bezpieczne połączenie za pomocą protokołu TLS"
-
-msgid "Voicemail Setup"
-msgstr "Ustawienia Poczty głosowej"
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-"Po włączeniu poczty głosowej, będziesz miał szansę na podanie adresów "
-"e-mail, które będą otrzymywać nagrane wiadomości głosowe. Musisz również "
-"skonfigurować serwer SMTP."
-
-msgid "Yes"
-msgstr "Tak"
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-"Możesz też zachować kopie nagrań poczty głosowej na urządzeniu, na którym "
-"działa PBX. Ścieżka określona tutaj zostanie utworzona, jeśli nie istnieje. "
-"Pamiętaj o ograniczonym miejscu na urządzeniach takich jak routery i włącz "
-"tę opcję tylko jeśli wiesz co robisz."
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
-"Twoje prawdziwe hasło SMTP nie jest pokazane dla Twojej ochrony. Zostanie "
-"zmienione tylko jeśli zmienisz wartość w tym polu."
diff --git a/applications/luci-app-pbx-voicemail/po/pt-br/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/pt-br/pbx-voicemail.po
deleted file mode 100644
index 2e3a51e7f9..0000000000
--- a/applications/luci-app-pbx-voicemail/po/pt-br/pbx-voicemail.po
+++ /dev/null
@@ -1,175 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2012-09-16 02:32+0200\n"
-"Last-Translator: Julio Cezar <jsilvestree@gmail.com>\n"
-"Language-Team: none\n"
-"Language: pt_BR\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n > 1);\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr "Endereços de correio eletrônicos que Recebem Correio de Voz"
-
-msgid "Enable Voicemail"
-msgstr "Habilitar o Correio de Voz"
-
-msgid "Global Voicemail Setup"
-msgstr "Configuração Global do Correio de Voz"
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-"Aqui você pode configurar um correio de voz global para este PBX. Uma vez "
-"que este sistema é previsto para ser executado em ambientes embarcados como "
-"roteadores, não existe armazenamento local do correio de voz - ele deve ser "
-"enviado por correio eletrônico. Desta maneira, você deve configurar um "
-"servidor (SMTP) de correio eletrônico (por exemplo, o servidor SMTP do seu "
-"provedor de Internet, do Google ou do Yahool), e fornecer uma lista dos "
-"endereços para os quais o correio de voz será enviado."
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-"Para este PBX enviar mensagens eletrônicas contendo as gravações do correio "
-"de voz, você precisa definir aqui um servidor SMTP. Seu provedor de Internet "
-"geralmente fornece um servidor SMTP para este propósito. Você também pode "
-"usar um servidor de terceiros como os fornecidos pelo GMail ou Yahoo."
-
-msgid "Last Sent Voicemail Log"
-msgstr "Registro do Último Correio de Voz Enviado"
-
-msgid "Local Storage Directory"
-msgstr "Diretório de Armazenamento Local"
-
-msgid "No"
-msgstr "Não"
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr "Servidor de correio eletrônico (SMTP) para envio"
-
-msgid "SMTP Password"
-msgstr "Senha do SMTP"
-
-msgid "SMTP Port Number"
-msgstr "Porta do SMTP"
-
-msgid "SMTP Server Authentication"
-msgstr "Autenticação do Servidor SMTP"
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr "Nome do Equipamento ou Endereço IP do Servidor SMTP"
-
-msgid "SMTP User Name"
-msgstr "Nome do Usuário do SMTP"
-
-msgid "Secure Connection Using TLS"
-msgstr "Proteja a Conexão Usando TLS"
-
-msgid "Voicemail Setup"
-msgstr "Configuração do Correio de Voz"
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-"Quando você habilita o correio de voz, você terá a oportunidade de "
-"especificar endereços de correio eletrônio que recebem o correio de voz "
-"gravado. Você precisa também configurar um servidor SMTP abaixo."
-
-msgid "Yes"
-msgstr "Sim"
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-"Você pode também manter cópias das mensagens de correio de voz no "
-"dispositivo executando o seu PBX. O caminho especificado aqui será criado se "
-"ele não existe. Cuidado com espaço limitado em dispositivos embarcados, como "
-"roteadores, e habilite esta opção apenas se você sabe o que você está "
-"fazendo."
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
-"Sua senha real do SMTP não é mostrada para a sua proteção. Ela será alterada "
-"apenas quando você modificar o valor neste campo."
-
-#~ msgid "Directory to save voicemail into"
-#~ msgstr "Diretório para salvar o correio de voz"
-
-#~ msgid "Email addresses to forward to"
-#~ msgstr "Endereços de correio eletrônicos para encaminhar"
-
-#~ msgid "Enabled"
-#~ msgstr "Habilitado"
-
-#~ msgid ""
-#~ "Here you can configure a global voicemail for this PBX. Since this system "
-#~ "is intended to run on embedded systems like routers, there is no local "
-#~ "storage of voicemail - it must be sent out by email. Therefore you need "
-#~ "to configure an outgoing mail (SMTP) server (for example the SMTP server "
-#~ "your ISP provides, or GMail), and provide a list of addresses the "
-#~ "voicemail will be sent to."
-#~ msgstr ""
-#~ "Aqui você pode configurar um correio de voz global para este PBX. Uma vez "
-#~ "que este sistema é previsto para ser executado em ambientes embarcados "
-#~ "como roteadores, não existe armazenamento local do correio de voz - ele "
-#~ "deve ser enviado por correio eletrônico. Desta maneira, você deve "
-#~ "configurar um servidor (SMTP) de correio eletrônico (por exemplo, o "
-#~ "servidor SMTP do seu provedor de Internet, ou o do GMail), e fornecer uma "
-#~ "lista dos endereços para os quais o correio de voz será enviado."
-
-#~ msgid ""
-#~ "In order for this PBX to send emails containing voicemail recordings, you "
-#~ "need to set up an SMTP server here. Your ISP usually provides an SMTP "
-#~ "server for that purpose. You can also set up a GMail, Yahoo, or other 3rd "
-#~ "party SMTP server."
-#~ msgstr ""
-#~ "Para este PBX enviar mensagens eletrônicas contendo as gravações do "
-#~ "correio de voz, você precisa definir aqui um servidor SMTP. Seu provedor "
-#~ "de Internet geralmente fornece um servidor SMTP para este propósito. Você "
-#~ "também pode usar o servidor SMTP do GMail, Yahoo, ou outro de terceiros."
-
-#~ msgid "SMTP port number"
-#~ msgstr "Número da porta do SMTP"
-
-#~ msgid "SMTP server authentication"
-#~ msgstr "Autenticação do servidor SMTP"
-
-#~ msgid "SMTP server hostname or IP"
-#~ msgstr "Nome do equipamento ou endereço IP do servidor SMTP"
-
-#~ msgid "SMTP user name"
-#~ msgstr "Nome do usuário do SMTP"
-
-#~ msgid "Timeout before sending callers to voicemail"
-#~ msgstr "Tempo de espera antes de enviar chamadas para correio de voz"
-
-#~ msgid "Use TLS (secure connection)"
-#~ msgstr "Usar TLS (conexão segura)"
-
-#~ msgid ""
-#~ "When you enable voicemail, you will have the opportunity to specify email "
-#~ "addresses which receive the message. You must also set up an SMTP server "
-#~ "below."
-#~ msgstr ""
-#~ "Quando você habilita o correio de voz, você terá a oportunidade de "
-#~ "especificar endereços de correio eletrônio que recebem a mensagem. Você "
-#~ "também deve configurar um servidor SMTP abaixo."
diff --git a/applications/luci-app-pbx-voicemail/po/pt/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/pt/pbx-voicemail.po
deleted file mode 100644
index f5cfd89e49..0000000000
--- a/applications/luci-app-pbx-voicemail/po/pt/pbx-voicemail.po
+++ /dev/null
@@ -1,91 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2013-06-03 18:28+0200\n"
-"Last-Translator: joao.f.vieira <joao.f.vieira@gmail.com>\n"
-"Language-Team: none\n"
-"Language: pt\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr ""
-
-msgid "Enable Voicemail"
-msgstr "Ativar Voicemail"
-
-msgid "Global Voicemail Setup"
-msgstr ""
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-
-msgid "Last Sent Voicemail Log"
-msgstr ""
-
-msgid "Local Storage Directory"
-msgstr ""
-
-msgid "No"
-msgstr "Não"
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr "Servidor de Envio de mail (SMTP)"
-
-msgid "SMTP Password"
-msgstr "Password SMTP"
-
-msgid "SMTP Port Number"
-msgstr "Porta SMTP"
-
-msgid "SMTP Server Authentication"
-msgstr "Servidor de Autenticação SMTP"
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr "Nome ou Endereço IP do Servidor SMTP"
-
-msgid "SMTP User Name"
-msgstr "Utilizador SMTP"
-
-msgid "Secure Connection Using TLS"
-msgstr "Ligação Segura usando TLS"
-
-msgid "Voicemail Setup"
-msgstr "Configuração do Voicemail"
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-
-msgid "Yes"
-msgstr "Sim"
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
diff --git a/applications/luci-app-pbx-voicemail/po/ro/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/ro/pbx-voicemail.po
deleted file mode 100644
index 42b66f6238..0000000000
--- a/applications/luci-app-pbx-voicemail/po/ro/pbx-voicemail.po
+++ /dev/null
@@ -1,89 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=3; plural=n==1 ? 0 : (n==0 || (n%100 > 0 && n%100 < "
-"20)) ? 1 : 2;\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr ""
-
-msgid "Enable Voicemail"
-msgstr ""
-
-msgid "Global Voicemail Setup"
-msgstr ""
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-
-msgid "Last Sent Voicemail Log"
-msgstr ""
-
-msgid "Local Storage Directory"
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr ""
-
-msgid "SMTP Password"
-msgstr ""
-
-msgid "SMTP Port Number"
-msgstr ""
-
-msgid "SMTP Server Authentication"
-msgstr ""
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr ""
-
-msgid "SMTP User Name"
-msgstr ""
-
-msgid "Secure Connection Using TLS"
-msgstr ""
-
-msgid "Voicemail Setup"
-msgstr ""
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
diff --git a/applications/luci-app-pbx-voicemail/po/ru/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/ru/pbx-voicemail.po
deleted file mode 100644
index ae49c6dd8f..0000000000
--- a/applications/luci-app-pbx-voicemail/po/ru/pbx-voicemail.po
+++ /dev/null
@@ -1,174 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: LuCI: pbx-voicemail\n"
-"POT-Creation-Date: \n"
-"PO-Revision-Date: 2012-08-15 17:42+0300\n"
-"Last-Translator: Roman A. aka BasicXP <x12ozmouse@ya.ru>\n"
-"Language-Team: Russian <x12ozmouse@ya.ru>\n"
-"Language: ru\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=3; plural=(n%10==1 && n%100!=11 ? 0 : n%10>=2 && n%"
-"10<=4 && (n%100<10 || n%100>=20) ? 1 : 2);\n"
-"X-Generator: Pootle 2.0.4\n"
-"X-Poedit-SourceCharset: UTF-8\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr "Адрес эл. почты для получения голосовой почты"
-
-msgid "Enable Voicemail"
-msgstr "Включить голосовую почту"
-
-msgid "Global Voicemail Setup"
-msgstr "Глобальные настройки голосовой почты"
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-"Здесь Вы можете настроить голосовую почту АТС. Так как данная система "
-"является встраиваемой и предназначена для таких устройств как, например, "
-"маршрутизаторы, локальное хранилище голосовой почты отсутствует. Голосовая "
-"почта пересылается через электронную почту. Следовательно, вам нужно указать "
-"сервер исходящей почты (SMTP) и перечислить адреса, на которые будет "
-"пересылаться голосовая почта."
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-"Чтобы отсылать электронную почту с записями голосовой почты, вам необходимо "
-"указать SMTP-сервер. Вы можете использовать SMTP-сервер вашего интернет-"
-"провайдера или любой другой, например, SMTP-сервер Google или Yahoo."
-
-msgid "Last Sent Voicemail Log"
-msgstr "Запись журнала последнего отправленного сообщения голосовой почты"
-
-msgid "Local Storage Directory"
-msgstr "Локальная директория хранения"
-
-msgid "No"
-msgstr "Нет"
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr "Сервер исходящей почты (SMTP)"
-
-msgid "SMTP Password"
-msgstr "Пароль SMTP"
-
-msgid "SMTP Port Number"
-msgstr "Номер порта SMTP"
-
-msgid "SMTP Server Authentication"
-msgstr "Аутентификация SMTP-сервера"
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr "Имя SMTP-сервера или IP-адрес"
-
-msgid "SMTP User Name"
-msgstr "Имя пользователя SMTP"
-
-msgid "Secure Connection Using TLS"
-msgstr "Защищенное соединение с использованием TLS"
-
-msgid "Voicemail Setup"
-msgstr "Настройка голосовой почты"
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-"При включении голосовой почты, у вас будет возможность указать адреса "
-"электронной почты, на которые будут отправляться записи голосовой почты. Вы "
-"также должны указать SMTP-сервер ниже."
-
-msgid "Yes"
-msgstr "Да"
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-"Вы также можете сохранять копии сообщений голосовой почты локально на "
-"устройстве с запущенной АТС. Указанный здесь путь будет создан в случае его "
-"отсутствия. Учитывайте, что пространство для хранения сообщений голосовой "
-"почты может быть ограничено вашим устройством."
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
-"Ваш настоящий пароль SMTP здесь не показан. Он будет изменён только тогда, "
-"когда вы измените значение в этом поле."
-
-#~ msgid "Directory to save voicemail into"
-#~ msgstr "Директория для сохранения голосовой почты"
-
-#~ msgid "Email addresses to forward to"
-#~ msgstr "Адрес эл. почты для перенаправления"
-
-#~ msgid "Enabled"
-#~ msgstr "Включено"
-
-#~ msgid ""
-#~ "Here you can configure a global voicemail for this PBX. Since this system "
-#~ "is intended to run on embedded systems like routers, there is no local "
-#~ "storage of voicemail - it must be sent out by email. Therefore you need "
-#~ "to configure an outgoing mail (SMTP) server (for example the SMTP server "
-#~ "your ISP provides, or GMail), and provide a list of addresses the "
-#~ "voicemail will be sent to."
-#~ msgstr ""
-#~ "Здесь Вы можете настроить голосовую почту АТС. Так как данная система "
-#~ "является встраиваемой и предназначена для таких устройств как, например, "
-#~ "маршрутизаторы, локальное хранилище голосовой почты отсутствует. "
-#~ "Голосовая почта пересылается через электронную почту. Следовательно, Вам "
-#~ "нужно указать сервер (SMTP) исходящей почты и перечислить адреса на "
-#~ "которые будет пересылаться голосовая почта."
-
-#~ msgid ""
-#~ "In order for this PBX to send emails containing voicemail recordings, you "
-#~ "need to set up an SMTP server here. Your ISP usually provides an SMTP "
-#~ "server for that purpose. You can also set up a GMail, Yahoo, or other 3rd "
-#~ "party SMTP server."
-#~ msgstr ""
-#~ "Чтобы отсылать электронную почту с записями голосовой почты, Вам "
-#~ "необходимо указать SMTP сервер. Вы можете использовать SMTP сервер вашего "
-#~ "интернет провайдера или любой другой, например, SMTP сервер GMail или "
-#~ "Yahoo."
-
-#~ msgid "SMTP port number"
-#~ msgstr "Номер порта SMTP"
-
-#~ msgid "SMTP server authentication"
-#~ msgstr "Аутентификация SMTP сервера"
-
-#~ msgid "SMTP server hostname or IP"
-#~ msgstr "Имя SMTP сервера или IP адрес"
-
-#~ msgid "SMTP user name"
-#~ msgstr "Имя пользователя SMTP"
-
-#~ msgid "Timeout before sending callers to voicemail"
-#~ msgstr "Таймаут перед перенаправлением звонящих на голосовую почту"
-
-#~ msgid "Use TLS (secure connection)"
-#~ msgstr "Использовать TLS (защищенное соединение)"
-
-#~ msgid ""
-#~ "When you enable voicemail, you will have the opportunity to specify email "
-#~ "addresses which receive the message. You must also set up an SMTP server "
-#~ "below."
-#~ msgstr ""
-#~ "При включении голосовой почты, у Вас будет возможность указать адреса "
-#~ "электронной почты на которые будут отправляться записи голосовой почты. "
-#~ "Вы также должны указать SMTP сервер ниже."
diff --git a/applications/luci-app-pbx-voicemail/po/sk/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/sk/pbx-voicemail.po
deleted file mode 100644
index c3a5c5ff92..0000000000
--- a/applications/luci-app-pbx-voicemail/po/sk/pbx-voicemail.po
+++ /dev/null
@@ -1,88 +0,0 @@
-msgid ""
-msgstr ""
-"Content-Type: text/plain; charset=UTF-8\n"
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=3; plural=(n==1) ? 0 : (n>=2 && n<=4) ? 1 : 2;\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr ""
-
-msgid "Enable Voicemail"
-msgstr ""
-
-msgid "Global Voicemail Setup"
-msgstr ""
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-
-msgid "Last Sent Voicemail Log"
-msgstr ""
-
-msgid "Local Storage Directory"
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr ""
-
-msgid "SMTP Password"
-msgstr ""
-
-msgid "SMTP Port Number"
-msgstr ""
-
-msgid "SMTP Server Authentication"
-msgstr ""
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr ""
-
-msgid "SMTP User Name"
-msgstr ""
-
-msgid "Secure Connection Using TLS"
-msgstr ""
-
-msgid "Voicemail Setup"
-msgstr ""
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
diff --git a/applications/luci-app-pbx-voicemail/po/sv/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/sv/pbx-voicemail.po
deleted file mode 100644
index 4945538752..0000000000
--- a/applications/luci-app-pbx-voicemail/po/sv/pbx-voicemail.po
+++ /dev/null
@@ -1,89 +0,0 @@
-msgid ""
-msgstr ""
-"Content-Type: text/plain; charset=UTF-8\n"
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"Language: sv\n"
-"MIME-Version: 1.0\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr "E-postadresser som ska ta emot röstbrev"
-
-msgid "Enable Voicemail"
-msgstr "Aktivera röstbrevlåda"
-
-msgid "Global Voicemail Setup"
-msgstr ""
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-
-msgid "Last Sent Voicemail Log"
-msgstr ""
-
-msgid "Local Storage Directory"
-msgstr "Lokal lagringsmapp"
-
-msgid "No"
-msgstr "Nej"
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr "Utgående mail (SMTP)-server"
-
-msgid "SMTP Password"
-msgstr "SMTP-lösenord"
-
-msgid "SMTP Port Number"
-msgstr "SMTP-portnummer"
-
-msgid "SMTP Server Authentication"
-msgstr "Autentisering för SMTP-server"
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr "SMTP-servern värdnamn eller IP-adress"
-
-msgid "SMTP User Name"
-msgstr "Användarnamn för SMTP"
-
-msgid "Secure Connection Using TLS"
-msgstr "Säker anslutning med användning av TLS"
-
-msgid "Voicemail Setup"
-msgstr ""
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-
-msgid "Yes"
-msgstr "Ja"
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
diff --git a/applications/luci-app-pbx-voicemail/po/templates/pbx-voicemail.pot b/applications/luci-app-pbx-voicemail/po/templates/pbx-voicemail.pot
deleted file mode 100644
index 35cdca33ac..0000000000
--- a/applications/luci-app-pbx-voicemail/po/templates/pbx-voicemail.pot
+++ /dev/null
@@ -1,81 +0,0 @@
-msgid ""
-msgstr "Content-Type: text/plain; charset=UTF-8"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr ""
-
-msgid "Enable Voicemail"
-msgstr ""
-
-msgid "Global Voicemail Setup"
-msgstr ""
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-
-msgid "Last Sent Voicemail Log"
-msgstr ""
-
-msgid "Local Storage Directory"
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr ""
-
-msgid "SMTP Password"
-msgstr ""
-
-msgid "SMTP Port Number"
-msgstr ""
-
-msgid "SMTP Server Authentication"
-msgstr ""
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr ""
-
-msgid "SMTP User Name"
-msgstr ""
-
-msgid "Secure Connection Using TLS"
-msgstr ""
-
-msgid "Voicemail Setup"
-msgstr ""
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
diff --git a/applications/luci-app-pbx-voicemail/po/tr/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/tr/pbx-voicemail.po
deleted file mode 100644
index 19e26bffa8..0000000000
--- a/applications/luci-app-pbx-voicemail/po/tr/pbx-voicemail.po
+++ /dev/null
@@ -1,88 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=1; plural=0;\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr ""
-
-msgid "Enable Voicemail"
-msgstr ""
-
-msgid "Global Voicemail Setup"
-msgstr ""
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-
-msgid "Last Sent Voicemail Log"
-msgstr ""
-
-msgid "Local Storage Directory"
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr ""
-
-msgid "SMTP Password"
-msgstr ""
-
-msgid "SMTP Port Number"
-msgstr ""
-
-msgid "SMTP Server Authentication"
-msgstr ""
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr ""
-
-msgid "SMTP User Name"
-msgstr ""
-
-msgid "Secure Connection Using TLS"
-msgstr ""
-
-msgid "Voicemail Setup"
-msgstr ""
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
diff --git a/applications/luci-app-pbx-voicemail/po/uk/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/uk/pbx-voicemail.po
deleted file mode 100644
index 565418376a..0000000000
--- a/applications/luci-app-pbx-voicemail/po/uk/pbx-voicemail.po
+++ /dev/null
@@ -1,89 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=3; plural=(n%10==1 && n%100!=11 ? 0 : n%10>=2 && n"
-"%10<=4 && (n%100<10 || n%100>=20) ? 1 : 2);\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr ""
-
-msgid "Enable Voicemail"
-msgstr ""
-
-msgid "Global Voicemail Setup"
-msgstr ""
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-
-msgid "Last Sent Voicemail Log"
-msgstr ""
-
-msgid "Local Storage Directory"
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr ""
-
-msgid "SMTP Password"
-msgstr ""
-
-msgid "SMTP Port Number"
-msgstr ""
-
-msgid "SMTP Server Authentication"
-msgstr ""
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr ""
-
-msgid "SMTP User Name"
-msgstr ""
-
-msgid "Secure Connection Using TLS"
-msgstr ""
-
-msgid "Voicemail Setup"
-msgstr ""
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
diff --git a/applications/luci-app-pbx-voicemail/po/vi/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/vi/pbx-voicemail.po
deleted file mode 100644
index 19e26bffa8..0000000000
--- a/applications/luci-app-pbx-voicemail/po/vi/pbx-voicemail.po
+++ /dev/null
@@ -1,88 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=1; plural=0;\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr ""
-
-msgid "Enable Voicemail"
-msgstr ""
-
-msgid "Global Voicemail Setup"
-msgstr ""
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-
-msgid "Last Sent Voicemail Log"
-msgstr ""
-
-msgid "Local Storage Directory"
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr ""
-
-msgid "SMTP Password"
-msgstr ""
-
-msgid "SMTP Port Number"
-msgstr ""
-
-msgid "SMTP Server Authentication"
-msgstr ""
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr ""
-
-msgid "SMTP User Name"
-msgstr ""
-
-msgid "Secure Connection Using TLS"
-msgstr ""
-
-msgid "Voicemail Setup"
-msgstr ""
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
diff --git a/applications/luci-app-pbx-voicemail/po/zh-cn/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/zh-cn/pbx-voicemail.po
deleted file mode 100644
index 5545c05fc0..0000000000
--- a/applications/luci-app-pbx-voicemail/po/zh-cn/pbx-voicemail.po
+++ /dev/null
@@ -1,107 +0,0 @@
-#
-# Yangfl <mmyangfl@gmail.com>, 2017.
-#
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2017-10-28 16:41+0800\n"
-"Last-Translator: Yangfl <mmyangfl@gmail.com>\n"
-"Language-Team:  <debian-l10n-chinese@lists.debian.org>\n"
-"Language: zh_CN\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=1; plural=0;\n"
-"X-Generator: Gtranslator 2.91.7\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr "接收语音邮箱的电子邮箱地址"
-
-msgid "Enable Voicemail"
-msgstr "启用语音邮箱"
-
-msgid "Global Voicemail Setup"
-msgstr "全局语音邮箱设置"
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-"在这里，您可以为 PBX 配置一个全球性的语音邮件。由于这个系统运行在嵌入式系统"
-"中，如路由器，这里并无本地语言邮件的储存空间 - 它必须通过电子邮件发送出去。因"
-"此，您需要配置一个外发邮件（SMTP）服务器（例如您的 ISP、谷歌或雅虎的 SMTP 服"
-"务器），并提供接收记录语音信箱的地址的列表。"
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-"为了这个 PBX 发送包含语音信箱录音的电子邮件，您需要在这里设置一个 SMTP 服务"
-"器。您的 ISP 通常会提供一个 SMTP 服务器。您也可以设立一个第三方的 SMTP 服务"
-"器，如谷歌或雅虎那样。"
-
-msgid "Last Sent Voicemail Log"
-msgstr "上一次发送语音信箱的日志"
-
-msgid "Local Storage Directory"
-msgstr "本地存储目录"
-
-msgid "No"
-msgstr "不"
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr "电子邮件发送服务器（SMTP）"
-
-msgid "SMTP Password"
-msgstr "SMTP 登录密码"
-
-msgid "SMTP Port Number"
-msgstr "SMTP 端口"
-
-msgid "SMTP Server Authentication"
-msgstr "SMTP 服务器认证"
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr "SMTP 服务器主机名或 IP 地址"
-
-msgid "SMTP User Name"
-msgstr "SMTP 用户名"
-
-msgid "Secure Connection Using TLS"
-msgstr "使用 TLS 安全连接"
-
-msgid "Voicemail Setup"
-msgstr "语音邮箱设置"
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-"当您启用了语音信箱，您将可以指定接收记录语音信箱的电子邮件地址。您也必须在下"
-"面设置 SMTP 服务器。"
-
-msgid "Yes"
-msgstr "是"
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-"您也可以在您运行 PBX 的设备上保留语音信箱留言的副本。此处指定的路径当不存在"
-"时，将会创建。谨防嵌入式设备上有限的存取空间，如路由器，所以此选项只有当您确"
-"定用途时才可使用。"
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
-"为了保护您，真正的 SMTP 密码将不会显示。只有当您改变框中的数值时它才会变更。"
diff --git a/applications/luci-app-pbx-voicemail/po/zh-tw/pbx-voicemail.po b/applications/luci-app-pbx-voicemail/po/zh-tw/pbx-voicemail.po
deleted file mode 100644
index 8464289997..0000000000
--- a/applications/luci-app-pbx-voicemail/po/zh-tw/pbx-voicemail.po
+++ /dev/null
@@ -1,107 +0,0 @@
-#
-# Yangfl <mmyangfl@gmail.com>, 2017.
-#
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2017-10-28 16:41+0800\n"
-"Last-Translator: Yangfl <mmyangfl@gmail.com>\n"
-"Language-Team:  <debian-l10n-chinese@lists.debian.org>\n"
-"Language: zh_TW\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=1; plural=0;\n"
-"X-Generator: Gtranslator 2.91.7\n"
-
-msgid "Email Addresses that Receive Voicemail"
-msgstr "接收語音郵箱的電子郵箱位址"
-
-msgid "Enable Voicemail"
-msgstr "啟用語音郵箱"
-
-msgid "Global Voicemail Setup"
-msgstr "全域性語音郵箱設定"
-
-msgid ""
-"Here you can configure a global voicemail for this PBX. Since this system is "
-"intended to run on embedded systems like routers, there is no local storage "
-"of voicemail - it must be sent out by email. Therefore you need to configure "
-"an outgoing mail (SMTP) server (for example your ISP's, Google's, or Yahoo's "
-"SMTP server), and provide a list of addresses that receive recorded "
-"voicemail."
-msgstr ""
-"在這裡，您可以為 PBX 配置一個全球性的語音郵件。由於這個系統執行在嵌入式系統"
-"中，如路由器，這裡並無本地語言郵件的儲存空間 - 它必須通過電子郵件傳送出去。因"
-"此，您需要配置一個外發郵件（SMTP）伺服器（例如您的 ISP、谷歌或雅虎的 SMTP 服"
-"務器），並提供接收記錄語音信箱的位址的列表。"
-
-msgid ""
-"In order for this PBX to send emails containing voicemail recordings, you "
-"need to set up an SMTP server here. Your ISP usually provides an SMTP server "
-"for that purpose. You can also set up a third party SMTP server such as the "
-"one provided by Google or Yahoo."
-msgstr ""
-"為了這個 PBX 傳送包含語音信箱錄音的電子郵件，您需要在這裡設定一個 SMTP 服務"
-"器。您的 ISP 通常會提供一個 SMTP 伺服器。您也可以設立一個第三方的 SMTP 服務"
-"器，如谷歌或雅虎那樣。"
-
-msgid "Last Sent Voicemail Log"
-msgstr "上一次傳送語音信箱的日誌"
-
-msgid "Local Storage Directory"
-msgstr "本地儲存目錄"
-
-msgid "No"
-msgstr "不"
-
-msgid "Outgoing mail (SMTP) Server"
-msgstr "電子郵件傳送伺服器（SMTP）"
-
-msgid "SMTP Password"
-msgstr "SMTP 登入密碼"
-
-msgid "SMTP Port Number"
-msgstr "SMTP 埠"
-
-msgid "SMTP Server Authentication"
-msgstr "SMTP 伺服器認證"
-
-msgid "SMTP Server Hostname or IP Address"
-msgstr "SMTP 伺服器主機名或 IP 位址"
-
-msgid "SMTP User Name"
-msgstr "SMTP 使用者名稱"
-
-msgid "Secure Connection Using TLS"
-msgstr "使用 TLS 安全連線"
-
-msgid "Voicemail Setup"
-msgstr "語音郵箱設定"
-
-msgid ""
-"When you enable voicemail, you will have the opportunity to specify email "
-"addresses that receive recorded voicemail. You must also set up an SMTP "
-"server below."
-msgstr ""
-"當您啟用了語音信箱，您將可以指定接收記錄語音信箱的電子郵件位址。您也必須在下"
-"面設定 SMTP 伺服器。"
-
-msgid "Yes"
-msgstr "是"
-
-msgid ""
-"You can also retain copies of voicemail messages on the device running your "
-"PBX. The path specified here will be created if it doesn't exist. Beware of "
-"limited space on embedded devices like routers, and enable this option only "
-"if you know what you are doing."
-msgstr ""
-"您也可以在您執行 PBX 的裝置上保留語音信箱留言的副本。此處指定的路徑當不存在"
-"時，將會建立。謹防嵌入式裝置上有限的存取空間，如路由器，所以此選項只有當您確"
-"定用途時才可使用。"
-
-msgid ""
-"Your real SMTP password is not shown for your protection. It will be changed "
-"only when you change the value in this box."
-msgstr ""
-"為了保護您，真正的 SMTP 密碼將不會顯示。只有當您改變框中的數值時它才會變更。"
diff --git a/applications/luci-app-pbx-voicemail/root/etc/config/pbx-voicemail b/applications/luci-app-pbx-voicemail/root/etc/config/pbx-voicemail
deleted file mode 100644
index 94e3e96ae8..0000000000
--- a/applications/luci-app-pbx-voicemail/root/etc/config/pbx-voicemail
+++ /dev/null
@@ -1,6 +0,0 @@
-config 'voicemail' 'global_voicemail'
-
-config 'voicemail' 'voicemail_smtp'
-
-config 'voicemail' 'voicemail_log'
-
diff --git a/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-move-greeting b/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-move-greeting
deleted file mode 100755
index 21fe69414b..0000000000
--- a/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-move-greeting
+++ /dev/null
@@ -1,6 +0,0 @@
-#!/bin/sh
-
-if [ -f "/tmp/voicemail/greeting.gsm" ]
-then
-    mv /tmp/voicemail/greeting.gsm /etc/pbx-voicemail/recordings/
-fi
diff --git a/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-msmtprc-account-auth.TEMPLATE b/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-msmtprc-account-auth.TEMPLATE
deleted file mode 100644
index 6b2026c108..0000000000
--- a/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-msmtprc-account-auth.TEMPLATE
+++ /dev/null
@@ -1,2 +0,0 @@
-user           |USER|
-password       |PASSWORD|
diff --git a/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-msmtprc-account-default.TEMPLATE b/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-msmtprc-account-default.TEMPLATE
deleted file mode 100644
index a001c64630..0000000000
--- a/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-msmtprc-account-default.TEMPLATE
+++ /dev/null
@@ -1,2 +0,0 @@
-account default : defaultacct
-
diff --git a/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-msmtprc-account.TEMPLATE b/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-msmtprc-account.TEMPLATE
deleted file mode 100644
index fd1f47959b..0000000000
--- a/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-msmtprc-account.TEMPLATE
+++ /dev/null
@@ -1,5 +0,0 @@
-account        defaultacct
-host           |HOST|
-port           |PORT|
-from           voicemail@pbx
-
diff --git a/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-msmtprc-defaults.TEMPLATE b/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-msmtprc-defaults.TEMPLATE
deleted file mode 100644
index a4456b8f87..0000000000
--- a/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-msmtprc-defaults.TEMPLATE
+++ /dev/null
@@ -1,5 +0,0 @@
-defaults
-auth             |AUTH|
-tls_certcheck    off
-tls              |TLS|
-
diff --git a/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-send-voicemail b/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-send-voicemail
deleted file mode 100755
index ba639d01a9..0000000000
--- a/applications/luci-app-pbx-voicemail/root/etc/pbx-voicemail/pbx-send-voicemail
+++ /dev/null
@@ -1,114 +0,0 @@
-#!/bin/sh
-#
-# Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com>
-#
-#    This file is part of luci-pbx-voicemail.
-#
-#    luci-pbx-voicemail is free software: you can redistribute it and/or modify
-#    it under the terms of the GNU General Public License as published by
-#    the Free Software Foundation, either version 3 of the License, or
-#    (at your option) any later version.
-#
-#    luci-pbx-voicemail is distributed in the hope that it will be useful,
-#    but WITHOUT ANY WARRANTY; without even the implied warranty of
-#    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-#    GNU General Public License for more details.
-#
-#    You should have received a copy of the GNU General Public License
-#    along with luci-pbx-voicemail.  If not, see <http://www.gnu.org/licenses/>.
-#
-#
-# Thanks to http://www.zedwood.com for providing an excellent example of how to
-# properly assemble an email message with a base64 encoded attachment.
-#
-
-LOGFILE=/tmp/voicemail/last_sent_voicemail.log
-
-# Redirect standard error and standard output to a log file.
-rm  -f "$LOGFILE"
-exec 1>"$LOGFILE"
-exec 2>&1
-
-# Appends its second argument to a file named in the first argument.
-append_to_file ()
-{
-    echo "$2">>$1;
-}
-
-# Grab the attachment name, which should be sent as the first argument, and
-# exit with a warning if there is no voicemail to send.
-ATTACHMENT="$1"
-[ ! -f "$ATTACHMENT" ] && echo "WARNING: Found no voicemail recording to send." && exit
-
-# Grab the callerID which should have been sent as an argument.
-CALLERID="$2"
-[ -z "$CALLERID" ]   && CALLERID="An unknown caller"
-
-# Determine addresses we would like to send the voicemail to and exit if none are found.
-TO="`uci -q get pbx-voicemail.global_voicemail.global_email_addresses | tr ' ' ','`"
-[ -z "$TO" ] && echo "WARNING: Found no addresses to send voicemail to." && exit
-
-# See whether we should retain a copy of the voicemail.
-SAVEPATH="`uci -q get pbx-voicemail.global_voicemail.global_save_path`"
-
-DATE="`date +%Y-%m-%d`"
-TIME="`date +%H:%M:%S`"
-FROM="voicemail@pbx"
-REPLY="do-not-reply@pbx"
-SUBJECT="Voicemail from $CALLERID, $DATE, $TIME"
-MSGBODY="$CALLERID has left voicemail for you on $DATE at $TIME."
-MIMETYPE="audio/wav"
-TMP1="/tmp/voicemail/tmpemail1.$$";
-TMP2="/tmp/voicemail/tmpemail2.$$";
-BOUNDARY="`date +%s | md5sum | awk '{print $1}'`"
-FILENAME="voicemail-$DATE-$TIME.WAV"
-
-# Clean up just in case.
-rm -f $TMP1 $TMP2
-
-append_to_file $TMP1 "From: $FROM"
-append_to_file $TMP1 "To: $TO"
-append_to_file $TMP1 "Reply-To: $REPLY"
-append_to_file $TMP1 "Subject: $SUBJECT"
-append_to_file $TMP1 "Content-Type: multipart/mixed; boundary=\""$BOUNDARY"\""
-append_to_file $TMP1 ""
-append_to_file $TMP1 "This is a MIME formatted message.  If you see this text it means that your"
-append_to_file $TMP1 "email software does not support MIME formatted messages."
-append_to_file $TMP1 ""
-append_to_file $TMP1 "--$BOUNDARY"
-append_to_file $TMP1 "Content-Type: text/plain; charset=ISO-8859-1; format=flowed"
-append_to_file $TMP1 "Content-Transfer-Encoding: 7bit"
-append_to_file $TMP1 "Content-Disposition: inline"
-append_to_file $TMP1 ""
-append_to_file $TMP1 "$MSGBODY"
-append_to_file $TMP1 ""
-append_to_file $TMP1 ""
-append_to_file $TMP1 "--$BOUNDARY"
-append_to_file $TMP1 "Content-Type: $MIMETYPE; name=\"$FILENAME\""
-append_to_file $TMP1 "Content-Transfer-Encoding: base64"
-append_to_file $TMP1 "Content-Disposition: attachment; filename=\"$FILENAME\";"
-append_to_file $TMP1 ""
-
-append_to_file $TMP2 ""
-append_to_file $TMP2 ""
-append_to_file $TMP2 "--$BOUNDARY--"
-append_to_file $TMP2 ""
-append_to_file $TMP2 ""
-
-# Cat everything together and pass to msmtprc to send out.
-( cat $TMP1
-  cat "$ATTACHMENT" | base64
-  cat $TMP2 ) | msmtp -t -C /etc/pbx-msmtprc
-
-# Clean up email temp files.
-rm -f $TMP1 $TMP2
-
-# Either delete or move the attachment based on the SAVEPATH variable.
-if [ -z "$SAVEPATH" ]
-then
-    rm -f "$ATTACHMENT"
-else
-    mkdir -p "$SAVEPATH"
-    mv --backup=t "$ATTACHMENT" "$SAVEPATH/$FILENAME"
-fi
-
diff --git a/applications/luci-app-pbx/COPYING b/applications/luci-app-pbx/COPYING
deleted file mode 100644
index 94a9ed024d..0000000000
--- a/applications/luci-app-pbx/COPYING
+++ /dev/null
@@ -1,674 +0,0 @@
-                    GNU GENERAL PUBLIC LICENSE
-                       Version 3, 29 June 2007
-
- Copyright (C) 2007 Free Software Foundation, Inc. <http://fsf.org/>
- Everyone is permitted to copy and distribute verbatim copies
- of this license document, but changing it is not allowed.
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diff --git a/applications/luci-app-pbx/CREDITS-SOUNDS b/applications/luci-app-pbx/CREDITS-SOUNDS
deleted file mode 100644
index 1fa64bc6cb..0000000000
--- a/applications/luci-app-pbx/CREDITS-SOUNDS
+++ /dev/null
@@ -1,7 +0,0 @@
-This file pertains to the sounds files included in root/etc/pbx-asterisk/sounds
-
-Recorded by:
-Allison Smith (http://www.theivrvoice.com)
-
-Financial Contributions by:
-Digium, Inc. (http://www.digium.com)
diff --git a/applications/luci-app-pbx/LICENSE-SOUNDS b/applications/luci-app-pbx/LICENSE-SOUNDS
deleted file mode 100644
index fe9c8221a2..0000000000
--- a/applications/luci-app-pbx/LICENSE-SOUNDS
+++ /dev/null
@@ -1,312 +0,0 @@
-This file pertains to the sounds files included in root/etc/pbx-asterisk/sounds
-
-LICENSE FOR VOICE PROMPT FILES
-------------------------------
-
-The voice prompt files distributed herewith are Copyright (C) 2003-2008
-Allison Smith, and provided under terms of the following License.  For
-more information, or to purchase custom voice prompt files, please
-visit:
-
-http://www.digium.com/ivr or http://www.theasteriskvoice.com
-
-LICENSE
--------
-
-THE WORK (AS DEFINED BELOW) IS PROVIDED UNDER THE TERMS OF THIS
-CREATIVE COMMONS PUBLIC LICENSE ("CCPL" OR "LICENSE"). THE WORK IS
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diff --git a/applications/luci-app-pbx/Makefile b/applications/luci-app-pbx/Makefile
deleted file mode 100644
index 772713b444..0000000000
--- a/applications/luci-app-pbx/Makefile
+++ /dev/null
@@ -1,19 +0,0 @@
-#
-# Copyright (C) 2008-2014 The LuCI Team <luci@lists.subsignal.org>
-#
-# This is free software, licensed under the Apache License, Version 2.0 .
-#
-
-include $(TOPDIR)/rules.mk
-
-LUCI_TITLE:=LuCI PBX Administration
-LUCI_DEPENDS:= @BROKEN \
-	+asterisk18 +asterisk18-app-authenticate +asterisk18-app-disa \
-	+asterisk18-app-setcallerid +asterisk18-app-system +asterisk18-chan-gtalk \
-	+asterisk18-codec-a-mu +asterisk18-codec-alaw +asterisk18-func-cut \
-	+asterisk18-res-clioriginate +asterisk18-func-channel +asterisk18-chan-local \
-	+asterisk18-app-record +asterisk18-app-senddtmf +asterisk18-res-crypto
-
-include ../../luci.mk
-
-# call BuildPackage - OpenWrt buildroot signature
diff --git a/applications/luci-app-pbx/luasrc/controller/pbx.lua b/applications/luci-app-pbx/luasrc/controller/pbx.lua
deleted file mode 100644
index b77814b150..0000000000
--- a/applications/luci-app-pbx/luasrc/controller/pbx.lua
+++ /dev/null
@@ -1,29 +0,0 @@
---[[
-    Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com>
-
-    This file is part of luci-pbx.
-
-    luci-pbx is free software: you can redistribute it and/or modify
-    it under the terms of the GNU General Public License as published by
-    the Free Software Foundation, either version 3 of the License, or
-    (at your option) any later version.
-
-    luci-pbx is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-    GNU General Public License for more details.
-
-    You should have received a copy of the GNU General Public License
-    along with luci-pbx.  If not, see <http://www.gnu.org/licenses/>.
-]]--
-
-module("luci.controller.pbx", package.seeall)
-
-function index()
-        entry({"admin", "services", "pbx"},                 cbi("pbx"),          "PBX",              80)
-        entry({"admin", "services", "pbx", "pbx-google"},   cbi("pbx-google"),   "Google Accounts",   1)
-        entry({"admin", "services", "pbx", "pbx-voip"},     cbi("pbx-voip"),     "SIP Accounts",      2)
-        entry({"admin", "services", "pbx", "pbx-users"},    cbi("pbx-users"),    "User Accounts",     3)
-        entry({"admin", "services", "pbx", "pbx-calls"},    cbi("pbx-calls"),    "Call Routing",      4)
-        entry({"admin", "services", "pbx", "pbx-advanced"}, cbi("pbx-advanced"), "Advanced Settings", 6)
-end
diff --git a/applications/luci-app-pbx/luasrc/model/cbi/pbx-advanced.lua b/applications/luci-app-pbx/luasrc/model/cbi/pbx-advanced.lua
deleted file mode 100644
index 34288c6632..0000000000
--- a/applications/luci-app-pbx/luasrc/model/cbi/pbx-advanced.lua
+++ /dev/null
@@ -1,293 +0,0 @@
---[[
-    Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com>
-
-    This file is part of luci-pbx.
-
-    luci-pbx is free software: you can redistribute it and/or modify
-    it under the terms of the GNU General Public License as published by
-    the Free Software Foundation, either version 3 of the License, or
-    (at your option) any later version.
-
-    luci-pbx is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-    GNU General Public License for more details.
-
-    You should have received a copy of the GNU General Public License
-    along with luci-pbx.  If not, see <http://www.gnu.org/licenses/>.
-]]--
-
-if     nixio.fs.access("/etc/init.d/asterisk")   then
-   server = "asterisk"
-elseif nixio.fs.access("/etc/init.d/freeswitch") then
-   server = "freeswitch"
-else
-   server = ""
-end
-
-appname         = "PBX"
-modulename      = "pbx-advanced"
-defaultbindport = 5060
-defaultrtpstart = 19850
-defaultrtpend   = 19900
-
--- Returns all the network related settings, including a constructed RTP range
-function get_network_info()
-        externhost = m.uci:get(modulename, "advanced", "externhost")
-        ipaddr     = m.uci:get("network", "lan", "ipaddr")
-        bindport   = m.uci:get(modulename, "advanced", "bindport")
-        rtpstart   = m.uci:get(modulename, "advanced", "rtpstart")
-        rtpend     = m.uci:get(modulename, "advanced", "rtpend")
-
-        if bindport == nil then bindport = defaultbindport end
-        if rtpstart == nil then rtpstart = defaultrtpstart end
-        if rtpend   == nil then rtpend   = defaultrtpend   end
-
-        if rtpstart == nil or rtpend == nil then
-                rtprange = nil
-        else
-                rtprange = rtpstart .. "-" .. rtpend
-        end
-
-        return bindport, rtprange, ipaddr, externhost
-end
-
--- If not present, insert empty rules in the given config & section named PBX-SIP and PBX-RTP
-function insert_empty_sip_rtp_rules(config, section)
-
-                -- Add rules named PBX-SIP and PBX-RTP if not existing
-                found_sip_rule = false
-                found_rtp_rule = false
-                m.uci:foreach(config, section,
-                        function(s1)
-                                if     s1._name == 'PBX-SIP' then
-                                        found_sip_rule = true
-                                elseif s1._name == 'PBX-RTP' then
-                                        found_rtp_rule = true
-                                end
-                        end)
-                
-                if found_sip_rule ~= true then
-                        newrule=m.uci:add(config, section)
-                        m.uci:set(config, newrule, '_name', 'PBX-SIP')
-                end
-                if found_rtp_rule ~= true then
-                        newrule=m.uci:add(config, section)
-                        m.uci:set(config, newrule, '_name', 'PBX-RTP')
-                end
-end
-
--- Delete rules in the given config & section named PBX-SIP and PBX-RTP
-function delete_sip_rtp_rules(config, section)
-
-                -- Remove rules named PBX-SIP and PBX-RTP
-                commit = false
-                m.uci:foreach(config, section,
-                        function(s1)
-                                if s1._name == 'PBX-SIP' or s1._name == 'PBX-RTP' then
-                                        m.uci:delete(config, s1['.name'])
-                                        commit = true
-                                end
-                        end)
-
-        -- If something changed, then we commit the config.
-        if commit == true then m.uci:commit(config) end
-end
-
--- Deletes QoS rules associated with this PBX.
-function delete_qos_rules()
-        delete_sip_rtp_rules ("qos", "classify")
-end
-
-
-function insert_qos_rules()
-        -- Insert empty PBX-SIP and PBX-RTP rules if not present.
-        insert_empty_sip_rtp_rules ("qos", "classify")
-
-        -- Get the network information
-        bindport, rtprange, ipaddr, externhost = get_network_info()
-
-        -- Iterate through the QoS rules, and if there is no other rule with the same port
-        -- range at the priority service level, insert this rule.
-        commit = false
-        m.uci:foreach("qos", "classify",
-                function(s1)
-                        if     s1._name == 'PBX-SIP' then
-                                if s1.ports ~= bindport or s1.target ~= "Priority" or s1.proto ~= "udp" then
-                                        m.uci:set("qos", s1['.name'], "ports",  bindport)
-                                        m.uci:set("qos", s1['.name'], "proto",  "udp")
-                                        m.uci:set("qos", s1['.name'], "target", "Priority")
-                                        commit = true
-                                end
-                        elseif s1._name == 'PBX-RTP' then
-                                if s1.ports ~= rtprange or s1.target ~= "Priority" or s1.proto ~= "udp" then
-                                        m.uci:set("qos", s1['.name'], "ports",  rtprange)
-                                        m.uci:set("qos", s1['.name'], "proto",  "udp")
-                                        m.uci:set("qos", s1['.name'], "target", "Priority")
-                                        commit = true
-                                end
-                        end
-                end)
-
-        -- If something changed, then we commit the qos config.
-        if commit == true then m.uci:commit("qos") end
-end
-
--- This function is a (so far) unsuccessful attempt to manipulate the firewall rules from here
--- Need to do more testing and eventually move to this mode.
-function maintain_firewall_rules()
-        -- Get the network information
-        bindport, rtprange, ipaddr, externhost = get_network_info()
-
-        commit = false
-        -- Only if externhost is set, do we control firewall rules.
-        if externhost ~= nil and bindport ~= nil and rtprange ~= nil then
-                -- Insert empty PBX-SIP and PBX-RTP rules if not present.
-                insert_empty_sip_rtp_rules ("firewall", "rule")
-
-                -- Iterate through the firewall rules, and if the dest_port and dest_ip setting of the\
-                -- SIP and RTP rule do not match what we want configured, set all the entries in the rule\
-                -- appropriately.
-                m.uci:foreach("firewall", "rule",
-                        function(s1)
-                                if     s1._name == 'PBX-SIP' then
-                                        if s1.dest_port ~= bindport then
-                                                m.uci:set("firewall", s1['.name'], "dest_port", bindport)
-                                                m.uci:set("firewall", s1['.name'], "src", "wan")
-                                                m.uci:set("firewall", s1['.name'], "proto", "udp")
-                                                m.uci:set("firewall", s1['.name'], "target", "ACCEPT")
-                                                commit = true
-                                        end
-                                elseif s1._name == 'PBX-RTP' then
-                                        if s1.dest_port ~= rtprange then
-                                                m.uci:set("firewall", s1['.name'], "dest_port", rtprange)
-                                                m.uci:set("firewall", s1['.name'], "src", "wan")
-                                                m.uci:set("firewall", s1['.name'], "proto", "udp")
-                                                m.uci:set("firewall", s1['.name'], "target", "ACCEPT")
-                                                commit = true
-                                        end
-                                end
-                        end)
-        else
-                -- We delete the firewall rules if one or more of the necessary parameters are not set.
-                sip_rule_name=nil
-                rtp_rule_name=nil
-        
-                -- First discover the configuration names of the rules.
-                m.uci:foreach("firewall", "rule",
-                        function(s1)
-                                if     s1._name == 'PBX-SIP' then
-                                        sip_rule_name = s1['.name']
-                                elseif s1._name == 'PBX-RTP' then
-                                        rtp_rule_name = s1['.name']
-                                end
-                        end)
-                
-                -- Then, using the names, actually delete the rules.
-                if sip_rule_name ~= nil then
-                        m.uci:delete("firewall", sip_rule_name)
-                        commit = true
-                end
-                if rtp_rule_name ~= nil then
-                        m.uci:delete("firewall", rtp_rule_name)
-                        commit = true
-                end
-        end
-
-        -- If something changed, then we commit the firewall config.
-        if commit == true then m.uci:commit("firewall") end
-end
-
-m = Map (modulename, translate("Advanced Settings"),
-         translate("This section contains settings that do not need to be changed under \
-         normal circumstances. In addition, here you can configure your system \
-         for use with remote SIP devices, and resolve call quality issues by enabling \
-         the insertion of QoS rules."))
-
--- Recreate the voip server config, and restart necessary services after changes are commited
--- to the advanced configuration. The firewall must restart because of "Remote Usage".
-function m.on_after_commit(self)
-
-        -- Make sure firewall rules are in place
-        maintain_firewall_rules()
-
-        -- If insertion of QoS rules is enabled
-        if m.uci:get(modulename, "advanced", "qos_enabled") == "yes" then
-                insert_qos_rules()
-        else
-                delete_qos_rules()
-        end
-
-        luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null")
-        luci.sys.call("/etc/init.d/"     .. server .. " restart 1\>/dev/null 2\>/dev/null")
-        luci.sys.call("/etc/init.d/firewall             restart 1\>/dev/null 2\>/dev/null")
-end
-
------------------------------------------------------------------------------
-s = m:section(NamedSection, "advanced", "settings", translate("Advanced Settings"))
-s.anonymous = true
-
-s:tab("general",  translate("General Settings"))
-s:tab("remote_usage", translate("Remote Usage"),
-      translatef("You can use your SIP devices/softphones with this system from a remote location \
-      as well, as long as your Internet Service Provider gives you a public IP. \
-      You will be able to call other local users for free (e.g. other Analog Telephone Adapters (ATAs)) \
-      and use your VoIP providers to make calls as if you were local to the PBX. \
-      After configuring this tab, go back to where users are configured and see the new \
-      Server and Port setting you need to configure the remote SIP devices with. Please note that if this \
-      PBX is not running on your router/gateway, you will need to configure port forwarding (NAT) on your \
-      router/gateway. Please forward the ports below (SIP port and RTP range) to the IP address of the \
-      device running this PBX."))
-
-s:tab("qos",  translate("QoS Settings"), 
-      translate("If you experience jittery or high latency audio during heavy downloads, you may want \
-      to enable QoS. QoS prioritizes traffic to and from your network for specified ports and IP \
-      addresses, resulting in better latency and throughput for sound in our case. If enabled below, \
-      a QoS rule for this service will be configured by the PBX automatically, but you must visit the \
-      QoS configuration page (Network->QoS) to configure other critical QoS settings like Download \
-      and Upload speed."))
-
-ringtime = s:taboption("general", Value, "ringtime", translate("Number of Seconds to Ring"),
-                 translate("Set the number of seconds to ring users upon incoming calls before hanging up \
-                 or going to voicemail, if the voicemail is installed and enabled."))
-ringtime.datatype = "port"
-ringtime.default = 30
-
-ua = s:taboption("general", Value, "useragent", translate("User Agent String"),
-                 translate("This is the name that the VoIP server will use to identify itself when \
-                 registering to VoIP (SIP) providers. Some providers require this to a specific \
-                 string matching a hardware SIP device."))
-ua.default = appname
-
-h = s:taboption("remote_usage", Value, "externhost", translate("Domain/IP Address/Dynamic Domain"),
-                translate("You can enter your domain name, external IP address, or dynamic domain name here. \
-                The best thing to input is a static IP address. If your IP address is dynamic and it changes, \
-                your configuration will become invalid. Hence, it's recommended to set up Dynamic DNS in this case. \
-                and enter your Dynamic DNS hostname here. You can configure Dynamic DNS with the luci-app-ddns package."))
-h.datatype = "host(0)"
-
-p = s:taboption("remote_usage", Value, "bindport", translate("External SIP Port"),
-                translate("Pick a random port number between 6500 and 9500 for the service to listen on. \
-                Do not pick the standard 5060, because it is often subject to brute-force attacks. \
-                When finished, (1) click \"Save and Apply\", and (2) look in the \
-                \"SIP Device/Softphone Accounts\" section for updated Server and Port settings \
-                for your SIP Devices/Softphones."))
-p.datatype = "port"
-
-p = s:taboption("remote_usage", Value, "rtpstart", translate("RTP Port Range Start"),
-                translate("RTP traffic carries actual voice packets. This is the start of the port range \
-                that will be used for setting up RTP communication. It's usually OK to leave this \
-                at the default value."))
-p.datatype = "port"
-p.default = defaultrtpstart
-
-p = s:taboption("remote_usage", Value, "rtpend", translate("RTP Port Range End"))
-p.datatype = "port"
-p.default = defaultrtpend
-
-p = s:taboption("qos", ListValue, "qos_enabled", translate("Insert QoS Rules"))
-p:value("yes", translate("Yes"))
-p:value("no",  translate("No"))
-p.default = "yes"
-
-return m
diff --git a/applications/luci-app-pbx/luasrc/model/cbi/pbx-calls.lua b/applications/luci-app-pbx/luasrc/model/cbi/pbx-calls.lua
deleted file mode 100644
index ca373d63a3..0000000000
--- a/applications/luci-app-pbx/luasrc/model/cbi/pbx-calls.lua
+++ /dev/null
@@ -1,424 +0,0 @@
---[[
-    Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com>
-
-    This file is part of luci-pbx.
-
-    luci-pbx is free software: you can redistribute it and/or modify
-    it under the terms of the GNU General Public License as published by
-    the Free Software Foundation, either version 3 of the License, or
-    (at your option) any later version.
-
-    luci-pbx is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-    GNU General Public License for more details.
-
-    You should have received a copy of the GNU General Public License
-    along with luci-pbx.  If not, see <http://www.gnu.org/licenses/>.
-]]--
-
-if     nixio.fs.access("/etc/init.d/asterisk")   then
-   server = "asterisk"
-elseif nixio.fs.access("/etc/init.d/freeswitch") then
-   server = "freeswitch"
-else
-   server = ""
-end
-
-modulename        = "pbx-calls"
-voipmodulename    = "pbx-voip"
-googlemodulename  = "pbx-google"
-usersmodulename   = "pbx-users"
-allvalidaccounts  = {}
-nallvalidaccounts = 0
-validoutaccounts  = {}
-nvalidoutaccounts = 0
-validinaccounts   = {}
-nvalidinaccounts  = 0
-allvalidusers     = {}
-nallvalidusers    = 0
-validoutusers     = {}
-nvalidoutusers    = 0
-
-
--- Checks whether the entered extension is valid syntactically.
-function is_valid_extension(exten)
-   return (exten:match("[#*+0-9NXZ]+$") ~= nil)
-end
-
-
-m = Map (modulename, translate("Call Routing"),
-         translate("This is where you indicate which Google/SIP accounts are used to call what \
-                   country/area codes, which users can use what SIP/Google accounts, how incoming \
-                   calls are routed, what numbers can get into this PBX with a password, and what \
-                   numbers are blacklisted."))
-
--- Recreate the config, and restart services after changes are commited to the configuration.
-function m.on_after_commit(self)
-        luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null")
-        luci.sys.call("/etc/init.d/"     .. server .. " restart 1\>/dev/null 2\>/dev/null")
-end
-
--- Add Google accounts to all valid accounts, and accounts valid for incoming and outgoing calls.
-m.uci:foreach(googlemodulename, "gtalk_jabber", 
-              function(s1)
-                 -- Add this provider to list of valid accounts.
-                 if s1.username ~= nil and s1.name ~= nil then
-                    allvalidaccounts[s1.name] = s1.username
-                    nallvalidaccounts = nallvalidaccounts + 1
-
-                    if s1.make_outgoing_calls == "yes" then
-                       -- Add provider to the associative array of valid outgoing accounts.
-                       validoutaccounts[s1.name] = s1.username
-                       nvalidoutaccounts = nvalidoutaccounts + 1
-                    end
-
-                    if s1.register == "yes" then
-                       -- Add provider to the associative array of valid outgoing accounts.
-                       validinaccounts[s1.name]  = s1.username
-                       nvalidinaccounts = nvalidinaccounts + 1
-                    end
-                 end
-              end)
-
--- Add SIP accounts to all valid accounts, and accounts valid for incoming and outgoing calls.
-m.uci:foreach(voipmodulename, "voip_provider", 
-              function(s1)
-                 -- Add this provider to list of valid accounts.
-                 if s1.defaultuser ~= nil and s1.host ~= nil and s1.name ~= nil then
-                    allvalidaccounts[s1.name] = s1.defaultuser .. "@" .. s1.host
-                    nallvalidaccounts = nallvalidaccounts + 1
-
-                    if s1.make_outgoing_calls == "yes" then
-                       -- Add provider to the associative array of valid outgoing accounts.
-                       validoutaccounts[s1.name] = s1.defaultuser .. "@" .. s1.host
-                       nvalidoutaccounts = nvalidoutaccounts + 1
-                    end
-
-                    if s1.register == "yes" then
-                       -- Add provider to the associative array of valid outgoing accounts.
-                       validinaccounts[s1.name]  = s1.defaultuser .. "@" .. s1.host
-                       nvalidinaccounts = nvalidinaccounts + 1
-                    end
-                 end
-              end)
-
--- Add Local User accounts to all valid users, and users allowed to make outgoing calls.
-m.uci:foreach(usersmodulename, "local_user",
-              function(s1)
-                 -- Add user to list of all valid users.
-                 if s1.defaultuser ~= nil then
-                    allvalidusers[s1.defaultuser] = true
-                    nallvalidusers = nallvalidusers + 1
-                    
-                    if s1.can_call == "yes" then
-                       validoutusers[s1.defaultuser] = true
-                       nvalidoutusers = nvalidoutusers + 1
-                    end
-                 end
-              end)
-
-
-----------------------------------------------------------------------------------------------------
--- If there are no accounts configured, or no accounts enabled for outgoing calls, display a warning.
--- Otherwise, display the usual help text within the section.
-if     nallvalidaccounts == 0 then
-   text = translate("NOTE: There are no Google or SIP provider accounts configured.")
-elseif nvalidoutaccounts == 0 then
-   text = translate("NOTE: There are no Google or SIP provider accounts enabled for outgoing calls.")
-else
-   text = translate("If you have more than one account that can make outgoing calls, you \
-   should enter a list of phone numbers and/or prefixes in the following fields for each \
-   provider listed. Invalid prefixes are removed silently, and only 0-9, X, Z, N, #, *, \
-   and + are valid characters. The letter X matches 0-9, Z matches 1-9, and N matches 2-9. \
-   For example to make calls to Germany through a provider, you can enter 49. To make calls \
-   to North America, you can enter 1NXXNXXXXXX. If one of your providers can make \"local\" \
-   calls to an area code like New York's 646, you can enter 646NXXXXXX for that \
-   provider. You should leave one account with an empty list to make calls with \
-   it by default, if no other provider's prefixes match. The system will automatically \
-   replace an empty list with a message that the provider dials all numbers not matched by another \
-   provider's prefixes. Be as specific as possible (i.e. 1NXXNXXXXXX is better than 1). Please note \
-   all international dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a \
-   space-separated list, and/or one per line by hitting enter after every one.")
-end
-
-
-s = m:section(NamedSection, "outgoing_calls", "call_routing", translate("Outgoing Calls"), text)
-s.anonymous = true
-
-for k,v in pairs(validoutaccounts) do
-   patterns = s:option(DynamicList, k, v)
-   
-   -- If the saved field is empty, we return a string
-   -- telling the user that this provider would dial any exten.
-   function patterns.cfgvalue(self, section)
-      value = self.map:get(section, self.option)
-      
-      if value == nil then
-         return {translate("Dials numbers unmatched elsewhere")}
-      else
-         return value
-      end
-   end
-   
-   -- Write only valid extensions into the config file.
-   function patterns.write(self, section, value)
-      newvalue = {}
-      nindex = 1
-      for index, field in ipairs(value) do
-         val = luci.util.trim(value[index])
-         if is_valid_extension(val) == true then
-            newvalue[nindex] = val
-            nindex = nindex + 1
-         end
-      end
-      DynamicList.write(self, section, newvalue)
-   end
-end
-
-----------------------------------------------------------------------------------------------------
--- If there are no accounts configured, or no accounts enabled for incoming calls, display a warning.
--- Otherwise, display the usual help text within the section.
-if     nallvalidaccounts == 0 then
-   text = translate("NOTE: There are no Google or SIP provider accounts configured.")
-elseif nvalidinaccounts == 0 then
-   text = translate("NOTE: There are no Google or SIP provider accounts enabled for incoming calls.")
-else
-   text = translate("For each provider enabled for incoming calls, here you can restrict which users to\
-                ring on incoming calls. If the list is empty, the system will indicate that all users \
-                enabled for incoming calls will ring. Invalid usernames will be rejected \
-                silently. Also, entering a username here overrides the user's setting to not receive \
-                incoming calls. This way, you can make certain users ring only for specific providers. \
-                Entries can be made in a space-separated list, and/or one per line by hitting enter after \
-                every one.")
-end
-
-
-s = m:section(NamedSection, "incoming_calls", "call_routing", translate("Incoming Calls"), text)
-s.anonymous = true
-
-for k,v in pairs(validinaccounts) do
-   users = s:option(DynamicList, k, v)
-   
-   -- If the saved field is empty, we return a string telling the user that
-   -- this provider would ring all users configured for incoming calls.
-   function users.cfgvalue(self, section)
-      value = self.map:get(section, self.option)
-      
-      if value == nil then
-         return {translate("Rings users enabled for incoming calls")}
-      else
-         return value
-      end
-   end
-   
-   -- Write only valid user names.
-   function users.write(self, section, value)
-      newvalue = {}
-      nindex = 1
-      for index, field in ipairs(value) do
-         trimuser = luci.util.trim(value[index])
-         if allvalidusers[trimuser] == true then
-            newvalue[nindex] = trimuser
-            nindex = nindex + 1
-         end
-      end
-      DynamicList.write(self, section, newvalue)
-   end
-end
-
-
-----------------------------------------------------------------------------------------------------
--- If there are no user accounts configured, no user accounts enabled for outgoing calls,
--- display a warning. Otherwise, display the usual help text within the section.
-if     nallvalidusers == 0 then
-   text = translate("NOTE: There are no local user accounts configured.")
-elseif nvalidoutusers == 0 then
-   text = translate("NOTE: There are no local user accounts enabled for outgoing calls.")
-else
-   text = translate("For each user enabled for outgoing calls you can restrict what providers the user \
-        can use for outgoing calls. By default all users can use all providers. To show up in the list \
-        below the user should be allowed to make outgoing calls in the \"User Accounts\" page. Enter VoIP \
-        providers in the format username@some.host.name, as listed in \"Outgoing Calls\" above. It's \
-        easiest to copy and paste the providers from above. Invalid entries, including providers not \
-        enabled for outgoing calls, will be rejected silently. Entries can be made in a space-separated \
-        list, and/or one per line by hitting enter after every one.")
-end
-
-
-s = m:section(NamedSection, "providers_user_can_use", "call_routing",
-     translate("Providers Used for Outgoing Calls"), text)
-s.anonymous = true
-
-for k,v in pairs(validoutusers) do
-   providers = s:option(DynamicList, k, k)
-
-   -- If the saved field is empty, we return a string telling the user
-   -- that this user uses all providers enavled for outgoing calls.
-   function providers.cfgvalue(self, section)
-      value = self.map:get(section, self.option)
-      
-      if value == nil then
-         return {translate("Uses providers enabled for outgoing calls")}
-      else
-         newvalue = {}
-         -- Convert internal names to user@host values.
-         for i,v in ipairs(value) do
-            newvalue[i] = validoutaccounts[v]
-         end
-         return newvalue
-      end
-   end
-   
-   -- Cook the new values prior to entering them into the config file.
-   -- Also, enter them only if they are valid.
-   function providers.write(self, section, value)
-      cookedvalue = {}
-      cindex = 1
-      for index, field in ipairs(value) do
-         cooked = string.gsub(luci.util.trim(value[index]), "%W", "_")
-         if validoutaccounts[cooked] ~= nil then
-            cookedvalue[cindex] = cooked
-            cindex = cindex + 1
-         end
-      end
-      DynamicList.write(self, section, cookedvalue)
-   end
-end
-
-----------------------------------------------------------------------------------------------------
-s = m:section(TypedSection, "callthrough_numbers", translate("Call-through Numbers"),
-        translate("Designate numbers that are allowed to call through this system and which user's \
-                  privileges they will have."))
-s.anonymous = true
-s.addremove = true
-
-num = s:option(DynamicList, "callthrough_number_list", translate("Call-through Numbers"),
-        translate("Specify numbers individually here. Press enter to add more numbers. \
-        You will have to experiment with what country and area codes you need to add \
-        to the number."))
-num.datatype = "uinteger"
-
-p = s:option(ListValue, "enabled", translate("Enabled"))
-p:value("yes", translate("Yes"))
-p:value("no",  translate("No"))
-p.default = "yes"
-
-user = s:option(Value, "defaultuser",  translate("User Name"),
-         translate("The number(s) specified above will be able to dial out with this user's providers. \
-                   Invalid usernames, including users not enabled for outgoing calls, are dropped silently. \
-                   Please verify that the entry was accepted."))
-function user.write(self, section, value)
-   trimuser = luci.util.trim(value)
-   if allvalidusers[trimuser] == true then
-      Value.write(self, section, trimuser)
-   end
-end
-
-pwd = s:option(Value, "pin", translate("PIN"),
-               translate("Your PIN disappears when saved for your protection. It will be changed \
-                         only when you enter a value different from the saved one. Leaving the PIN \
-                         empty is possible, but please beware of the security implications."))
-pwd.password = true
-pwd.rmempty = false
-
--- We skip reading off the saved value and return nothing.
-function pwd.cfgvalue(self, section)
-    return "" 
-end
-
--- We check the entered value against the saved one, and only write if the entered value is
--- something other than the empty string, and it differes from the saved value.
-function pwd.write(self, section, value)
-    local orig_pwd = m:get(section, self.option)
-    if value and #value > 0 and orig_pwd ~= value then
-        Value.write(self, section, value)
-    end
-end
-
-----------------------------------------------------------------------------------------------------
-s = m:section(TypedSection, "callback_numbers", translate("Call-back Numbers"),
-        translate("Designate numbers to whom the system will hang up and call back, which provider will \
-                   be used to call them, and which user's privileges will be granted to them."))
-s.anonymous = true
-s.addremove = true
-
-num = s:option(DynamicList, "callback_number_list", translate("Call-back Numbers"),
-        translate("Specify numbers individually here. Press enter to add more numbers. \
-        You will have to experiment with what country and area codes you need to add \
-        to the number."))
-num.datatype = "uinteger"
-
-p = s:option(ListValue, "enabled", translate("Enabled"))
-p:value("yes", translate("Yes"))
-p:value("no",  translate("No"))
-p.default = "yes"
-
-delay = s:option(Value, "callback_hangup_delay", translate("Hang-up Delay"),
-            translate("How long to wait before hanging up. If the provider you use to dial automatically forwards \
-            to voicemail, you can set this value to a delay that will allow you to hang up before your call gets \
-            forwarded and you get billed for it."))
-delay.datatype = "uinteger"
-delay.default = 0
-
-user = s:option(Value, "defaultuser",  translate("User Name"),
-         translate("The number(s) specified above will be able to dial out with this user's providers. \
-                   Invalid usernames, including users not enabled for outgoing calls, are dropped silently. \
-                   Please verify that the entry was accepted."))
-function user.write(self, section, value)
-   trimuser = luci.util.trim(value)
-   if allvalidusers[trimuser] == true then
-      Value.write(self, section, trimuser)
-   end
-end
-
-pwd = s:option(Value, "pin", translate("PIN"),
-               translate("Your PIN disappears when saved for your protection. It will be changed \
-                         only when you enter a value different from the saved one. Leaving the PIN \
-                         empty is possible, but please beware of the security implications."))
-pwd.password = true
-pwd.rmempty = false
-
--- We skip reading off the saved value and return nothing.
-function pwd.cfgvalue(self, section)
-    return "" 
-end
-
--- We check the entered value against the saved one, and only write if the entered value is
--- something other than the empty string, and it differes from the saved value.
-function pwd.write(self, section, value)
-    local orig_pwd = m:get(section, self.option)
-    if value and #value > 0 and orig_pwd ~= value then
-        Value.write(self, section, value)
-    end
-end
-
-provider = s:option(Value, "callback_provider",  translate("Call-back Provider"),
-         translate("Enter a VoIP provider to use for call-back in the format username@some.host.name, as listed in \
-         \"Outgoing Calls\" above. It's easiest to copy and paste the providers from above. Invalid entries, including \
-         providers not enabled for outgoing calls, will be rejected silently."))
-function provider.write(self, section, value)
-    cooked = string.gsub(luci.util.trim(value), "%W", "_")
-    if validoutaccounts[cooked] ~= nil then
-        Value.write(self, section, value)
-    end
-end
-
-----------------------------------------------------------------------------------------------------
-s = m:section(NamedSection, "blacklisting", "call_routing", translate("Blacklisted Numbers"),
-              translate("Enter phone numbers that you want to decline calls from automatically. \
-              You should probably omit the country code and any leading zeroes, but please \
-              experiment to make sure you are blocking numbers from your desired area successfully."))
-s.anonymous = true
-
-b = s:option(DynamicList, "blacklist1", translate("Dynamic List of Blacklisted Numbers"),
-            translate("Specify numbers individually here. Press enter to add more numbers."))
-b.cast = "string"
-b.datatype = "uinteger"
-
-b = s:option(Value, "blacklist2", translate("Space-Separated List of Blacklisted Numbers"),
-            translate("Copy-paste large lists of numbers here."))
-b.template = "cbi/tvalue"
-b.rows = 3
-
-return m
diff --git a/applications/luci-app-pbx/luasrc/model/cbi/pbx-google.lua b/applications/luci-app-pbx/luasrc/model/cbi/pbx-google.lua
deleted file mode 100644
index 3c36a168d9..0000000000
--- a/applications/luci-app-pbx/luasrc/model/cbi/pbx-google.lua
+++ /dev/null
@@ -1,122 +0,0 @@
---[[
-    Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com>
-
-    This file is part of luci-pbx.
-
-    luci-pbx is free software: you can redistribute it and/or modify
-    it under the terms of the GNU General Public License as published by
-    the Free Software Foundation, either version 3 of the License, or
-    (at your option) any later version.
-
-    luci-pbx is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-    GNU General Public License for more details.
-
-    You should have received a copy of the GNU General Public License
-    along with luci-pbx.  If not, see <http://www.gnu.org/licenses/>.
-]]--
-
-if     nixio.fs.access("/etc/init.d/asterisk")   then
-   server = "asterisk"
-elseif nixio.fs.access("/etc/init.d/freeswitch") then
-   server = "freeswitch"
-else
-   server = ""
-end
-
-modulename           = "pbx-google"
-googlemodulename     = "pbx-google"
-defaultstatus        = "dnd"
-defaultstatusmessage = "PBX online, may lose messages"
-
-m = Map (modulename, translate("Google Accounts"),
-         translate("This is where you set up your Google (Talk and Voice) Accounts, in order to start \
-                using them for dialing and receiving calls (voice chat and real phone calls). Please \
-                make at least one voice call using the Google Talk plugin installable through the \
-                GMail interface, and then log out from your account everywhere. Click \"Add\" \
-                to add as many accounts as you wish."))
-
--- Recreate the config, and restart services after changes are commited to the configuration.
-function m.on_after_commit(self)
-   -- Create a field "name" for each account that identifies the account in the backend.
-   commit = false
-   m.uci:foreach(modulename, "gtalk_jabber", 
-                 function(s1)
-                    if s1.username ~= nil then
-                       name=string.gsub(s1.username, "%W", "_")
-                       if s1.name ~= name then
-                          m.uci:set(modulename, s1['.name'], "name", name)
-                          commit = true
-                       end
-                    end
-                 end)
-   if commit == true then  m.uci:commit(modulename) end
-   
-   luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null")
-   luci.sys.call("/etc/init.d/asterisk             restart 1\>/dev/null 2\>/dev/null")
-end
-
------------------------------------------------------------------------------
-s = m:section(TypedSection, "gtalk_jabber", translate("Google Voice/Talk Accounts"))
-s.anonymous = true
-s.addremove = true
-
-s:option(Value, "username",     translate("Email"))
-
-pwd = s:option(Value, "secret", translate("Password"),
-               translate("When your password is saved, it disappears from this field and is not displayed \
-                         for your protection. The previously saved password will be changed only when you \
-                         enter a value different from the saved one."))
-pwd.password = true
-pwd.rmempty = false
-
--- We skip reading off the saved value and return nothing.
-function pwd.cfgvalue(self, section)
-    return "" 
-end
-
--- We check the entered value against the saved one, and only write if the entered value is
--- something other than the empty string, and it differes from the saved value.
-function pwd.write(self, section, value)
-    local orig_pwd = m:get(section, self.option)
-    if value and #value > 0 and orig_pwd ~= value then
-        Value.write(self, section, value)
-    end
-end
-
-
-p = s:option(ListValue, "register",
-             translate("Enable Incoming Calls (set Status below)"),
-             translate("When somebody starts voice chat with your GTalk account or calls the GVoice, \
-                       number (if you have Google Voice), the call will be forwarded to any users \
-                        that are online (registered using a SIP device or softphone) and permitted to \
-                        receive the call. If you have Google Voice, you must go to your GVoice settings and \
-                        forward calls to Google chat in order to actually receive calls made to your \
-                        GVoice number. If you have trouble receiving calls from GVoice, experiment \
-                        with the Call Screening option in your GVoice Settings. Finally, make sure no other \
-                        client is online with this account (browser in gmail, mobile/desktop Google Talk \
-                        App) as it may interfere."))
-p:value("yes", translate("Yes"))
-p:value("no",  translate("No"))
-p.default = "yes"
-
-p = s:option(ListValue, "make_outgoing_calls", translate("Enable Outgoing Calls"),
-      translate("Use this account to make outgoing calls as configured in the \"Call Routing\" section."))
-p:value("yes", translate("Yes"))
-p:value("no",  translate("No"))
-p.default = "yes"
-
-st = s:option(ListValue, "status", translate("Google Talk Status"))
-st:depends("register", "yes")
-st:value("dnd", translate("Do Not Disturb"))
-st:value("away",  translate("Away"))
-st:value("available",  translate("Available"))
-st.default = defaultstatus
-
-stm = s:option(Value, "statusmessage", translate("Google Talk Status Message"),
-             translate("Avoid using anything but alpha-numeric characters, space, comma, and period."))
-stm:depends("register", "yes")
-stm.default = defaultstatusmessage
-
-return m
diff --git a/applications/luci-app-pbx/luasrc/model/cbi/pbx-users.lua b/applications/luci-app-pbx/luasrc/model/cbi/pbx-users.lua
deleted file mode 100644
index c7c8b4d8bb..0000000000
--- a/applications/luci-app-pbx/luasrc/model/cbi/pbx-users.lua
+++ /dev/null
@@ -1,133 +0,0 @@
---[[
-    Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com>
-
-    This file is part of luci-pbx.
-
-    luci-pbx is free software: you can redistribute it and/or modify
-    it under the terms of the GNU General Public License as published by
-    the Free Software Foundation, either version 3 of the License, or
-    (at your option) any later version.
-
-    luci-pbx is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-    GNU General Public License for more details.
-
-    You should have received a copy of the GNU General Public License
-    along with luci-pbx.  If not, see <http://www.gnu.org/licenses/>.
-]]--
-
-if     nixio.fs.access("/etc/init.d/asterisk")   then
-   server = "asterisk"
-elseif nixio.fs.access("/etc/init.d/freeswitch") then
-   server = "freeswitch"
-else
-   server = ""
-end
-
-modulename         = "pbx-users"
-modulenamecalls    = "pbx-calls"
-modulenameadvanced = "pbx-advanced"
-
-        
-m = Map (modulename, translate("User Accounts"), 
-        translate("Here you must configure at least one SIP account, that you \
-                will use to register with this service. Use this account either in an Analog Telephony \
-                Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid on your \
-                smartphone, or Ekiga, Linphone, or X-Lite on your computer. By default, all SIP accounts \
-                will ring simultaneously if a call is made to one of your VoIP provider accounts or GV \
-                numbers."))
-
--- Recreate the config, and restart services after changes are commited to the configuration.
-function m.on_after_commit(self)
-        luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null")
-        luci.sys.call("/etc/init.d/"     .. server .. " restart 1\>/dev/null 2\>/dev/null")
-end
-
-externhost = m.uci:get(modulenameadvanced, "advanced", "externhost")
-bindport   = m.uci:get(modulenameadvanced, "advanced", "bindport")
-ipaddr     = m.uci:get("network", "lan", "ipaddr")
-
------------------------------------------------------------------------------
-s = m:section(NamedSection, "server", "user", translate("Server Setting"))
-s.anonymous = true
-
-if ipaddr == nil or ipaddr == "" then
-   ipaddr = "(IP address not static)"
-end
-
-if bindport ~= nil then
-   just_ipaddr = ipaddr
-   ipaddr = ipaddr .. ":" .. bindport
-end
-
-s:option(DummyValue, "ipaddr", translate("Server Setting for Local SIP Devices"),
-         translate("Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices you will \
-                   use ONLY locally and never from a remote location.")).default = ipaddr
-
-if externhost ~= nil then
-   if bindport ~= nil then
-      just_externhost = externhost
-      externhost = externhost .. ":" .. bindport
-   end
-   s:option(DummyValue, "externhost", translate("Server Setting for Remote SIP Devices"),
-            translate("Enter this hostname (or hostname:port) in the Server/Registrar setting of SIP \
-                      devices you will use from a remote location (they will work locally too).")
-                     ).default = externhost
-end
-
-if bindport ~= nil then
-        s:option(DummyValue, "bindport", translate("Port Setting for SIP Devices"),
-        translatef("If setting Server/Registrar to %s or %s does not work for you, try setting \
-        it to %s or %s and entering this port number in a separate field that specifies the \
-        Server/Registrar port number. Beware that some devices have a confusing \
-        setting that sets the port where SIP requests originate from on the SIP \
-        device itself (the bind port). The port specified on this page is NOT this bind port \
-        but the port this service listens on.", 
-        ipaddr, externhost, just_ipaddr, just_externhost)).default = bindport
-end
-
------------------------------------------------------------------------------
-s = m:section(TypedSection, "local_user", translate("SIP Device/Softphone Accounts"))
-s.anonymous = true
-s.addremove = true
-
-s:option(Value, "fullname", translate("Full Name"),
-         translate("You can specify a real name to show up in the Caller ID here."))
-
-du = s:option(Value, "defaultuser",  translate("User Name"),
-         translate("Use (four to five digit) numeric user name if you are connecting normal telephones \
-                   with ATAs to this system (so they can dial user names)."))
-du.datatype = "uciname"
-
-pwd = s:option(Value, "secret", translate("Password"),
-               translate("Your password disappears when saved for your protection. It will be changed \
-                         only when you enter a value different from the saved one."))
-pwd.password = true
-pwd.rmempty = false
-
--- We skip reading off the saved value and return nothing.
-function pwd.cfgvalue(self, section)
-    return "" 
-end
-
--- We check the entered value against the saved one, and only write if the entered value is
--- something other than the empty string, and it differes from the saved value.
-function pwd.write(self, section, value)
-    local orig_pwd = m:get(section, self.option)
-    if value and #value > 0 and orig_pwd ~= value then
-        Value.write(self, section, value)
-    end
-end
-
-p = s:option(ListValue, "ring", translate("Receives Incoming Calls"))
-p:value("yes", translate("Yes"))
-p:value("no",  translate("No"))
-p.default = "yes"
-
-p = s:option(ListValue, "can_call", translate("Makes Outgoing Calls"))
-p:value("yes", translate("Yes"))
-p:value("no",  translate("No"))
-p.default = "yes"
-
-return m
diff --git a/applications/luci-app-pbx/luasrc/model/cbi/pbx-voip.lua b/applications/luci-app-pbx/luasrc/model/cbi/pbx-voip.lua
deleted file mode 100644
index 9b46202855..0000000000
--- a/applications/luci-app-pbx/luasrc/model/cbi/pbx-voip.lua
+++ /dev/null
@@ -1,116 +0,0 @@
---[[
-    Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com>
-
-    This file is part of luci-pbx.
-
-    luci-pbx is free software: you can redistribute it and/or modify
-    it under the terms of the GNU General Public License as published by
-    the Free Software Foundation, either version 3 of the License, or
-    (at your option) any later version.
-
-    luci-pbx is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-    GNU General Public License for more details.
-
-    You should have received a copy of the GNU General Public License
-    along with luci-pbx.  If not, see <http://www.gnu.org/licenses/>.
-]]--
-
-if     nixio.fs.access("/etc/init.d/asterisk")   then
-   server = "asterisk"
-elseif nixio.fs.access("/etc/init.d/freeswitch") then
-   server = "freeswitch"
-else
-   server = ""
-end
-
-modulename = "pbx-voip"
-
-m = Map (modulename, translate("SIP Accounts"),
-         translate("This is where you set up your SIP (VoIP) accounts ts like Sipgate, SipSorcery, \
-        the popular Betamax providers, and any other providers with SIP settings in order to start \
-        using them for dialing and receiving calls (SIP uri and real phone calls). Click \"Add\" to \
-        add as many accounts as you wish."))
-
--- Recreate the config, and restart services after changes are commited to the configuration.
-function m.on_after_commit(self)
-   commit = false
-   -- Create a field "name" for each account that identifies the account in the backend.
-   m.uci:foreach(modulename, "voip_provider", 
-                 function(s1)
-                    if s1.defaultuser ~= nil and s1.host ~= nil then
-                       name=string.gsub(s1.defaultuser.."_"..s1.host, "%W", "_")
-                       if s1.name ~= name then
-                          m.uci:set(modulename, s1['.name'], "name", name)
-                          commit = true
-                       end
-                    end
-                 end)
-   if commit == true then m.uci:commit(modulename) end
-
-   luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null")
-   luci.sys.call("/etc/init.d/"     .. server .. " restart 1\>/dev/null 2\>/dev/null")
-end
-
------------------------------------------------------------------------------
-s = m:section(TypedSection, "voip_provider", translate("SIP Provider Accounts"))
-s.anonymous = true
-s.addremove = true
-
-s:option(Value, "defaultuser",  translate("User Name"))
-pwd = s:option(Value, "secret", translate("Password"),
-               translate("When your password is saved, it disappears from this field and is not displayed \
-                         for your protection. The previously saved password will be changed only when you \
-                         enter a value different from the saved one."))
-
-
-
-pwd.password = true
-pwd.rmempty = false
-
--- We skip reading off the saved value and return nothing.
-function pwd.cfgvalue(self, section)
-    return "" 
-end
-
--- We check the entered value against the saved one, and only write if the entered value is
--- something other than the empty string, and it differes from the saved value.
-function pwd.write(self, section, value)
-    local orig_pwd = m:get(section, self.option)
-    if value and #value > 0 and orig_pwd ~= value then
-        Value.write(self, section, value)
-    end
-end
-
-h = s:option(Value, "host", translate("SIP Server/Registrar"))
-h.datatype = "host(0)"
-
-p = s:option(ListValue, "register", translate("Enable Incoming Calls (Register via SIP)"),
-             translate("This option should be set to \"Yes\" if you have a DID \(real telephone number\) \
-                        associated with this SIP account or want to receive SIP uri calls through this \
-                        provider.")) 
-p:value("yes", translate("Yes"))
-p:value("no",  translate("No"))
-p.default = "yes"
-
-p = s:option(ListValue, "make_outgoing_calls", translate("Enable Outgoing Calls"),
-             translate("Use this account to make outgoing calls."))
-p:value("yes", translate("Yes"))
-p:value("no",  translate("No"))
-p.default = "yes"
-
-from = s:option(Value, "fromdomain",
-                translate("SIP Realm (needed by some providers)"))
-from.optional = true
-from.datatype = "host(0)"
-
-port = s:option(Value, "port", translate("SIP Server/Registrar Port"))
-port.optional = true
-port.datatype = "port"
-
-op = s:option(Value, "outboundproxy", translate("Outbound Proxy"))
-op.optional = true
-op.datatype = "host(0)"
-
-return m
diff --git a/applications/luci-app-pbx/luasrc/model/cbi/pbx.lua b/applications/luci-app-pbx/luasrc/model/cbi/pbx.lua
deleted file mode 100644
index 4c5fcbdecd..0000000000
--- a/applications/luci-app-pbx/luasrc/model/cbi/pbx.lua
+++ /dev/null
@@ -1,115 +0,0 @@
---[[
-    Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com>
-
-    This file is part of luci-pbx.
-
-    luci-pbx is free software: you can redistribute it and/or modify
-    it under the terms of the GNU General Public License as published by
-    the Free Software Foundation, either version 3 of the License, or
-    (at your option) any later version.
-
-    luci-pbx is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-    GNU General Public License for more details.
-
-    You should have received a copy of the GNU General Public License
-    along with luci-pbx.  If not, see <http://www.gnu.org/licenses/>.
-]]--
-
-modulename = "pbx"
-
-
-if     nixio.fs.access("/etc/init.d/asterisk")   then
-   server = "asterisk"
-elseif nixio.fs.access("/etc/init.d/freeswitch") then
-   server = "freeswitch"
-else
-   server = ""
-end
-
-
--- Returns formatted output of string containing only the words at the indices
--- specified in the table "indices".
-function format_indices(string, indices)
-   if indices == nil then
-      return "Error: No indices to format specified.\n" 
-   end
-
-   -- Split input into separate lines.
-   lines = luci.util.split(luci.util.trim(string), "\n")
-   
-   -- Split lines into separate words.
-   splitlines = {}
-   for lpos,line in ipairs(lines) do
-      splitlines[lpos] = luci.util.split(luci.util.trim(line), "%s+", nil, true)
-   end
-   
-   -- For each split line, if the word at all indices specified
-   -- to be formatted are not null, add the formatted line to the
-   -- gathered output.
-   output = ""
-   for lpos,splitline in ipairs(splitlines) do
-      loutput = ""
-      for ipos,index in ipairs(indices) do
-         if splitline[index] ~= nil then
-            loutput = loutput .. string.format("%-40s", splitline[index])
-         else
-            loutput = nil
-            break
-         end
-      end
-      
-      if loutput ~= nil then
-         output = output .. loutput .. "\n"
-      end
-   end
-   return output
-end
-
-
-m = Map (modulename, translate("PBX Main Page"),
-      translate("This configuration page allows you to configure a phone system (PBX) service which \
-      permits making phone calls through multiple Google and SIP (like Sipgate, \
-      SipSorcery, and Betamax) accounts and sharing them among many SIP devices. \
-      Note that Google accounts, SIP accounts, and local user accounts are configured in the \
-      \"Google Accounts\", \"SIP Accounts\", and \"User Accounts\" sub-sections. \
-      You must add at least one User Account to this PBX, and then configure a SIP device or \
-      softphone to use the account, in order to make and receive calls with your Google/SIP \
-      accounts. Configuring multiple users will allow you to make free calls between all users, \
-      and share the configured Google and SIP accounts. If you have more than one Google and SIP \
-      accounts set up, you should probably configure how calls to and from them are routed in \
-      the \"Call Routing\" page. If you're interested in using your own PBX from anywhere in the \
-      world, then visit the \"Remote Usage\" section in the \"Advanced Settings\" page."))
-
------------------------------------------------------------------------------------------
-s = m:section(NamedSection, "connection_status", "main",
-              translate("PBX Service Status"))
-s.anonymous = true
-
-s:option (DummyValue, "status", translate("Service Status"))
-
-sts = s:option(DummyValue, "_sts") 
-sts.template = "cbi/tvalue"
-sts.rows = 20
-
-function sts.cfgvalue(self, section)
-
-   if server == "asterisk" then
-      regs = luci.sys.exec("asterisk -rx 'sip show registry' | sed 's/peer-//'")
-      jabs = luci.sys.exec("asterisk -rx 'jabber show connections' | grep onnected")
-      usrs = luci.sys.exec("asterisk -rx 'sip show users'")
-      chan = luci.sys.exec("asterisk -rx 'core show channels'")
-
-      return format_indices(regs, {1, 5}) ..
-             format_indices(jabs, {2, 4}) .. "\n" ..
-             format_indices(usrs, {1}   ) .. "\n" .. chan
-
-   elseif server == "freeswitch" then
-      return "Freeswitch is not supported yet.\n"
-   else
-      return "Neither Asterisk nor FreeSwitch discovered, please install Asterisk, as Freeswitch is not supported yet.\n"
-   end
-end
-
-return m
diff --git a/applications/luci-app-pbx/po/ca/pbx.po b/applications/luci-app-pbx/po/ca/pbx.po
deleted file mode 100644
index c8a0a9967e..0000000000
--- a/applications/luci-app-pbx/po/ca/pbx.po
+++ /dev/null
@@ -1,509 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2014-07-01 05:14+0200\n"
-"Last-Translator: Alex <alexhenrie24@gmail.com>\n"
-"Language-Team: none\n"
-"Language: ca\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Advanced Settings"
-msgstr "Ajusts avançats"
-
-msgid "Available"
-msgstr "Disponible"
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-"Eviteu utilitzar res excepte caràcters alfanumèrics, espai, coma, i punt."
-
-msgid "Away"
-msgstr "Fora"
-
-msgid "Blacklisted Numbers"
-msgstr "Nombres prohibits"
-
-msgid "Call Routing"
-msgstr "Encaminament de trucades"
-
-msgid "Call-back Numbers"
-msgstr "Nombres de trucada de tornada"
-
-msgid "Call-back Provider"
-msgstr "Proveïdor de trucada de tornada"
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr "Copieu i enganxeu llistes grans de nombres aquí."
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-"Designeu els nombres que es permeten trucar a través d'aquest sistema i els "
-"privilegis de qual usuari tindran."
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-"Designeu els nombres als quals el sistema penjarà i trucarà de tornada, qual "
-"proveïdor s'emprarà per a trucar-los, i els privilegis de qual usuari se "
-"lis concedirà."
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr "Truca els nombres que no coincideixen d'altra manera"
-
-msgid "Do Not Disturb"
-msgstr "No molestis"
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Email"
-msgstr ""
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr "Habilita trucades entrants (registreu via SIP)"
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr "Habilita trucades entrants (establiu l'Estat a baix)"
-
-msgid "Enable Outgoing Calls"
-msgstr "Habilita trucades sortints"
-
-msgid "Enabled"
-msgstr "Habilitat"
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr "Port SIP extern"
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr "Nom complet"
-
-msgid "General Settings"
-msgstr "Ajusts generals"
-
-msgid "Google Accounts"
-msgstr "Comptes de Google"
-
-msgid "Google Talk Status"
-msgstr ""
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr ""
-
-msgid "Hang-up Delay"
-msgstr "Retard de penja"
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-"Quant temps per a esperar abans de penjar. Si el proveïdor que empreu per a "
-"trucar automàticament redirigeix al correu de veu, podeu estableix aquest "
-"valor a un retard que us permet penjar abans que la teva trucada es "
-"redirigeixi i s'us cobri per ella."
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr "Trucades entrants"
-
-msgid "Insert QoS Rules"
-msgstr "Insereix regles QoS"
-
-msgid "Makes Outgoing Calls"
-msgstr "Fa trucades sortints"
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr "NOTA: No hi ha cap compte configurat ni del Google ni de proveïdor SIP."
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-"NOTA: No hi ha cap compte habilitat ni del Google ni de proveïdor SIP per "
-"als trucades entrants."
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-"NOTA: No hi ha cap compte habilitat ni del Google ni de proveïdor SIP per "
-"als trucades sortints."
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr "NOTA: No hi ha cap compte d'usuari local configurat."
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-"NOTA: No hi ha cap compte d'usuari local habilitat per als trucades "
-"sortints."
-
-msgid "No"
-msgstr "No"
-
-msgid "Number of Seconds to Ring"
-msgstr "Nombre de segons a sonar"
-
-msgid "Outbound Proxy"
-msgstr "Servidor intermediari de sortida"
-
-msgid "Outgoing Calls"
-msgstr "Trucades sortints"
-
-msgid "PBX Main Page"
-msgstr "Pàgina principal PBX"
-
-msgid "PBX Service Status"
-msgstr "Estat del servei PBX"
-
-msgid "PIN"
-msgstr "PIN"
-
-msgid "Password"
-msgstr "Contrasenya"
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr "Ajust de port per als dispositius SIP"
-
-msgid "Providers Used for Outgoing Calls"
-msgstr "Proveïdors utilitzats per als trucades sortints"
-
-msgid "QoS Settings"
-msgstr "Ajusts QoS"
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr "Rep trucades entrants"
-
-msgid "Remote Usage"
-msgstr "Ús remot"
-
-msgid "Rings users enabled for incoming calls"
-msgstr "Truca als usuaris habilitats per a rebre trucades"
-
-msgid "SIP Accounts"
-msgstr "Comptes SIP"
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr "Comptes de proveïdor SIP"
-
-msgid "SIP Realm (needed by some providers)"
-msgstr "Regne SIP (necessitat per alguns proveïdors)"
-
-msgid "SIP Server/Registrar"
-msgstr "Servidor/Registrador SIP"
-
-msgid "SIP Server/Registrar Port"
-msgstr "Port del Servidor/Registrador SIP"
-
-msgid "Server Setting"
-msgstr "Ajust de servidor"
-
-msgid "Server Setting for Local SIP Devices"
-msgstr "Ajust de servidor pels dispositius SIP locals"
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr "Ajust de servidor pels dispositius SIP remots"
-
-msgid "Service Status"
-msgstr "Estat de servei"
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-"Estableix el nombre de segons per a sonar als usuaris abans de penjar o anar "
-"al correu de veu, si el correu de veu està instal·lat i habilitat."
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr "Llista de nombres prohibits separats per espai"
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-"Especifiqueu els nombres individualment aquí. Premeu Enter per afegir més "
-"nombres."
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-"Utilitza aquest compte per fer trucades sortints com configurat en la secció "
-"\"Encaminament de trucades\"."
-
-msgid "Use this account to make outgoing calls."
-msgstr "Utilitza aquest compte per fer trucades sortints."
-
-msgid "User Accounts"
-msgstr "Comptes d'usuari"
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr "Nom d'usuari"
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr "Sí"
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
diff --git a/applications/luci-app-pbx/po/cs/pbx.po b/applications/luci-app-pbx/po/cs/pbx.po
deleted file mode 100644
index 8b69ef15d8..0000000000
--- a/applications/luci-app-pbx/po/cs/pbx.po
+++ /dev/null
@@ -1,487 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2014-07-12 20:19+0200\n"
-"Last-Translator: koli <lukas.koluch@gmail.com>\n"
-"Language-Team: none\n"
-"Language: cs\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=3; plural=(n==1) ? 0 : (n>=2 && n<=4) ? 1 : 2;\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Advanced Settings"
-msgstr "Pokročilé nastavení"
-
-msgid "Available"
-msgstr "Dostupné"
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-
-msgid "Away"
-msgstr "Pryč"
-
-msgid "Blacklisted Numbers"
-msgstr ""
-
-msgid "Call Routing"
-msgstr ""
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr ""
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr "Nevyrušovat"
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Email"
-msgstr "Email"
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr "Povolit příchozí hovory (Registrace přes SIP)"
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr ""
-
-msgid "Enable Outgoing Calls"
-msgstr "Povolit odchozí hovory"
-
-msgid "Enabled"
-msgstr "Povoleno"
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr "Externí SIP port"
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr "Celé jméno (jméno a příjmení)"
-
-msgid "General Settings"
-msgstr "Obecné nastavení"
-
-msgid "Google Accounts"
-msgstr "Google účty"
-
-msgid "Google Talk Status"
-msgstr "Stav Google Talk"
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr "Google Voice/Talk účty"
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr "Příchozí volání"
-
-msgid "Insert QoS Rules"
-msgstr "Vložte QoS pravidla"
-
-msgid "Makes Outgoing Calls"
-msgstr ""
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr "Ne"
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr "Odchozí volání"
-
-msgid "PBX Main Page"
-msgstr "Hlavní stránka PBX"
-
-msgid "PBX Service Status"
-msgstr "Stav PBX služby"
-
-msgid "PIN"
-msgstr "PIN"
-
-msgid "Password"
-msgstr "Heslo"
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr "Nastavení QoS"
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr ""
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr "SIP účty"
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr "Stav služby"
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr "Uživatelské účty"
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr "Uživatelské jméno"
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr "Ano"
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
diff --git a/applications/luci-app-pbx/po/de/pbx.po b/applications/luci-app-pbx/po/de/pbx.po
deleted file mode 100644
index 3bc4bd428e..0000000000
--- a/applications/luci-app-pbx/po/de/pbx.po
+++ /dev/null
@@ -1,699 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2013-01-30 18:17+0200\n"
-"Last-Translator: DAC324 <gerd_roethig@web.de>\n"
-"Language-Team: none\n"
-"Language: de\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Advanced Settings"
-msgstr "Erweiterte Einstellungen"
-
-msgid "Available"
-msgstr "Verfügbar"
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr "Nur alphanumerische Zeichen, Komma, Punkt und Leerzeichen verwenden"
-
-msgid "Away"
-msgstr "Abwesend"
-
-msgid "Blacklisted Numbers"
-msgstr "Nicht erlaubte Nummern (Blacklist)"
-
-msgid "Call Routing"
-msgstr "Anrufweiterleitung"
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr "Durchwahl Nummern"
-
-msgid "Copy-paste large lists of numbers here."
-msgstr "Hier können per Copy & Paste größere Nummernlisten eingefügt werden."
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr "Wählt Nummern an, für die es keine andere Übereinstimmung gibt"
-
-msgid "Do Not Disturb"
-msgstr "Beschäftigt"
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr "Domäne/IP-Adresse/Dynamische Domäne"
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr "Dynamische Liste nicht erlaubter Nummern (Dynamische Blacklist)"
-
-msgid "Email"
-msgstr "E-Mail"
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr "Eingehende Anrufe akzeptieren (registrieren via SIP)"
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr "Eingehende Anrufe akzeptieren (Status unten einstellen)"
-
-msgid "Enable Outgoing Calls"
-msgstr "Ausgehende Anrufe aktivieren"
-
-msgid "Enabled"
-msgstr "Aktiv"
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-"Geben Sie Telefonnummern ein, von denen Anrufe automatisch zurückgewiesen "
-"werden sollen. Sie sollten die Ländervorwahl und alle führenden Nullen "
-"weglassen, aber experimentieren Sie ruhig, damit Sie auch wirklich alle "
-"Nummern blockieren, die blockiert werden sollen."
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-"Geben SIe diese IP (oder IP:Port) in der Server-/Registrar-Einstellung der "
-"SIP-Geräte an, die Sie NUR local und niemals von einem entfernten Ort "
-"einsetzen werden."
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-"Geben SIe diese IP (oder IP:Port) in der Server-/Registrar-Einstellung der "
-"SIP-Geräte an, die Sie von einem entfernten Ort einsetzen werden (sie "
-"funktionieren auch lokal)."
-
-msgid "External SIP Port"
-msgstr "Externer SIP Port"
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-"Hier können Sie für jeden Dienstanbieter, der für eingehende Anrufe "
-"eingerichtet ist, festlegen, welche Nutzer ein Klingelzeichen bei "
-"eingehenden Anrufen erhalten. Ist die Liste leer, klingelt es bei allen "
-"Nutzern, die eingehende Anrufe empfangen dürfen. Ungültige Benutzernamen "
-"werden ohne Fehlermeldung zurückgewiesen. Außerdem überschreibt der Eintrag "
-"eines Benutzernamens an dieser Stelle die evtl. vorhandene Einstellung für "
-"diesen Benutzer, keine eingehenden Anrufe zu erhalten. Auf diese Weise kann "
-"eingestellt werden, dass die Nutzer nur bei bestimmten Dienstanbietern ein "
-"Klingelzeichen erhalten. Einträge in dieser Liste können entweder durch "
-"Leerzeichen getrennt oder als ein Eintrag pro Zeile (Eingabetaste nach jedem "
-"Eintrag) eingegeben werden."
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-"Hier können Sie für jeden Benutzer, der für abgehende Anrufe eingerichtet "
-"ist, festlegen, welche Dienstanbieter verwendet werden dürfen. In der "
-"Voreinstellung dürfen alle Benutzer auch alle Dienstanbieter verwenden. Um "
-"in der Liste unten aufzutauchen, sollte dem Benutzer auf der Seite "
-"\"Benutzerkonten\" erlaubt werden, abgehende Anrufe machen zu dürfen. Geben "
-"Sie VoIP-Dienstanbieter im Format Benutzername@Servername an, wie bereits "
-"oben unter \"Abgehende Anrufe\". Am einfachsten kopieren Sie die "
-"Dienstanbieter von dort und fügen sie hier wieder ein. Ungültige Einträge, "
-"einschließlich nicht für abgehende Anrufe zugelassene Dienstanbieter, werden "
-"ohne Fehlermeldung zurückgewiesen. Einträge in dieser Liste können entweder "
-"durch Leerzeichen getrennt und/oder als ein Eintrag pro Zeile (Eingabetaste "
-"nach jedem Eintrag) eingegeben werden."
-
-msgid "Full Name"
-msgstr "Vollständiger Name"
-
-msgid "General Settings"
-msgstr "Allgemeine Einstellungen"
-
-msgid "Google Accounts"
-msgstr "Google-Konten"
-
-msgid "Google Talk Status"
-msgstr "Status für Google Talk"
-
-msgid "Google Talk Status Message"
-msgstr "Statusbenachrichtigung für Google Talk"
-
-msgid "Google Voice/Talk Accounts"
-msgstr "Google Voice/Talk-Konten"
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-"Hier müssen Sie wenigstens ein SIP-Konto angeben, welches Sie zur Anmeldung "
-"an diesen Dienst nutzen. Verwenden Sie dieses Konto entweder in einem "
-"Adapter für analoges Telefonieren (ATA) oder einer SIP-Software wie "
-"CSipSimple, Linphone, oder Sipdroid auf Ihrem Smartphone, oder Ekiga, "
-"Linphone, oder X-Lite auf Ihrem Computer. In der Voreinstellung klingeln "
-"alle SIP-Konten gleichzeitig, wenn ein Anruf auf eines Ihrer VoIP-Konten "
-"oder Ihre GV-Nummern gemacht wird."
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-"Wenn EInstellen des Servers/Registrars auf %s oder %s bei Ihnen nicht "
-"funktioniert, versuchen Sie die Einstellung %s oder %s und geben Sie die "
-"Portnummer in ein separates Feld für Server/Registrat-Portnummer ein. "
-"Achtung: Einige Geräte haben eine verwirrende Einstellung, die den Port "
-"setzt, von dem die SIP-Anfragen auf dem Gerät selbst herkommen (der Bindungs-"
-"Port). Der Port auf dieser Seite meint NICHT diesen Bindungs-Port, sondern "
-"den Port, an dem der Dienst lauscht."
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-"Wenn Sie stotternden oder stark verzögerten Ton während großer Downloads "
-"haben, sollten Sie QoS einschalten. QoS priorisiert Verkehr von und zu Ihrem "
-"Netzwerk für bestimmte Ports und IP-Adressen mit dem Ergebnis einer besseren "
-"Tonübertragung in unserem Fall. Wenn unten eingeschaltet, wird eine QoS-"
-"Regel automatisch vom PBX eingerichtet, aber Sie müssen die QoS-"
-"Konfigurationsseite (Netzwerk->QoS) aufrufen, um andere kritische QoS-"
-"Einstellungen wie Upload-und Download-Geschwindigkeit vorzunehmen."
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-"Wenn Sie mehr als ein Konto für abgehende Anrufe haben, sollten Sie eine "
-"Liste von Telefonnummern/Vorwahlen in den folgenden Feldern für jeden "
-"aufgeführten Dienstanbieter eintragen. Ungültige Vorwahlen werden ohne "
-"Fehlermeldung entfernt, nur 0-9, X, Z, N, #, *, und + sind gültige Zeichen. "
-"Der Buchstabe X entspricht 0-9, Z entrpricht 1-9, N entspricht 2-9. Zum "
-"Beispiel können Sie 49 eingeben, um Anrufe nach Deutschland über einen "
-"Dienstanbieter zu tätigen. Für Anrufe nach Nordamerika geben Sie 1NXXNXXXXXX "
-"an. Unterstützt ein Dienstanbieter Ortsgespräche, wie im Gebiet 646 von New "
-"York, geben Sie 646NXXXXXX für diesen Anbieter ein. Ein Konto sollte eine "
-"leere Liste behalten, damit Sie darüber standardmäßig Anrufe tätigen können, "
-"wenn keine der Vorwahlen für die anderen Anbieter übereinstimmt. Das System "
-"ersetzt eine leere Liste automatisch mit dem Eintrag, dass dieser Anbieter "
-"alle Vorwahlen unterstützt, die von den anderen Anbietern nicht unterstützt "
-"werden. Seien Sie so spezifisch wie möglich (1NXXNXXXXXX ist besser als 1). "
-"Bitte beachten Sie, dass alle internationalen Vorwahl-Codes (wie 00, 011, "
-"010, 0011) verworfen werden. Einträge können durch Leezeichen getrennt und/"
-"oder einzeln pro Zeile (Abschließen mit Eingabe-Taste) eingegeben werden."
-
-msgid "Incoming Calls"
-msgstr "Eingehende Anrufe"
-
-msgid "Insert QoS Rules"
-msgstr "QoS-Regeln einfügen"
-
-msgid "Makes Outgoing Calls"
-msgstr "Macht ausgehende Anrufe"
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-"ACHTUNG: Es sind keine Konten für Google oder einen SIP-Dienstanbieter "
-"eingerichtet."
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-"ACHTUNG: Es sind keine Konten für Google oder einen SIP-Dienstanbieter für "
-"eingehende Anrufe eingerichtet."
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-"ACHTUNG: Es sind keine Konten für Google oder einen SIP-Dienstanbieter für "
-"abgehende Anrufe eingerichtet."
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr "ACHTUNG: Es sind keine lokalen Benutzerkonten eingerichtet."
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-"ACHTUNG: Es sind keine lokalen Benutzerkonten für abgehende Anrufe "
-"eingerichtet."
-
-msgid "No"
-msgstr "Nein"
-
-msgid "Number of Seconds to Ring"
-msgstr "Dauer des Klingelns in Sekunden"
-
-msgid "Outbound Proxy"
-msgstr "Proxy für ausgehende Verbindungen"
-
-msgid "Outgoing Calls"
-msgstr "Abgehende Anrufe"
-
-msgid "PBX Main Page"
-msgstr "PBX-Hauptseite"
-
-msgid "PBX Service Status"
-msgstr "PBX-Dienststatus"
-
-msgid "PIN"
-msgstr "PIN"
-
-msgid "Password"
-msgstr "Passwort"
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr "Port-Einstellung für SIP-Geräte"
-
-msgid "Providers Used for Outgoing Calls"
-msgstr "Provider für abgehende Anrufe"
-
-msgid "QoS Settings"
-msgstr "QoS Einstellungen"
-
-msgid "RTP Port Range End"
-msgstr "Ende des RTP-Port-Bereichs"
-
-msgid "RTP Port Range Start"
-msgstr "Anfang des RTP-Port-Bereichs"
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-"RTP-Verkehr überträgt die aktuellen Sprachpakete. Dies ist der Anfang des "
-"Port-Bereichs, der für die Einrichtung der RTP-Verbindung verwendet wird. "
-"Normalerweise kann hier die Voreinstellung belassen werden."
-
-msgid "Receives Incoming Calls"
-msgstr "Empfängt eingehende Anrufe"
-
-msgid "Remote Usage"
-msgstr "Benutzung aus der Ferne"
-
-msgid "Rings users enabled for incoming calls"
-msgstr "Für eingehende Anrufe freigeschaltete Nutzer erhalten Klingelzeichen"
-
-msgid "SIP Accounts"
-msgstr "SIP-Konten"
-
-msgid "SIP Device/Softphone Accounts"
-msgstr "SIP-Geräte-/Softphone-Konten"
-
-msgid "SIP Provider Accounts"
-msgstr "SIP-Dienstanbieter-Konten"
-
-msgid "SIP Realm (needed by some providers)"
-msgstr "SIP-Bereich (von manchen Dienstanbietern benötigt)"
-
-msgid "SIP Server/Registrar"
-msgstr "SIP-Server/Registrar"
-
-msgid "SIP Server/Registrar Port"
-msgstr "SIP-Server/Registrar Port"
-
-msgid "Server Setting"
-msgstr "Servereinstellung"
-
-msgid "Server Setting for Local SIP Devices"
-msgstr "Servereinstellung für lokale SIP-Geräte"
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr "Servereinstellung für entfernte SIP-Geräte"
-
-msgid "Service Status"
-msgstr "Dienst-Status"
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-"Stellen Sie ein (in Sekunden), wie lange es bei den Benutzern klingeln soll, "
-"bevor aufgelegt oder zur Voicemail (falls installiert und aktiv) "
-"übergegangen wird.  "
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr "Mit Leerzeichen unterteilte Liste gesperrter Nummern"
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-"Geben Sie die Nummern hier einzeln an. Drücken Sie Eingabe, um weitere "
-"Nummern hinzuzufügen."
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-"Die oben angegebene(n) Nummer(n) können für ausgehende Anrufe mit den "
-"Dienstanbietern dieses Nutzers verwendet werden. Ungültige Benutzernamen, "
-"einschließlich Nutzer, die nicht für ausgehende Anrufe freigeschaltet sind, "
-"werden ohne Fehlermeldung verworfen. Bitte überprüfen Sie deshalb, ob der "
-"Eintrag akzeptiert wurde."
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-"Diese Konfigurationsseite erlaubt Ihnen die Einrichtung eines "
-"Telefonsystemdienstes (PBX), der Anrufe über mehrere Google- und SIP-Konten "
-"(wie Sipgate, SipSorcery und Betamax) erlaubt. Sie können diese Konten für "
-"viele SIP-Geräte verwenden. Beachten Sie, dass Google-, SIP- und lokale "
-"Benutzer-Konten in den Abschnitten \"Google-Konten\", \"SIP-Konten\" und "
-"\"Benutzerkonten\" eingerichtet werden. Sie müssen mindestens ein "
-"Benutzerkonto für diesen PBX vorsehen und dann ein SIP-Gerät oder Softphone "
-"für die Benutzung dieses Kontos einrichten, damit Sie Anrufe mit Ihren "
-"Google-/SIP-Konten tätigen oder empfangen können. Wenn Sie mehr als ein "
-"Google- / SIP-Konto eingerichtet haben, sollten Sie auf der Seite "
-"\"Anrufweiterleitung\" einrichten, wie diese Anrufe behandelt werden. Wenn "
-"Sie Ihr PBX von irgendwo auf der Welt nutzen wollen, schauen Sie auf den "
-"Abschnitt \"Benutzung aus der Ferne\" auf der Seite \"Erweiterte "
-"Einstellungen\". "
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-"Dies ist der Name, den der VoIP-Server verwenden wird, um sich selbst bei "
-"der Registrierung beim VoIP-Dienstanbieter zu  identifizieren. Einige "
-"Anbieter verlangen, dass dies ein spezieller Begriff ist, der einem Hardware-"
-"SIP-Gerät entspricht."
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-"Hier geben Sie an, welche Google-/SIP-Konten für welche Ländervorwahlen "
-"benutzt werden sollen, welche Nutzer welche Konten verwenden dürfen, wie "
-"Anrufe weitergeleitet werden, welche Nummern mit Password in diesen PBX "
-"kommen, und welche Nummern ausgeschlossen werden."
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-"Hier stellen Sie Ihre Google (Talk und Voice) Konten ein, um sie für "
-"abgehende und ankommende Anrufe nutzen zu können (Voice Chat und Telefon-"
-"Anrufe). Bitte tätigen Sie wenigstens einen Sprach-Anruf mit dem Google-Talk-"
-"Plugin, das über das GMail-Interface zu installieren ist, und melden Sie "
-"sich dann überall aus Ihrem Konto ab. Klicken Sie auf \"Hinzufügen\" um so "
-"viele Konten hinzuzufügen, wie Sie wollen."
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-"Hier stellen Sie Ihre SIP (VoIP) Konten, wie Sipgate, SipSorcery, die "
-"populären Betamax-Anbieter, und alle anderen Anbieter mit SIP-Einstellungen "
-"ein, um sie für abgehende und ankommende Anrufe nutzen zu können (SIP uri "
-"und Telefon-Anrufe). Klicken Sie auf \"Hinzufügen\" um so viele Konten "
-"hinzuzufügen, wie Sie wollen."
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-"Diese Option sollte auf \"Ja\" gesetzt werden, wenn Sie eine DID (reale "
-"Telefonnummer) haben, die mit diesem SIP-Konto verknüpft ist, oder wenn Sie "
-"SIP-Anrufe über diesen Anbieter empfangen wollen."
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-"Dieser Abschnitt enthält Einstellungen, die unter normalen Umständen nicht "
-"geändert werden müssen. Zusätzlich konnen Sie hier Ihr System für die "
-"Verwendung mit entfernten SIP-Geräten einrichten und Probleme bei der "
-"Tonqualität beheben, indem Sie die Festlegung von QoS-Regeln aktivieren."
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-"Verwenden Sie eine vier- bis fünfstellige Nummer als Benutzernamen, wenn Sie "
-"normale Telefone mit ATA an dieses System anschließen (damit diese Namen "
-"über deren Zifferntastatur eingegeben werden können)."
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-"Dieses Konto für abgehende Anrufe verwenden, wie im Abschnitt "
-"\"Anrufweiterleitung\" eingestellt."
-
-msgid "Use this account to make outgoing calls."
-msgstr "Dieses Konto für abgehende Anrufe verwenden."
-
-msgid "User Accounts"
-msgstr "Benutzerkonten"
-
-msgid "User Agent String"
-msgstr "Benutzeridentifikation (User Agent)"
-
-msgid "User Name"
-msgstr "Benutzername"
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr "Verwendet für abgehende Anrufe eingerichtete Dienstanbieter"
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-"Wenn jemand einen Voice-Chat mit Ihrem GTalk-Konto oder die GVoice-Nummer "
-"(falls Sie Google Voice haben) anruft, wird der Anruf an jeden Benutzer "
-"weiter geleitet, der Online ist (mit SIP-Gerät oder Softphone) und den Anruf "
-"empfangen darf. Wenn Sie Google Voice haben, müssen Sie in Ihre GVoice-"
-"Einstellungen gehen und Anrufe zu Google Chat weiter leiten, damit Sie "
-"Anrufe auf Ihre GVoice-Nummer empfangen können. Bei Problemen mit dem "
-"Empfang von Anrufen über GVoice, experimentieren Sie mit der Option "
-"\"Anrufprüfung\" in den GVoice-Einstellungen. Stellen Sie schließlich "
-"sicher, dass kein anderer Client mit diesem Konto Online ist (z.B. Browser "
-"in GMail, Google Talk App mobil oder auf PC), denn das könnte Einfluss haben."
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-"Wenn Ihr Passwort gespeichert wird, verschwindet es aus diesem Feld und wird "
-"zu Ihrem Schutz nicht angezeigt. Ein vorher gespeichertes Passwort wird nur "
-"geändert, wenn Sie ein geändertes Passwort eingeben."
-
-msgid "Yes"
-msgstr "Ja"
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-"Sie können hier einen Klarnamen angeben, der als Name des Anrufers erscheint."
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-"Sie können Ihre SIP-Geräte/Softphones mit diesem System auch von einem "
-"entfernten Ort aus benutzen, so lange Ihnen Ihr Internet-Dienstanbieter eine "
-"öffentliche IP-Adresse zuweist. Sie können andere lokale Benutzer kostenlos "
-"anrufen (z.B. andere Analog-Telefon-Adapter (ATA)) und Ihre VoIP-Anbieter "
-"für Anrufe verwenden, als ob Sie am lokalen PBX angeschlossen wären. Nach "
-"der Einrichtung dieses Tabs gehen Sie zu den Benutzereinstellungen zurück "
-"und schauen Sie nach den neuen Einstellungen für Server und Port, die Sie an "
-"den entfernten SIP-Geräten vornehmen müssen. Bitte beachten Sie, dass Sie "
-"NAT/Portweiterleitung auf dem Router/Gateway einrichten müssen, falls dieser "
-"PBX nicht auf Ihrem Router/Gateway läuft. Bitte leiten Sie die unten "
-"angegebenen Ports (SIP-Port und RTP-Bereich) auf die IP-Adresse dieses PBX "
-"weiter."
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-"Ihre PIN verschwindet beim Speichern aus diesem Feld und wird zu Ihrem "
-"Schutz nicht angezeigt. Eine vorher gespeicherte PIN wird nur geändert, wenn "
-"Sie eine geänderte PIN eingeben. Sie können die PIN leer lassen, aber denken "
-"Sie an die Konsequenzen für die Sicherheit."
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-"Ihr Passwort verschwindet beim Speichern und wird zu Ihrem Schutz nicht "
-"angezeigt. Es wird nur geändert, wenn Sie ein anderes Passwort eingeben."
-
-#~ msgid ""
-#~ "Designate numbers that are allowed to call through this system and which "
-#~ "user's privileges it will have."
-#~ msgstr ""
-#~ "Nummern auswählen, die durch dieses System anrufen können, und deren "
-#~ "Benutzerrechte einstellen"
-
-#~ msgid ""
-#~ "Pick a random port number between 6500 and 9500 for the service to listen "
-#~ "on. Do not pick the standard 5060, because it is often subject to brute-"
-#~ "force attacks. When finished, (1) click \"Save and Apply\", and (2) click "
-#~ "the \"Restart VoIP Service\" button above. Finally, (3) look in the \"SIP "
-#~ "Device/Softphone Accounts\" section for updated Server and Port settings "
-#~ "for your SIP Devices/Softphones."
-#~ msgstr ""
-#~ "Wählen Sie eine zufällige Portnummer zwischen 6500 und 9000 für den Dienst "
-#~ "aus. Nehmen Sie nicht die standardmäßige 5060, weil sie oft attackiert wird. "
-#~ "Wenn fertig (1) klicken Sie auf \"Speichern und Anwenden\" und (2) auf \"VoIP-"
-#~ "Dienst neu starten\" oben. Schließlich (3) sehen Sie im Abschnitt \"SIP-Geräte"
-#~ "/Softphone-Konten\" nach aktualisierten Einstellungen für Ihre SIP-"
-#~ "Geräte/Softphones."
-
-#~ msgid ""
-#~ "You can enter your domain name, external IP address, or dynamic domain "
-#~ "name here Please keep in mind that if your IP address is dynamic and it "
-#~ "changes your configuration will become invalid. Hence, it's recommended "
-#~ "to set up Dynamic DNS in this case."
-#~ msgstr ""
-#~ "Sie können Ihren Domänennamen, externe IP-Adresse, oder dynamischen "
-#~ "Domänennamen hier angeben.Bitte beachten Sie, dass Ihre Konfiguration "
-#~ "ungältig wird, wenn Sie eine dynamische IP-Adresse besitzen und sich diese "
-#~ "ändert. Für diesen Fall wird deshalb die Einrichtung von dnamischem DNS "
-#~ "empfohlen."
-
-#~ msgid "Account Status"
-#~ msgstr "Konto-Status"
-
-#~ msgid "Account Status Message"
-#~ msgstr "Konto-Status Meldung"
-
-#~ msgid "Domain Name/Dynamic Domain Name"
-#~ msgstr "DNS Name (auch dynamisch möglich)"
diff --git a/applications/luci-app-pbx/po/el/pbx.po b/applications/luci-app-pbx/po/el/pbx.po
deleted file mode 100644
index 717e2563b4..0000000000
--- a/applications/luci-app-pbx/po/el/pbx.po
+++ /dev/null
@@ -1,493 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2012-03-31 15:41+0200\n"
-"Last-Translator: Vasilis <acinonyx@openwrt.gr>\n"
-"Language-Team: none\n"
-"Language: el\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-"X-Generator: Pootle 2.0.4\n"
-
-msgid "Advanced Settings"
-msgstr ""
-
-msgid "Available"
-msgstr ""
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-
-msgid "Away"
-msgstr ""
-
-msgid "Blacklisted Numbers"
-msgstr ""
-
-msgid "Call Routing"
-msgstr ""
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr ""
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr "Μην Ενοχλείτε"
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Email"
-msgstr ""
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr ""
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr ""
-
-msgid "Enable Outgoing Calls"
-msgstr ""
-
-msgid "Enabled"
-msgstr "Ενεργοποιημένο"
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr ""
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr "Πλήρες Όνομα"
-
-msgid "General Settings"
-msgstr "Γενικές Ρυθμίσεις"
-
-msgid "Google Accounts"
-msgstr "Λογαριασμοί Google"
-
-msgid "Google Talk Status"
-msgstr ""
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr "Λογαριασμοί Google Voice/Talk"
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr "Εισερχόμενες Κλήσεις"
-
-msgid "Insert QoS Rules"
-msgstr ""
-
-msgid "Makes Outgoing Calls"
-msgstr ""
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr "Όχι"
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr "Εξερχόμενες Κλήσεις"
-
-msgid "PBX Main Page"
-msgstr ""
-
-msgid "PBX Service Status"
-msgstr ""
-
-msgid "PIN"
-msgstr "PIN"
-
-msgid "Password"
-msgstr "Κωδικός πρόσβασης"
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr ""
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr ""
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr "Λογαριασμοί SIP"
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr ""
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr ""
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr ""
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-#~ msgid "Account Status"
-#~ msgstr "Κατάσταση Λογαριασμού"
-
-#~ msgid "Account Status Message"
-#~ msgstr "Μήνυμα Κατάστασης Λογαριασμού"
diff --git a/applications/luci-app-pbx/po/en/pbx.po b/applications/luci-app-pbx/po/en/pbx.po
deleted file mode 100644
index 8b995e1a39..0000000000
--- a/applications/luci-app-pbx/po/en/pbx.po
+++ /dev/null
@@ -1,502 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-
-msgid "Advanced Settings"
-msgstr "Advanced Settings"
-
-msgid "Available"
-msgstr "Available"
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-
-msgid "Away"
-msgstr "Away"
-
-msgid "Blacklisted Numbers"
-msgstr "Blacklisted Numbers"
-
-msgid "Call Routing"
-msgstr "Call Routing"
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr "Call-through Numbers"
-
-msgid "Copy-paste large lists of numbers here."
-msgstr "Copy-paste large lists of numbers here."
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr "Do Not Disturb"
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr "Dynamic List of Blacklisted Numbers"
-
-msgid "Email"
-msgstr "Email"
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr "Enable Incoming Calls (Register via SIP)"
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr ""
-
-msgid "Enable Outgoing Calls"
-msgstr "Enable Outgoing Calls"
-
-msgid "Enabled"
-msgstr "Enabled"
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr "External SIP Port"
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr "Full Name"
-
-msgid "General Settings"
-msgstr "General Settings"
-
-msgid "Google Accounts"
-msgstr "Google Accounts"
-
-msgid "Google Talk Status"
-msgstr ""
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr "Google Voice/Talk Accounts"
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr "Incoming Calls"
-
-msgid "Insert QoS Rules"
-msgstr "Insert QoS Rules"
-
-msgid "Makes Outgoing Calls"
-msgstr "Makes Outgoing Calls"
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr "No"
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr "Outbound Proxy"
-
-msgid "Outgoing Calls"
-msgstr "Outgoing Calls"
-
-msgid "PBX Main Page"
-msgstr "PBX Main Page"
-
-msgid "PBX Service Status"
-msgstr ""
-
-msgid "PIN"
-msgstr "PIN"
-
-msgid "Password"
-msgstr "Password"
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr "Port Setting for SIP Devices"
-
-msgid "Providers Used for Outgoing Calls"
-msgstr "Providers Used for Outgoing Calls"
-
-msgid "QoS Settings"
-msgstr "QoS Settings"
-
-msgid "RTP Port Range End"
-msgstr "RTP Port Range End"
-
-msgid "RTP Port Range Start"
-msgstr "RTP Port Range Start"
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr "Receives Incoming Calls"
-
-msgid "Remote Usage"
-msgstr "Remote Usage"
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr "SIP Accounts"
-
-msgid "SIP Device/Softphone Accounts"
-msgstr "SIP Device/Softphone Accounts"
-
-msgid "SIP Provider Accounts"
-msgstr "SIP Provider Accounts"
-
-msgid "SIP Realm (needed by some providers)"
-msgstr "SIP Realm (needed by some providers)"
-
-msgid "SIP Server/Registrar"
-msgstr "SIP Server/Registrar"
-
-msgid "SIP Server/Registrar Port"
-msgstr "SIP Server/Registrar Port"
-
-msgid "Server Setting"
-msgstr "Server Setting"
-
-msgid "Server Setting for Local SIP Devices"
-msgstr "Server Setting for Local SIP Devices"
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr "Server Setting for Remote SIP Devices"
-
-msgid "Service Status"
-msgstr "Service Status"
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr "Space-Separated List of Blacklisted Numbers"
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr "Specify numbers individually here. Press enter to add more numbers."
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-
-msgid "Use this account to make outgoing calls."
-msgstr "Use this account to make outgoing calls."
-
-msgid "User Accounts"
-msgstr "User Accounts"
-
-msgid "User Agent String"
-msgstr "User Agent String"
-
-msgid "User Name"
-msgstr "User Name"
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr "Yes"
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr "You can specify a real name to show up in the Caller ID here."
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-#~ msgid "Account Status"
-#~ msgstr "Account Status"
-
-#~ msgid "Account Status Message"
-#~ msgstr "Account Status Message"
-
-#~ msgid "Domain Name/Dynamic Domain Name"
-#~ msgstr "Domain Name/Dynamic Domain Name"
-
-#~ msgid "Enable Incoming Calls (See Status, Message below)"
-#~ msgstr "Enable Incoming Calls (See Status, Message below)"
-
-#~ msgid "Service Control and Connection Status"
-#~ msgstr "Service Control and Connection Status"
diff --git a/applications/luci-app-pbx/po/es/pbx.po b/applications/luci-app-pbx/po/es/pbx.po
deleted file mode 100644
index 8071b61f08..0000000000
--- a/applications/luci-app-pbx/po/es/pbx.po
+++ /dev/null
@@ -1,677 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2014-06-15 13:15+0200\n"
-"Last-Translator: José Vicente <josevteg@gmail.com>\n"
-"Language-Team: none\n"
-"Language: es\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Advanced Settings"
-msgstr "Configuración avanzada"
-
-msgid "Available"
-msgstr "Disponible"
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr "Usar sólo caracteres alfanuméricos, espacio, coma y punto."
-
-msgid "Away"
-msgstr "No disponible"
-
-msgid "Blacklisted Numbers"
-msgstr "Lista negra"
-
-msgid "Call Routing"
-msgstr "Enrutado de llamadas"
-
-msgid "Call-back Numbers"
-msgstr "Números de call-back"
-
-msgid "Call-back Provider"
-msgstr "Proveedor de call-back"
-
-msgid "Call-through Numbers"
-msgstr "Números call-through"
-
-msgid "Copy-paste large lists of numbers here."
-msgstr "Pegue aquí grandes listas de números."
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-"Listar los números a los que se permitirá llamar desde este sistema y qué "
-"privilegios de usuario tendrán."
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-"Listar los números a los que el sistema colgará y volverá a llamar, qué "
-"proveedor se usará para llamarles y qué privilegios de usuario se les dará."
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr "Marca el resto de números en cualquier lugar"
-
-msgid "Do Not Disturb"
-msgstr "No molestar"
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr "Dominio/Dirección IP/Dominio dinámico"
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr "Lista dinámica de números en lista negra"
-
-msgid "Email"
-msgstr "e-mail"
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr "Permitir llamadas entrantes (registrar vía SIP)"
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr "Permitir llamadas entrantes (ver estado abajo)"
-
-msgid "Enable Outgoing Calls"
-msgstr "Permitir llamadas salientes"
-
-msgid "Enabled"
-msgstr "Activado"
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-"Proveedor VoIP para callbacks en formato nombredeusuario@algun.nombre.host, "
-"tal y como se detalla arriba en \"Llamadas salientes\". Puede copiar y pegar "
-"los proveedores desde ahí. Las entradas no válidas, incluyendo a proveedores "
-"no habilitados para llamadas saliente, serán rechazadas sin mostrar aviso."
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-"Números de teléfono de los que se reclina la llamada automáticamente. Es "
-"posible que tenga que omitir el código de país y ceros precedentes, pero "
-"experimente para asegurarse que bloquea los números correctamente."
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-"Ponga esta IP (o IP:puerto) en el parámetro Servidor/Registrador de los "
-"dispositivos SIP que usará SOLO localmente y nunca desde una posición remota."
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-"Ponga este nombre de máquina en el parámetro Servidor/Registrador de los "
-"dispositivos SIP que usará desde posiciones remotas (también vale "
-"localmente)."
-
-msgid "External SIP Port"
-msgstr "Puerto externo SIP"
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-"Para cada proveedor al que se habilita a hacer llamadas entrantes puede "
-"restringir a qué usuarios llamar. Si se deja vacío el sistema indicará que "
-"llamará a todos los usuarios que puedan recibir llamadas entrantes. Los "
-"nombres de usuario no válidos se rechazarán sin aviso. Estos nombres de "
-"usuario hacen ignorar la configuración de usuario de no recibir llamadas. De "
-"esta manera puede hacer que a ciertos usuarios sólo les llamen ciertos "
-"proveedores. Puede separar los nombres con espacios o poniéndolos en líneas "
-"diferentes."
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-"Para cada usuario habilitado a hacer llamadas salientes puede restringir qué "
-"proveedores usar. Por defecto todos los usuarios pueden usar a todos los "
-"proveedores. Para mostrarse en la lista el usuario debe poder hacer llamadas "
-"salientes (ver página \"Cuentas de usuario\"). Ponga los proveedores en "
-"formato username@some.host.name igual que se listan en \"Llamadas salientes"
-"\" arriba. Los nombres no válidos se rechazarán sin aviso.Puede separar los "
-"nombres con espacios o poniéndolos en líneas diferentes."
-
-msgid "Full Name"
-msgstr "Nombre completo"
-
-msgid "General Settings"
-msgstr "Configuración general"
-
-msgid "Google Accounts"
-msgstr "Cuentas en google"
-
-msgid "Google Talk Status"
-msgstr "Estado de Google Talk"
-
-msgid "Google Talk Status Message"
-msgstr "Mensaje de estado de Google Talk"
-
-msgid "Google Voice/Talk Accounts"
-msgstr "Cuentas Google Voice/Talk"
-
-msgid "Hang-up Delay"
-msgstr "Retraso para descolgar"
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-"Configure una cuenta SIP que usará para conectar con este servicio. Úsela "
-"tanpo en un adaptador de telefonía analógico (ATA) o en un programa SIP como "
-"CSipSimple, Linphone, o Sipdroid para smartphones, o Ekiga, Linphone, o X-"
-"Lite para ordenadores. Por defecto, todas las cuentas SIP sonarán a la vez "
-"si se hace una llamada desde una de las cuentas de su proveedor de VoIP o "
-"números GV."
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-"Cuánto esperar antes de descolgar. Si el proveedor que usas para marcar "
-"automáticamente desvía a un correo de voz puedes ajustar este valor con un "
-"retraso que permitirá descolgar antes de que se desvíe la llamada y se "
-"facture."
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-"Si la configuración Servidor/Registrador en %s o %s no le funciona, prueba a "
-"poner %s o %s e introduzca este número de puerto en un campo separado que "
-"especifique el número de puerto del Servidor/Registrador. Algunos "
-"dispositivos tienen una configuración extraña que muestra este puerto desde "
-"el que el SIP origina peticiones en el mismo dispositivo SIP (el puerto "
-"asociado). El puerto que está configurando aquí NO es este puerto asociado "
-"sino el puerto en el que el servicio escucha."
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-"Si nota saltos o retrasos en el audio mientras realiza descargas puede "
-"querer activar QoS. QoS prioriza el tráfico a y desde su red para ciertos "
-"puertos y direcciones IP mejorando la latencia y el rendimiento del sonido "
-"en dicho caso. Al activarlo el PBX creará una regla QoS para este servicio, "
-"pero deberá rellenar en la página de configuración de QoS (Red/QoS) otros "
-"parámetros necesarios como la velocidad de subida y la de bajada."
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-"Si tiene más de una cuenta para hacer llamadas salientes, debe introducir "
-"una lista de números de teléfono y/o prefijos para cada proveedor. Los "
-"prefijos no válidos se rechazarán sin aviso y solo son caracteres válidos "
-"0-9, X, Z, N, #, *, y +. La letra X equivale a 0-9, Z a 1-9 y N a 2-9. Por "
-"ejemplo para hacer llamadas a Alemania con su proveedor debe introducir 49. "
-"Para hacer llamadas a Estados Unidos 1NXXNXXXXXX. Si uno de sus proveedores "
-"puede hacer llamadas locales a un código de área como el 646 de Nueva York "
-"debe introducir 646NXXXXXX para ese proveedor. Debería dehar una cuenta con "
-"una lista vacía para que haga las llamadas por defecto en caso de que ningún "
-"prefijo encaje. El sistema reemplazará automáticamente la lista vacía con el "
-"mensaje de que el proveedor marca todos los números que no estén en los "
-"prefijos de otros proveedores. Sea todo lo específico que pueda (ej. "
-"1NXXNXXXXXX es mejor que 1). Todos los códigos internaciones de marcado se "
-"descartan (ej. 00, 011, 010, 0011). Las entradas pueden ser una lista "
-"separada por espacios y/o cambios de línea."
-
-msgid "Incoming Calls"
-msgstr "Llamadas entrantes"
-
-msgid "Insert QoS Rules"
-msgstr "Reglas QoS"
-
-msgid "Makes Outgoing Calls"
-msgstr "Realizar llamadas salientes"
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr "NOTA: Sin cuentas configuradas de Google o porveedor SIP."
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-"NOTA: Sin cuentas configuradas de Google o porveedor SIP para llamadas "
-"entrantes."
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-"NOTA: Sin cuentas configuradas de Google o porveedor SIP para llamadas "
-"salientes."
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr "NOTA: Sin cuentas locales configuradas."
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr "NOTA: Sin cuentas locales habilitadas para llamadas saientes."
-
-msgid "No"
-msgstr "No"
-
-msgid "Number of Seconds to Ring"
-msgstr "Número de segundos a sonar"
-
-msgid "Outbound Proxy"
-msgstr "Proxy saliente"
-
-msgid "Outgoing Calls"
-msgstr "Llamadas salientes"
-
-msgid "PBX Main Page"
-msgstr "Página principal de PBX"
-
-msgid "PBX Service Status"
-msgstr "Estado del servicio PBX"
-
-msgid "PIN"
-msgstr "PIN"
-
-msgid "Password"
-msgstr "Contraseña"
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-"Escoge un número de puerto aleatorio entre 6500 y 9500 para el servicio. No "
-"elijas el estándar 5060 ya que es objeto, a menudo, de ataques por fuerza "
-"bruta. Cuando hayas terminado pulsa en \"Salvar y aplicar\" y busca en la "
-"sección \"Cuentas SIP del dispositivo/softphone\" el puerto actual para tus "
-"dispositivos/softphones SIP."
-
-msgid "Port Setting for SIP Devices"
-msgstr "Configuración de puerto para dispositivos SIP"
-
-msgid "Providers Used for Outgoing Calls"
-msgstr "Proveedores usados para llamadas salientess"
-
-msgid "QoS Settings"
-msgstr "Configuración de QoS"
-
-msgid "RTP Port Range End"
-msgstr "Fin del rango de puertos RTP"
-
-msgid "RTP Port Range Start"
-msgstr "Inicio del rango de puertos RTP"
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-"El tráfico RTP es el que lleva los paquetes de voz. Este es el inicio del "
-"rango de puertos que se usará para comunicaciones RTP. Suele ser correcto "
-"dejar el valor por defecto."
-
-msgid "Receives Incoming Calls"
-msgstr "Recibe llamadas entrantes"
-
-msgid "Remote Usage"
-msgstr "Uso remoto"
-
-msgid "Rings users enabled for incoming calls"
-msgstr "Llama a usuarios habilitados a recibir llamadas"
-
-msgid "SIP Accounts"
-msgstr "Cuentas SIP"
-
-msgid "SIP Device/Softphone Accounts"
-msgstr "Dispositivo SIP/Cuentas Softphone"
-
-msgid "SIP Provider Accounts"
-msgstr "Cuentas del proveedor SIP"
-
-msgid "SIP Realm (needed by some providers)"
-msgstr "Ámbito SIP (necesario para algunos proveedores)"
-
-msgid "SIP Server/Registrar"
-msgstr "Servidor/Registrador del SIP"
-
-msgid "SIP Server/Registrar Port"
-msgstr "Puerto del Servidor/Registrador del SIP"
-
-msgid "Server Setting"
-msgstr "Configuración del servidor"
-
-msgid "Server Setting for Local SIP Devices"
-msgstr "Dispositivos SIP locales"
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr "Dispositivos SIP remotos"
-
-msgid "Service Status"
-msgstr "Estado del servicio"
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-"Segundos que se llamará a los usuarios antes de colgar o pasar a correo voz "
-"(si el correo voz está instalado y habilitado)."
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr "Lista negra (separar números con espacios)"
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr "Números individuales. Pulse enter para añadir más."
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-"Especifica números individualmente. Pulsa enter para añadir más. Tendrás que "
-"experimentar con qué códigos de país y área necesitas añadir al número."
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-"Estos números podrán llamar con los proveedores de este usuario. Los nombres "
-"de usuario no válidos se descartan sin aviso. Por favor, verifique que los "
-"números se aceptan."
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-"Aquí puede configurar un servicio de sistema telefónico (PBX) que le "
-"permitirá hacer llamadas por múltiples cuentas Google y SIP (como Sipgate, "
-"SipSorcery, and Betamax) y compartirlas entre muchos dispositivos SIP. Tenga "
-"en cuenta que las cuentas Google, SIP y locales deben configurarse en "
-"subsecciones diferentes. Debe añadir al menos una cuenta de usuarioa este "
-"PBX y configurar un dispositivo SIP o softphone para usarla para recibir las "
-"llamadas de sus cuentas Google/SIP. Configurar múltiples usuarios le "
-"permitirá hacer llamadas gratuitas entre los usuarios y compartir las "
-"cuentas Google/SIP configuradas. Si tiene más de una cuenta Google/SIP "
-"configurada tendrá que configurar cómo se enrutan en la página \"Enrutado de "
-"llamadas\". Si está interesado en usar su PBX desde cualquier sitio del "
-"mundo puede visitar la sección \"Uso remoto\" en la página \"Configuración "
-"avanzada\"."
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-"Nombre del servidor VoIP que usará para identificarse cuando se registre en "
-"proveedores de VoIP (SIP). Algunos requieres que sea una cadena específica a "
-"una dispositivo hardware."
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-"Indique las cuentas Google/SIP que usará para llamar a qué códigos de país/"
-"zona, qué usuarios pueden usuarios pueden usar qué cuentas SIP/Google y cómo "
-"se enrutan las llamadas entrantes, qué números pueden entrar en esta PBX con "
-"una contraseña y qué números están en lista negra."
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-"Configure sus cuentas Google (Talk y Voz) para empezar a usarlas para hacer "
-"y recibir llamadas (chat de voz y teléfono real). Haga al menos una llamada "
-"de voz con el plugin de Google Talk (instalable desde GMail) y desconéctese "
-"de la cuenta en cualquier otro sitio."
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-"Configure sus cuentas SIP (VoIP) como Sipgate, SipSorcery, los popular "
-"proveedores Betamax y cualquier otro proveedor para empezar a usarlos para "
-"hacer y recibir llamadas (uri SIP y teléfono real)."
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-"Debería ser \"Sí\" si tiene un DID (teléfono real) asociado a esta cuenta "
-"SIP o quiere recibir llamads uri SIP de este proveedor."
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-"Algunos de estos parámetros no suele ser necesario cambiarlos. Además puede "
-"configurar su sistema para usar con dispositivos SIP remotos y resolver "
-"problemas de calidad de llamada habilitando algunas reglas QoS."
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-"Use nombre de usuario númericos (cuatro o cinco dígitos) si conecta a "
-"teléfonos normales con ATAs a este sistema (para que puedan marcar números "
-"de usuario)."
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-"Cuenta para llamadas salientes como se configura en la sección \"Enrutado de "
-"llamadas\"."
-
-msgid "Use this account to make outgoing calls."
-msgstr "Cuenta para llamadas salientes."
-
-msgid "User Accounts"
-msgstr "Cuentas de usuario"
-
-msgid "User Agent String"
-msgstr "Cadena \"User Agent\""
-
-msgid "User Name"
-msgstr "Nombre de usuario"
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr "Usar proveedores habilitados para llamadas salientes"
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-"Cuando alguien inicia un chat de voz con su cuenta de GTalk o llame al "
-"número de GVoice (si tiene Google Voice) la llamada se transferirá a "
-"cualquier usuario que esté conectado (registrado usando un dispositivo SIP o "
-"softphone) y se le permitirá recibir la llamada. Si tiene Google Voice debe "
-"ir a la configuración de GVoice y traspasar las llamadas a Google chat para "
-"recibir las hechas a si número de GVoice. Si tiene problemas recibiendo "
-"llamadas de GVoice pruebe con la opción \"Call Screening\" en la "
-"configuración de GVoice. Asegúrese de que ningún otro cliente esté conectado "
-"con esta cuenta (navegador en gmail, o una aplicación para móvil o "
-"escritorio) ya que podría interferir."
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-"Cuando se salve su contraseña desaparece de este campo y no se muestra para "
-"su seguridad. La contraseña sólo se podrá cambiar si introduce un valor "
-"diferente al salvado."
-
-msgid "Yes"
-msgstr "Sí"
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-"Puedes introducir el nombre de dominio, dirección IP external o nombre "
-"dinámino aquí. Lo mejor es introducir una dirección IP estática. Si la "
-"dirección es dinámica la configuración sería inválida cuando cambiase. En "
-"estos casos es recomendable configurar Dynamic DNS e introducir tu nombre de "
-"host Dynamic DNS. Puedes instalar y configurar Dynamic DNS con el paquete "
-"luci-app-ddns."
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr "Nombre real a mostrar en el \"Caller ID\"."
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-"Puede usar sus dispositivos SIP/softphones con este sistema desde una "
-"ubicación remota mientras su proveedor de internet le dé una dirección IP "
-"pública. Podrá llamar a usuarios locales gratis (ej. otros adaptadores de "
-"teléfonos analógicos) y podrá usar sus proveedores de VoIP para hacer "
-"llamadas como si estuviese en su PBX local. Tras configurar esta pestaña "
-"vuelva a la configuración de usuarios y veo el nuevo servidor y puerto que "
-"debe configurar en sus dispositivos SIP remotos. Tenga en cuenta que si este "
-"PBX no funciona en su router/pasarela, tendrá que configurar el traspaso de "
-"puertos (NAT) en su router/pasarela. Traspase los puertos indicados (Puerto "
-"SIP y rango RTP) hacia la dirección IP del dispositivo en que corre esta PBX."
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-"Su PIN desaparecerá cuando se salve para su protección. Se cambiará solo "
-"cuando introduzca un valor diferente al salvado. No se puede dejar el PIN "
-"vacío."
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-"Su contraseña desaparecerá cuando se salve para su protección. Sólo se puede "
-"cambiar si entra un valor diferente al salvado."
-
-#~ msgid ""
-#~ "Designate numbers that are allowed to call through this system and which "
-#~ "user's privileges it will have."
-#~ msgstr ""
-#~ "Números a los que se permite llamar por este sistema y privilegios de "
-#~ "usuario."
-
-#~ msgid ""
-#~ "Pick a random port number between 6500 and 9500 for the service to listen "
-#~ "on. Do not pick the standard 5060, because it is often subject to brute-"
-#~ "force attacks. When finished, (1) click \"Save and Apply\", and (2) click "
-#~ "the \"Restart VoIP Service\" button above. Finally, (3) look in the \"SIP "
-#~ "Device/Softphone Accounts\" section for updated Server and Port settings "
-#~ "for your SIP Devices/Softphones."
-#~ msgstr ""
-#~ "Puerto aleatorio entre 6500 y 9500 en el que escuche el servicio. No elija "
-#~ "el estándar 5060 porque es susceptible de ataques por fuerza bruta. Cuando "
-#~ "termine (1) pulsa \"Salvar y aplicar\" y (2) pulse \"Rearrancar servicio VoIP\". "
-#~ "Finalmente (3) busque en la sección \"Dispositivo SIP/Cuentas softphone\" la "
-#~ "configuración del puerto."
-
-#~ msgid ""
-#~ "You can enter your domain name, external IP address, or dynamic domain "
-#~ "name here Please keep in mind that if your IP address is dynamic and it "
-#~ "changes your configuration will become invalid. Hence, it's recommended "
-#~ "to set up Dynamic DNS in this case."
-#~ msgstr ""
-#~ "Nombre de dominio, dirección IP externa o nombre de dominio dinámico. Si su "
-#~ "dirección IP es dinámica y cambia su configuración podría resultar no "
-#~ "válida. Se recomienda el uso de DNS dinámico en estos casos."
diff --git a/applications/luci-app-pbx/po/fr/pbx.po b/applications/luci-app-pbx/po/fr/pbx.po
deleted file mode 100644
index 971a696488..0000000000
--- a/applications/luci-app-pbx/po/fr/pbx.po
+++ /dev/null
@@ -1,484 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n > 1);\n"
-
-msgid "Advanced Settings"
-msgstr ""
-
-msgid "Available"
-msgstr ""
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-
-msgid "Away"
-msgstr ""
-
-msgid "Blacklisted Numbers"
-msgstr ""
-
-msgid "Call Routing"
-msgstr ""
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr ""
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr ""
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Email"
-msgstr ""
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr ""
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr ""
-
-msgid "Enable Outgoing Calls"
-msgstr ""
-
-msgid "Enabled"
-msgstr ""
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr ""
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr ""
-
-msgid "General Settings"
-msgstr ""
-
-msgid "Google Accounts"
-msgstr ""
-
-msgid "Google Talk Status"
-msgstr ""
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr ""
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr ""
-
-msgid "Insert QoS Rules"
-msgstr ""
-
-msgid "Makes Outgoing Calls"
-msgstr ""
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr ""
-
-msgid "PBX Main Page"
-msgstr ""
-
-msgid "PBX Service Status"
-msgstr ""
-
-msgid "PIN"
-msgstr ""
-
-msgid "Password"
-msgstr ""
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr ""
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr ""
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr ""
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr ""
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr ""
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr ""
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
diff --git a/applications/luci-app-pbx/po/he/pbx.po b/applications/luci-app-pbx/po/he/pbx.po
deleted file mode 100644
index 2a458214df..0000000000
--- a/applications/luci-app-pbx/po/he/pbx.po
+++ /dev/null
@@ -1,484 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-
-msgid "Advanced Settings"
-msgstr ""
-
-msgid "Available"
-msgstr ""
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-
-msgid "Away"
-msgstr ""
-
-msgid "Blacklisted Numbers"
-msgstr ""
-
-msgid "Call Routing"
-msgstr ""
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr ""
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr ""
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Email"
-msgstr ""
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr ""
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr ""
-
-msgid "Enable Outgoing Calls"
-msgstr ""
-
-msgid "Enabled"
-msgstr ""
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr ""
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr ""
-
-msgid "General Settings"
-msgstr ""
-
-msgid "Google Accounts"
-msgstr ""
-
-msgid "Google Talk Status"
-msgstr ""
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr ""
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr ""
-
-msgid "Insert QoS Rules"
-msgstr ""
-
-msgid "Makes Outgoing Calls"
-msgstr ""
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr ""
-
-msgid "PBX Main Page"
-msgstr ""
-
-msgid "PBX Service Status"
-msgstr ""
-
-msgid "PIN"
-msgstr ""
-
-msgid "Password"
-msgstr ""
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr ""
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr ""
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr ""
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr ""
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr ""
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr ""
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
diff --git a/applications/luci-app-pbx/po/hu/pbx.po b/applications/luci-app-pbx/po/hu/pbx.po
deleted file mode 100644
index 2a458214df..0000000000
--- a/applications/luci-app-pbx/po/hu/pbx.po
+++ /dev/null
@@ -1,484 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-
-msgid "Advanced Settings"
-msgstr ""
-
-msgid "Available"
-msgstr ""
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-
-msgid "Away"
-msgstr ""
-
-msgid "Blacklisted Numbers"
-msgstr ""
-
-msgid "Call Routing"
-msgstr ""
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr ""
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr ""
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Email"
-msgstr ""
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr ""
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr ""
-
-msgid "Enable Outgoing Calls"
-msgstr ""
-
-msgid "Enabled"
-msgstr ""
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr ""
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr ""
-
-msgid "General Settings"
-msgstr ""
-
-msgid "Google Accounts"
-msgstr ""
-
-msgid "Google Talk Status"
-msgstr ""
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr ""
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr ""
-
-msgid "Insert QoS Rules"
-msgstr ""
-
-msgid "Makes Outgoing Calls"
-msgstr ""
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr ""
-
-msgid "PBX Main Page"
-msgstr ""
-
-msgid "PBX Service Status"
-msgstr ""
-
-msgid "PIN"
-msgstr ""
-
-msgid "Password"
-msgstr ""
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr ""
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr ""
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr ""
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr ""
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr ""
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr ""
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
diff --git a/applications/luci-app-pbx/po/it/pbx.po b/applications/luci-app-pbx/po/it/pbx.po
deleted file mode 100644
index 6da8e45d96..0000000000
--- a/applications/luci-app-pbx/po/it/pbx.po
+++ /dev/null
@@ -1,487 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2012-12-15 19:31+0200\n"
-"Last-Translator: claudyus <claudyus84@gmail.com>\n"
-"Language-Team: none\n"
-"Language: it\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Advanced Settings"
-msgstr "Opzioni avanzate"
-
-msgid "Available"
-msgstr ""
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-
-msgid "Away"
-msgstr ""
-
-msgid "Blacklisted Numbers"
-msgstr ""
-
-msgid "Call Routing"
-msgstr ""
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr ""
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr ""
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Email"
-msgstr ""
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr ""
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr ""
-
-msgid "Enable Outgoing Calls"
-msgstr ""
-
-msgid "Enabled"
-msgstr ""
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr ""
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr ""
-
-msgid "General Settings"
-msgstr ""
-
-msgid "Google Accounts"
-msgstr ""
-
-msgid "Google Talk Status"
-msgstr ""
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr ""
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr ""
-
-msgid "Insert QoS Rules"
-msgstr ""
-
-msgid "Makes Outgoing Calls"
-msgstr ""
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr ""
-
-msgid "PBX Main Page"
-msgstr ""
-
-msgid "PBX Service Status"
-msgstr ""
-
-msgid "PIN"
-msgstr ""
-
-msgid "Password"
-msgstr ""
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr ""
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr ""
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr ""
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr ""
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr ""
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr ""
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
diff --git a/applications/luci-app-pbx/po/ja/pbx.po b/applications/luci-app-pbx/po/ja/pbx.po
deleted file mode 100644
index 76199f4191..0000000000
--- a/applications/luci-app-pbx/po/ja/pbx.po
+++ /dev/null
@@ -1,493 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2012-04-21 07:57+0200\n"
-"Last-Translator: Kentaro <kentaro.matsuyama@gmail.com>\n"
-"Language-Team: none\n"
-"Language: ja\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=1; plural=0;\n"
-"X-Generator: Pootle 2.0.4\n"
-
-msgid "Advanced Settings"
-msgstr "詳細設定"
-
-msgid "Available"
-msgstr ""
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-
-msgid "Away"
-msgstr ""
-
-msgid "Blacklisted Numbers"
-msgstr ""
-
-msgid "Call Routing"
-msgstr ""
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr ""
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr ""
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Email"
-msgstr "Eメール"
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr ""
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr ""
-
-msgid "Enable Outgoing Calls"
-msgstr ""
-
-msgid "Enabled"
-msgstr ""
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr "外部SIPポート"
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr ""
-
-msgid "General Settings"
-msgstr "基本設定"
-
-msgid "Google Accounts"
-msgstr "Google アカウント"
-
-msgid "Google Talk Status"
-msgstr ""
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr "Google Voice/Talk アカウント"
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr ""
-
-msgid "Insert QoS Rules"
-msgstr "QoS ルール設定を有効にする"
-
-msgid "Makes Outgoing Calls"
-msgstr "発信を許可する"
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr "いいえ"
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr ""
-
-msgid "PBX Main Page"
-msgstr "PBX メインページ"
-
-msgid "PBX Service Status"
-msgstr ""
-
-msgid "PIN"
-msgstr "PIN"
-
-msgid "Password"
-msgstr "パスワード"
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr "QoS 設定"
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr "受信を許可する"
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr "SIP アカウント"
-
-msgid "SIP Device/Softphone Accounts"
-msgstr "SIP デバイス/ソフトフォン アカウント"
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr "サーバー設定"
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr ""
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr ""
-
-msgid "User Agent String"
-msgstr "ユーザーエージェント名"
-
-msgid "User Name"
-msgstr "ユーザー名"
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr "はい"
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-#~ msgid "Account Status"
-#~ msgstr "アカウントのステータス"
-
-#~ msgid "Account Status Message"
-#~ msgstr "アカウントステータス・メッセージ"
diff --git a/applications/luci-app-pbx/po/ms/pbx.po b/applications/luci-app-pbx/po/ms/pbx.po
deleted file mode 100644
index 23403f290f..0000000000
--- a/applications/luci-app-pbx/po/ms/pbx.po
+++ /dev/null
@@ -1,483 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-
-msgid "Advanced Settings"
-msgstr ""
-
-msgid "Available"
-msgstr ""
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-
-msgid "Away"
-msgstr ""
-
-msgid "Blacklisted Numbers"
-msgstr ""
-
-msgid "Call Routing"
-msgstr ""
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr ""
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr ""
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Email"
-msgstr ""
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr ""
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr ""
-
-msgid "Enable Outgoing Calls"
-msgstr ""
-
-msgid "Enabled"
-msgstr ""
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr ""
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr ""
-
-msgid "General Settings"
-msgstr ""
-
-msgid "Google Accounts"
-msgstr ""
-
-msgid "Google Talk Status"
-msgstr ""
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr ""
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr ""
-
-msgid "Insert QoS Rules"
-msgstr ""
-
-msgid "Makes Outgoing Calls"
-msgstr ""
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr ""
-
-msgid "PBX Main Page"
-msgstr ""
-
-msgid "PBX Service Status"
-msgstr ""
-
-msgid "PIN"
-msgstr ""
-
-msgid "Password"
-msgstr ""
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr ""
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr ""
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr ""
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr ""
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr ""
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr ""
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
diff --git a/applications/luci-app-pbx/po/no/pbx.po b/applications/luci-app-pbx/po/no/pbx.po
deleted file mode 100644
index 2a458214df..0000000000
--- a/applications/luci-app-pbx/po/no/pbx.po
+++ /dev/null
@@ -1,484 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-
-msgid "Advanced Settings"
-msgstr ""
-
-msgid "Available"
-msgstr ""
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-
-msgid "Away"
-msgstr ""
-
-msgid "Blacklisted Numbers"
-msgstr ""
-
-msgid "Call Routing"
-msgstr ""
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr ""
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr ""
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Email"
-msgstr ""
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr ""
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr ""
-
-msgid "Enable Outgoing Calls"
-msgstr ""
-
-msgid "Enabled"
-msgstr ""
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr ""
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr ""
-
-msgid "General Settings"
-msgstr ""
-
-msgid "Google Accounts"
-msgstr ""
-
-msgid "Google Talk Status"
-msgstr ""
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr ""
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr ""
-
-msgid "Insert QoS Rules"
-msgstr ""
-
-msgid "Makes Outgoing Calls"
-msgstr ""
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr ""
-
-msgid "PBX Main Page"
-msgstr ""
-
-msgid "PBX Service Status"
-msgstr ""
-
-msgid "PIN"
-msgstr ""
-
-msgid "Password"
-msgstr ""
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr ""
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr ""
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr ""
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr ""
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr ""
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr ""
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
diff --git a/applications/luci-app-pbx/po/pl/pbx.po b/applications/luci-app-pbx/po/pl/pbx.po
deleted file mode 100644
index 4e80a45815..0000000000
--- a/applications/luci-app-pbx/po/pl/pbx.po
+++ /dev/null
@@ -1,508 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2014-05-05 04:37+0200\n"
-"Last-Translator: piosl <sleczek.piotr@gmail.com>\n"
-"Language-Team: none\n"
-"Language: pl\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=3; plural=(n==1 ? 0 : n%10>=2 && n%10<=4 && (n%100<10 "
-"|| n%100>=20) ? 1 : 2);\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Advanced Settings"
-msgstr "Ustawienia zaawansowane"
-
-msgid "Available"
-msgstr "Dostępny"
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr "Unikaj znaków innych niż alfanumeryczne, spacja, przecinek i kropka."
-
-msgid "Away"
-msgstr "Oddalony"
-
-msgid "Blacklisted Numbers"
-msgstr "Numery na czarnej liście"
-
-msgid "Call Routing"
-msgstr "Przekierowanie połączeń"
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-# Chodzi tu o numery, przez które dzwoni się, aby obniżyć koszta połączeń zagranicznych. Jeśli ktoś ma pomysł na lepsze tłumaczenie, proszę zmienić. W sieci nie znalazłem.
-msgid "Call-through Numbers"
-msgstr "Numery pośredniczące"
-
-msgid "Copy-paste large lists of numbers here."
-msgstr "Wklej tu wielkie listy numerów."
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr "Nie przeszkadzać"
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr "Domena/adres IP/dynamiczna domena"
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr "Dynamiczna czarna lista numerów"
-
-msgid "Email"
-msgstr "E-mail"
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr "Włącz połączenia przychodzące (rejestruj przez SIP)"
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr "Włącz połączenia przychodzące (zobacz status poniżej)"
-
-msgid "Enable Outgoing Calls"
-msgstr "Włącz połączenia wychodzące"
-
-msgid "Enabled"
-msgstr "Włączone"
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-"Podaj numery telefonów, które powinny być automatycznie odrzucane. "
-"Prawdopodobnie powinieneś pominąć numer kierunkowy kraju i zera z przodu, "
-"ale samemu to przetestuj, aby upewnić się, że blokowanie działa prawidłowo "
-"dla Twojego położenia."
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-"Podaj to IP (lub parę IP:port) w ustawieniach serwera/rejestratora urządzeń "
-"SIP których będziesz używać WYŁĄCZNIE lokalnie i nigdy z zewnątrz."
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-"Podaj tę nazwę hosta (lub parę nazwa hosta:port) w ustawieniach serwera/"
-"rejestratora urządzeń SIP których będziesz używać z zewnątrz (będą też "
-"działać lokalnie)."
-
-msgid "External SIP Port"
-msgstr "Zewnętrzny port SIP"
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-"Dla każdego użytkownika z prawem wykonywania połączeń wychodzących możesz "
-"ograniczyć których operatorów mogą używać do tych połączeń. Domyślnie każdy "
-"użytkownik może używać dowolnego operatora. Użytkownik musi mieć prawo "
-"wykonywania połączeń wychodzących ustawione na stronie \"Konta użytkowników"
-"\", aby pojawić się na poniższej liście. Podaj operatorów VoIP w formacie "
-"nazwa.użytkownika@jakaś.nazwa.hosta, tak jak są wypisani w \"Połączeniach "
-"wychodzących\" powyżej. Łatwiej jest skopiować powyższych operatorów. "
-"Nieprawidłowe wpisy, włącznie z operatorami bez prawa do połączeń "
-"wychodzących, będą odrzucani bez komunikatów. Wpisy mogą być rozdzielone "
-"spacjami albo podane po jednym w wierszu."
-
-msgid "Full Name"
-msgstr "Pełne imię i nazwisko"
-
-msgid "General Settings"
-msgstr "Ustawienia ogólne"
-
-msgid "Google Accounts"
-msgstr "Konta Google"
-
-msgid "Google Talk Status"
-msgstr "Status Google Talk"
-
-msgid "Google Talk Status Message"
-msgstr "Opis Google Talk"
-
-msgid "Google Voice/Talk Accounts"
-msgstr "Konta Google Voice/Talk"
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr "Połączenia przychodzące"
-
-msgid "Insert QoS Rules"
-msgstr ""
-
-msgid "Makes Outgoing Calls"
-msgstr ""
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr ""
-
-msgid "PBX Main Page"
-msgstr ""
-
-msgid "PBX Service Status"
-msgstr ""
-
-msgid "PIN"
-msgstr ""
-
-msgid "Password"
-msgstr ""
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr ""
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr ""
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr ""
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr ""
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr ""
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr ""
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
diff --git a/applications/luci-app-pbx/po/pt-br/pbx.po b/applications/luci-app-pbx/po/pt-br/pbx.po
deleted file mode 100644
index fd93e4fffb..0000000000
--- a/applications/luci-app-pbx/po/pt-br/pbx.po
+++ /dev/null
@@ -1,744 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2014-08-04 09:00+0200\n"
-"Last-Translator: Luiz Angelo <luizluca@gmail.com>\n"
-"Language-Team: none\n"
-"Language: pt_BR\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n > 1);\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Advanced Settings"
-msgstr "Configurações Avançadas"
-
-msgid "Available"
-msgstr "Disponível"
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-"Evite usar qualquer carácter que não seja um alfanumérico, espaço, vírgula "
-"ou ponto."
-
-msgid "Away"
-msgstr "Ausente"
-
-msgid "Blacklisted Numbers"
-msgstr "Números na Lista Negra"
-
-msgid "Call Routing"
-msgstr "Roteamento de Chamada"
-
-# 20140630: edersg: tradução
-msgid "Call-back Numbers"
-msgstr "Voltar a discar os números"
-
-# 20140630: edersg: tradução
-msgid "Call-back Provider"
-msgstr "Voltar a chamar o provedor"
-
-msgid "Call-through Numbers"
-msgstr "Números de Ligação Direta"
-
-msgid "Copy-paste large lists of numbers here."
-msgstr "Copie e cole aqui listas de números extensas."
-
-# 20140630: edersg: tradução
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-"Designar os números que estão autorizados a chamar por este sistema e quais "
-"privilégios do usuário eles terão."
-
-# 20140630: edersg: tradução
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-"Designar números para os quais o sistema irá desligar e ligar de volta, qual "
-"provedor será utilizado para chamá-los, e quais privilégios do usuário "
-"serão concedidos a eles."
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr "Disca números que não casam em qualquer lugar."
-
-msgid "Do Not Disturb"
-msgstr "Não Perturbe"
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr "Domínio/Endereço IP/Domínio Dinâmico"
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr "Lista Dinâmica dos Números da Lista Negra"
-
-msgid "Email"
-msgstr "Email"
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr "Habilitar Chamadas Recebidas (Registrar pelo SIP)"
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr "Habilitar Chamadas Recebidas (defina o Estado abaixo)"
-
-msgid "Enable Outgoing Calls"
-msgstr "Habilitar Chamadas para Fora"
-
-msgid "Enabled"
-msgstr "Habilitado"
-
-# 20140630: edersg: tradução
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-"Digite um provedor VoIP para utilizar para voltar a chamada no formato "
-"username@some.host.name conforme listado acima em \"Chamadas Originadas\". É "
-"mais fácil copiar e colar os provedores. As entradas inválidas, incluindo "
-"provedores não habilitados para chamadas de saída, serão rejeitados em "
-"silêncio."
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-"Entre com os números de telefone que você deseja rejeitar automaticamente. "
-"Você pode omitir o código do país e qualquer zeros no início, mas, por "
-"favor, teste para ter certeza que você está bloqueando da área desejada com "
-"sucesso."
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-"Entre este endereço IP (ou IP:porta) na configuração de servidor/registrador "
-"dos seus dispositivos SIP que você irá usar SOMENTE localmente e nunca de um "
-"local remoto."
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-"Entre com o nome do equipamento (ou equipamento:porta) na configuração de "
-"servidor/Registrar do seus dispositivos SIP que você irá usar de um local "
-"remoto (eles também funcionarão localmente)."
-
-msgid "External SIP Port"
-msgstr "Porta SIP Externa"
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-"Para cada provedor habilitado para receber chamadas, aqui você pode "
-"restringir quais usuários tocarão quando receber chamadas. Se a lista "
-"estiver vazia, o sistema indicará que todos os usuários com recepção de "
-"chamadas habilitada tocarão. Nome de usuários inválidos serão rejeitados "
-"silenciosamente. Além disto, entrar com um nome de usuário aqui sobrescreve "
-"a configuração do usuário para não receber chamadas. Desta forma, você pode "
-"fazer com que alguns usuários toquem somente para alguns provedores "
-"específicos. As entradas podem ser inseridas usando uma lista separada por "
-"espaço ou um por nova linha."
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-"Para cada usuário habilitado para realizar chamadas externas, você pode "
-"restringir quais provedores o usuário poderá usar. Por padrão, todos os "
-"usuários podem usar todos os provedores. Para aparecer na lista abaixo, o "
-"usuário deve estar habilitado para realizar chamadas externas na página de "
-"\"Contas de Usuários\". Entre com os provedores de VoIP no formato "
-"usuário@algum.nome.de.equipamento, como listado em \"Chamadas Efetuadas\" "
-"abaixo. É mais fácil copiar e colar os provedores da lista abaixo. Entradas "
-"inválidas, includindo provedores não habilitados para chamadas externas, "
-"serão rejeitadas silenciosamente. As entradas podem ser inseridas usando uma "
-"lista separada por espaço ou um por nova linha."
-
-msgid "Full Name"
-msgstr "Nome Completo"
-
-msgid "General Settings"
-msgstr "Configurações Gerais"
-
-msgid "Google Accounts"
-msgstr "Contas do Google"
-
-msgid "Google Talk Status"
-msgstr "Estado do Google Talk"
-
-msgid "Google Talk Status Message"
-msgstr "Mensagem de Estado do Google Talk"
-
-msgid "Google Voice/Talk Accounts"
-msgstr "Contas do Google Voice/Talk"
-
-# 20140630: edersg: tradução
-msgid "Hang-up Delay"
-msgstr "Atraso de hang-up"
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-"Aqui você deve configurar pelo menos uma conta SIP, que você irá usar para "
-"se cadastrar neste serviço. Use essa conta, seja em um adaptador de "
-"telefonia analógica (ATA), ou em um softphone SIP como Linphone, CSipSimple, "
-"ou Sipdroid em seu smartphone, ou o Ekiga, Linphone, ou X-Lite no seu "
-"computador. Por padrão, ao receber uma chamada em uma das suas contas nos "
-"provedores VoIP ou em números GV, todas as contas SIP tocarão "
-"simultaneamente."
-
-# 20140630: edersg: tradução
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-"Quanto tempo esperar antes de desligar. Se o provedor que você utiliza para "
-"discar automaticamente encaminha para a caixa postal de voz, você pode "
-"definir este valor para um atraso que irá permitir que você desligue sua "
-"chamada antes de ser encaminhada e cobrado financeiramente por isso."
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-"Se definir o servidor/registrador como %s ou %s não funcionar para você, "
-"tente defini-lo como %s ou %s e entre com este número de porta em um campo "
-"separado que especifica o número da porta do servidor/registrador. Fique "
-"ciente que alguns dispositivos têm uma configuração confusa que define a "
-"porta de origem das solicitações SIP no dispositivo SIP em si (a porta local "
-"no dispositivo). A porta especificada nesta página não é essa porta de "
-"ligação, mas a porta na qual o serviço escutará serviço."
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-"Se você sentir falhas ou alta latência enquanto baixa conteúdos pesados​​, "
-"você pode querer habilitar o <abbr title=\"Quality of Service, Qualidade de "
-"serviço\">QoS</abbr>. O <abbr title=\"Quality of Service, Qualidade de "
-"serviço\">QoS</abbr> prioriza o tráfego de e para a sua rede para endereços "
-"IP e portas específicas, resultando em melhor latência e redimento de som. "
-"Se ativado, será configurada automaticamente pelo PABX uma regra de <abbr "
-"title=\"Quality of Service, Qualidade de serviço\">QoS</abbr> para este "
-"serviço, mas você deve visitar a página de configuração de <abbr title="
-"\"Quality of Service, Qualidade de serviço\">QoS</abbr> (Rede -> QoS) para "
-"configurar outras configurações críticas de QoS como as velocidades da sua "
-"conexão."
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-"Se você tiver mais de uma conta que pode fazer chamadas externas, você deve "
-"informar uma lista de números de telefone e/ou prefixos nos seguintes campos "
-"para cada provedor listados. Prefixos inválidos são removidos "
-"silênciosamente, e some os caracteres 0-9, X, Z, N, # *,, e + são válidos. A "
-"letra X corresponde a 0-9, Z corresponde a 1-9, e N corresponde a 2-9. Por "
-"exemplo, para fazer chamadas para a Alemanha através de um provedor, você "
-"pode digitar 49. Para fazer chamadas para a América do Norte, você pode "
-"entrar 1NXXNXXXXXX. Se um de seus provedores pode fazer chamadas locais para "
-"um código de área como Nova York (646), você pode entrar com 646NXXXXXX para "
-"esse provedor. Você deve deixar uma conta com uma lista vazia para fazer "
-"chamadas com ele por padrão para o caso do prefixo não casar com nenhum "
-"outro fornecedor. O sistema irá substituir automaticamente uma lista vazia "
-"com uma mensagem que os este provedor será utilizado caso nenhuma das regras "
-"dos demais provedores casem. Seja tão específico quanto possível (isto é "
-"1NXXNXXXXXX é melhor do que 1). Por favor, note que todos os códigos de "
-"discagem internacionais são descartados (por exemplo 00, 011, 010, 0011). As "
-"entradas podem ser feitas em uma lista separada por espaços ou por nova "
-"linha."
-
-msgid "Incoming Calls"
-msgstr "Chamadas Recebidas"
-
-msgid "Insert QoS Rules"
-msgstr "Inserir Regras QoS"
-
-msgid "Makes Outgoing Calls"
-msgstr "Realiza Chamadas para Fora"
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr "NOTA: Não existe uma conta Google ou provedor SIP configurado."
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-"NOTA: Não existe uma conta Google ou provedor SIP habilitado para receber "
-"chamadas."
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-"NOTA: Não existe uma conta Google ou provedor SIP habilitado para efetuar "
-"chamadas externas."
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr "NOTA: Não existe uma conta local configurada."
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-"NOTA: Não existe uma conta local configurada para efetuar chamadas externas."
-
-msgid "No"
-msgstr "Não"
-
-msgid "Number of Seconds to Ring"
-msgstr "Número de Segundos para Tocar"
-
-msgid "Outbound Proxy"
-msgstr "Proxy Externo"
-
-msgid "Outgoing Calls"
-msgstr "Chamadas Efetuadas"
-
-msgid "PBX Main Page"
-msgstr "Página Principal do PBX"
-
-msgid "PBX Service Status"
-msgstr "Estado do Serviço PBX"
-
-msgid "PIN"
-msgstr "PIN"
-
-msgid "Password"
-msgstr "Senha"
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-"Escolha uma porta aleatória entre 6500 e 9500 onde o serviço irá escutar. "
-"Não escolha a porta padrão 5060 pois ela é frequentemente alvo de ataques de "
-"força bruta. Quanto terminar, (1) clique em \"Salvar e Aplicar\", e (2) olhe "
-"na seção \"Dispositivo SIP/Contas do Softphone\" para as configurações "
-"atualizadas do servidor e porta para o seu Dispositivo SIP/Softphone."
-
-msgid "Port Setting for SIP Devices"
-msgstr "Configuração da Porta para Dispositivos SIP"
-
-msgid "Providers Used for Outgoing Calls"
-msgstr "Provedores Usados para as Chamadas para Fora"
-
-msgid "QoS Settings"
-msgstr "Configurações de QoS"
-
-msgid "RTP Port Range End"
-msgstr "Final da Faixa de Portas RTP"
-
-msgid "RTP Port Range Start"
-msgstr "Inicio da Faixa de Portas RTP"
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-"O tráfego RTP transporta de fato os pacotes de voz. Este é o início do "
-"intervalo de portas que será usado para a estabelecer uma comunicação RTP. "
-"Geralmente não é um problema deixar esta configuração com o valor padrão."
-
-msgid "Receives Incoming Calls"
-msgstr "Recebe Chamadas para Dentro"
-
-msgid "Remote Usage"
-msgstr "Uso Remoto"
-
-msgid "Rings users enabled for incoming calls"
-msgstr "Toca usuários habilitados para receber chamadas"
-
-msgid "SIP Accounts"
-msgstr "Contas SIP"
-
-msgid "SIP Device/Softphone Accounts"
-msgstr "Contas de Dispositivos SIP/Telefones em Software"
-
-msgid "SIP Provider Accounts"
-msgstr "Contas dos Provedores SIP"
-
-msgid "SIP Realm (needed by some providers)"
-msgstr "Domínio SIP (necessário para alguns provedores)"
-
-msgid "SIP Server/Registrar"
-msgstr "Servidor SIP/Registrador"
-
-msgid "SIP Server/Registrar Port"
-msgstr "Porta do Servidor SIP/Registrador"
-
-msgid "Server Setting"
-msgstr "Configuração do Servidor"
-
-msgid "Server Setting for Local SIP Devices"
-msgstr "Configuração do Servidor para Dispositivos SIP Locais"
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr "Configuração do Servidor para Dispositivos SIP Remotos"
-
-msgid "Service Status"
-msgstr "Estado do Serviço"
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-"Define o número de segundos para tocar o telefone ao receber chamadas antes "
-"de desligar ou ir para a caixa postal, se o correio de voz estiver instalado "
-"e habilitado."
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr "Números na Lista Negra separados por Espaço"
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-"Especifique os números individualmente aqui. Pressione o Enter para "
-"adicionar mais números."
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-"Especifique aqui os números individualmente. Pressione o \"Enter\" para "
-"adicionar mais números. Você terá que experimentar com qual código de país "
-"ou de área você precisa adicionar aos números."
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-"O número(s) acima especificados serão capazes de discar com os provedores "
-"deste usuário. Nomes inválidos, incluindo usuários não habilitados para "
-"chamadas externas, serão descartados silenciosamente. Por favor, verifique "
-"se a entrada foi aceita."
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-"Esta página de configuração permite configurar um sistema de serviço de "
-"telefone (PABX), que permite fazer chamadas telefônicas através do Google "
-"múltipla e SIP (como Sipgate, SipSorcery e Betamax) contas e compartilhá-los "
-"entre diversos dispositivos SIP. Note-se que as contas do Google, contas "
-"SIP, e contas de usuários locais são configurados em \"Contas do Google\", "
-"\"Contas SIP\" e \"Contas de Usuário\" sub-seções. Você deve adicionar pelo "
-"menos uma conta de usuário para este PABX e configurar um dispositivo SIP ou "
-"softphone para usar a conta, a fim de fazer e receber chamadas com o "
-"Google / SIP contas. Configurando vários usuários permitem que você faça "
-"chamadas gratuitas entre todos os usuários, e partilhar o Google configurado "
-"e contas SIP. Se você tem mais de um Google e contas SIP configurado, você "
-"provavelmente deve configurar como as chamadas de e para eles são "
-"encaminhados para a \"Call Routing\" página. Se você está interessado em "
-"usar o seu próprio PABX de qualquer lugar do mundo, então, visitar o "
-"\"Remote Uso\" na seção \"Advanced Settings\" página."
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-"Este é o nome que o servidor VoIP será usado para identificar-se quando se "
-"registrar para VoIP (SIP) fornecedores. Alguns provedores exigem isso para "
-"uma seqüência específica de correspondência de um dispositivo de hardware "
-"SIP."
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-"Este é o local onde você indica quais contas Google/SIP serão usadas para "
-"chamar quais códigos de área/país, que usuários poderão usar quais contas "
-"Google/SIP, como as chamadas recebidas serão roteadas, que números podem ser "
-"recebidos por este PBX com uma senha e qual números estão banidos."
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-"Este é o local onde você configura suas contas Google (Talk e Voice) para "
-"poder usá-las para realizar ou receber chamadas (conversa por voz e chamadas "
-"para telefones reais). Por favor, realize ao menos uma chamada de voz usando "
-"o plugin do Google Talk, instalável na interface do GMail. Após esta "
-"chamada, saia da sua conta em todos os serviços. Clique em \"Adicionar\" "
-"para adicionar quantas contas você desejar."
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-"Este é o local onde você configura suas contas SIP (VoIP) como Sipgate, "
-"SipSorcery, os populares provedores Betamax, e qualquer outro provedor com "
-"suporte a SIP para permitir o uso destas contas para efetuar e receber "
-"chamadas (URI de SIP e chamads para números reais). Clique em \"Adicionar\" "
-"para adicionar quantas contas você desejar."
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-"Esta opção deve estar definida como \"Sim\" se você tem um DDR (Discagem "
-"Direta a Ramal) associado com esta conta SIP or quer receber chamadas URI de "
-"SIP através deste provedor."
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-"Esta seção contém configurações que não precisam ser modificadas em "
-"condições normais. Aqui você pode configurar seu sistema para usar com "
-"dispositivos SIP remotos e resolver problemas com a qualidade das chamadas "
-"através da inserção de regras de <abbr title=\"Quality of Service, Qualidade "
-"de serviço\">QoS</abbr>."
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-"Use o nome de usuário numérico (4 a 5 dígitos) se você estiver conectando "
-"telefones normais com ATAs para este sistema (para que eles possam discar os "
-"nomes de seus usuários)."
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-"Use esta conta para realizar chamadas externas como configurado na seção de "
-"\"Roteamento de Chamada\"."
-
-msgid "Use this account to make outgoing calls."
-msgstr "Use esta conta para realizar chamadas externas."
-
-msgid "User Accounts"
-msgstr "Contas de Usuários"
-
-msgid "User Agent String"
-msgstr "Texto para o Agente do Usuário"
-
-msgid "User Name"
-msgstr "Nome do Usuário"
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr "Usa provedores habilitados para chamadas externas"
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-"Quando alguém iniciar uma conversa por voz com sua conta do GTalk ou chamar "
-"seu número GVoice (se você tiver uma conta Google Voice), a chamada será "
-"encaminhada para qualquer usuários que estão conectados (registados "
-"utilizando um dispositivo SIP ou softphone) e autorizados a receber a "
-"chamada. Se você tiver uma conta Google Voice, você deve ir para as "
-"configurações da sua conta GVoice e encaminhar as chamadas para o Google "
-"Chat, a fim de realmente receber chamadas feitas para o seu número GVoice. "
-"Se você tiver problemas para receber chamadas oriundas do GVoice, "
-"experimente a opção \"Call Screening/Monitoramento de Chamadas\" na "
-"configurações da sua conta GVoice. Finalmente, certifique-se de nenhum outro "
-"cliente está online com essa conta (navegador contado no GMail, aplicativo "
-"Google Talk no Desktop ou Celular), pois isto pode interferir."
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-"Quando a sua senha for salva, ela desaparece deste campo e não será exibida "
-"para sua proteção. A senha será alterada somente quando você informar uma "
-"nova senha diferente da que foi salva anteriormente."
-
-msgid "Yes"
-msgstr "Sim"
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-"Você pode informar aqui o nome do domínio, endereço IP externo, ou um nome "
-"de domínio dinâmico. O melhor é informar um endereço IP estático. Se o seu "
-"endereço IP é dinâmico e ele muda, sua configuração se tornará inválida. "
-"Desta forma, é recomendado configurar um serviço de domínios dinâmicos e "
-"utilizar este nome aqui. Você pode configurar o serviço de domínios "
-"dinâmicos com o pacote luci-app-ddns."
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-"Você pode especificar um nome real para aparecer no identificador de "
-"chamadas."
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-"Você pode usar seus dispositivos SIP/softphones com este sistema a partir de "
-"um local remoto, desde que o seu provedor de Internet lhe forneça um "
-"endereço IP público. Você poderá ligar para outros usuários locais sem custo "
-"(por exemplo, outros adaptadores de telefone analógico (ATAs)) e usar seus "
-"provedores de VoIP para fazer chamadas como se fossem originadas do local do "
-"seu PBX. Depois de configurar esta aba, volte para onde os usuários são "
-"configurados e veja as novas configurações de servidor e porta com as quais "
-"você precisa configurar os seus dispositivos SIP remotos. Por favor, note "
-"que se este PABX não está rodando no seu roteador, você terá que configurar "
-"o redirecionamento de portas (NAT) no seu roteador. Por favor, encaminhe as "
-"portas abaixo (porta SIP e intervalo de porta RTP) para o endereço IP do "
-"dispositivo que executa este PBX."
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-"Seu PIN desaparece deste campo quando for salvo e não será exibido para sua "
-"proteção. Ele será alterada somente quando você informar um PIN diferente do "
-"que foi salvo anteriormente. É possível deixá-lo em branco mas fique atento "
-"quanto as implicações na segurança."
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-"Sua senha desaparece deste campo quando for salva e não será exibida para "
-"sua proteção. A senha será alterada somente quando você informar uma nova "
-"senha diferente da que foi salva anteriormente."
-
-#~ msgid ""
-#~ "Designate numbers that are allowed to call through this system and which "
-#~ "user's privileges it will have."
-#~ msgstr ""
-#~ "Números definidos que poderão realizar chamadas através deste sistema e "
-#~ "quais privilégios o usuário terá."
-
-#~ msgid ""
-#~ "Pick a random port number between 6500 and 9500 for the service to listen "
-#~ "on. Do not pick the standard 5060, because it is often subject to brute-"
-#~ "force attacks. When finished, (1) click \"Save and Apply\", and (2) click "
-#~ "the \"Restart VoIP Service\" button above. Finally, (3) look in the \"SIP "
-#~ "Device/Softphone Accounts\" section for updated Server and Port settings "
-#~ "for your SIP Devices/Softphones."
-#~ msgstr ""
-#~ "Escolha um número de porta aleatória entre 6500 e 9500 para o serviço "
-#~ "escutar. Não escolher o padrão 5060, porque é frequentemente alvo de ataques "
-#~ "de força bruta. Quando terminar, (1) clique em \"Salvar e Aplicar\", e (2) "
-#~ "clique no \"Reiniciar o serviço VoIP\" acima. Finalmente, (3) olhe na seção "
-#~ "\"Contas de Dispositivos SIP/Telefones em Software\" para atualizar o endereço "
-#~ "e porta do servidor para seu Dispositivos SIP/Telefones em Software."
-
-#~ msgid ""
-#~ "You can enter your domain name, external IP address, or dynamic domain "
-#~ "name here Please keep in mind that if your IP address is dynamic and it "
-#~ "changes your configuration will become invalid. Hence, it's recommended "
-#~ "to set up Dynamic DNS in this case."
-#~ msgstr ""
-#~ "Você pode digitar aqui o seu nome de domínio, endereço IP externo, ou nome "
-#~ "de domínio dinâmico. Tenha em mente que se o seu endereço IP é dinâmico e "
-#~ "ele mudar, a sua configuração se tornará inválida. Por isso, é recomendado "
-#~ "configurar um DNS dinâmico neste caso."
-
-#~ msgid "Account Status"
-#~ msgstr "Estado da Conta"
-
-#~ msgid "Account Status Message"
-#~ msgstr "Mensagem do Estado da Conta"
-
-#~ msgid "Domain Name/Dynamic Domain Name"
-#~ msgstr "Nome do Domínio/Nome do Domínio Dinâmico"
-
-#~ msgid "Enable Incoming Calls (See Status, Message below)"
-#~ msgstr "Habilitar Chamadas Recebidas (Veja o Estado, Mensagem abaixo)"
-
-#~ msgid "Service Control and Connection Status"
-#~ msgstr "Controle do Serviço e Estado da Conexão"
diff --git a/applications/luci-app-pbx/po/pt/pbx.po b/applications/luci-app-pbx/po/pt/pbx.po
deleted file mode 100644
index 75b6c8cd1a..0000000000
--- a/applications/luci-app-pbx/po/pt/pbx.po
+++ /dev/null
@@ -1,487 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2013-09-22 19:17+0200\n"
-"Last-Translator: Low <pedroloureiro1@sapo.pt>\n"
-"Language-Team: none\n"
-"Language: pt\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Advanced Settings"
-msgstr ""
-
-msgid "Available"
-msgstr "Disponível"
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-
-msgid "Away"
-msgstr ""
-
-msgid "Blacklisted Numbers"
-msgstr ""
-
-msgid "Call Routing"
-msgstr ""
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr ""
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr ""
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Email"
-msgstr ""
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr ""
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr ""
-
-msgid "Enable Outgoing Calls"
-msgstr ""
-
-msgid "Enabled"
-msgstr "Ativado"
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr ""
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr "Nome Completo"
-
-msgid "General Settings"
-msgstr ""
-
-msgid "Google Accounts"
-msgstr ""
-
-msgid "Google Talk Status"
-msgstr ""
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr ""
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr ""
-
-msgid "Insert QoS Rules"
-msgstr ""
-
-msgid "Makes Outgoing Calls"
-msgstr ""
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr "Não"
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr ""
-
-msgid "PBX Main Page"
-msgstr ""
-
-msgid "PBX Service Status"
-msgstr ""
-
-msgid "PIN"
-msgstr ""
-
-msgid "Password"
-msgstr ""
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr ""
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr ""
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr ""
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr ""
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr ""
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr ""
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr "Sim"
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
diff --git a/applications/luci-app-pbx/po/ro/pbx.po b/applications/luci-app-pbx/po/ro/pbx.po
deleted file mode 100644
index 49e8daccf4..0000000000
--- a/applications/luci-app-pbx/po/ro/pbx.po
+++ /dev/null
@@ -1,488 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2014-06-28 18:50+0200\n"
-"Last-Translator: xxvirusxx <condor20_05@yahoo.it>\n"
-"Language-Team: none\n"
-"Language: ro\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=3; plural=(n==1 ? 0 : (n==0 || (n%100 > 0 && n%100 < "
-"20)) ? 1 : 2);;\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Advanced Settings"
-msgstr "Setări avansate"
-
-msgid "Available"
-msgstr "Disponibil"
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-
-msgid "Away"
-msgstr ""
-
-msgid "Blacklisted Numbers"
-msgstr ""
-
-msgid "Call Routing"
-msgstr ""
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr ""
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr "Nu deranjaţi"
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr "Domeniu/Adresă IP/Domeniu dinamic"
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Email"
-msgstr ""
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr ""
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr ""
-
-msgid "Enable Outgoing Calls"
-msgstr ""
-
-msgid "Enabled"
-msgstr "Activat"
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr ""
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr "Nume complet"
-
-msgid "General Settings"
-msgstr "Setări generale"
-
-msgid "Google Accounts"
-msgstr "Conturi Google"
-
-msgid "Google Talk Status"
-msgstr ""
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr ""
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr ""
-
-msgid "Insert QoS Rules"
-msgstr ""
-
-msgid "Makes Outgoing Calls"
-msgstr ""
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr ""
-
-msgid "PBX Main Page"
-msgstr ""
-
-msgid "PBX Service Status"
-msgstr ""
-
-msgid "PIN"
-msgstr ""
-
-msgid "Password"
-msgstr "Parolă"
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr "Setări QoS"
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr ""
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr ""
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr ""
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr ""
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr ""
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
diff --git a/applications/luci-app-pbx/po/ru/pbx.po b/applications/luci-app-pbx/po/ru/pbx.po
deleted file mode 100644
index e85c947e1a..0000000000
--- a/applications/luci-app-pbx/po/ru/pbx.po
+++ /dev/null
@@ -1,525 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2013-09-06 10:28+0200\n"
-"Last-Translator: datasheet <michael.gritsaenko@gmail.com>\n"
-"Language-Team: none\n"
-"Language: ru\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=3; plural=(n%10==1 && n%100!=11 ? 0 : n%10>=2 && n%"
-"10<=4 && (n%100<10 || n%100>=20) ? 1 : 2);\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Advanced Settings"
-msgstr "Расширенные установки"
-
-msgid "Available"
-msgstr "Доступен"
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-"Старайтесь не использовать ничего, кроме алфавитно-цифровых символов, "
-"пробелов, запятых и точек."
-
-msgid "Away"
-msgstr "Отошел"
-
-msgid "Blacklisted Numbers"
-msgstr "Номера в \"черном\" списке"
-
-msgid "Call Routing"
-msgstr "Маршрутизация вызовов"
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr "Номера сквозных вызовов"
-
-msgid "Copy-paste large lists of numbers here."
-msgstr "Вставьте большие списки номеров здесь"
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr "Не беспокоить"
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr "Динамический список запрещенных номеров"
-
-msgid "Email"
-msgstr "Эл. почта"
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr "Разрешить входящие вызовы (регистрация через SIP)"
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr "Разрешить входящие звонки (см. ниже Статус)"
-
-msgid "Enable Outgoing Calls"
-msgstr "Разрешить исходящие вызовы"
-
-msgid "Enabled"
-msgstr "Включено"
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-"Введите телефонные номера, звонки с которых вы хотите автоматически "
-"отклонять. Вы, вероятно, не должны вводить код страны и ведущие нули, но, "
-"чтобы удостовериться в этом, пожалуйста проверьте, что звонки из "
-"нежелательной зоны успешно блокируются."
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-"Введите этот IP (или IP порт) в установках Сервера/Регистратора SIP "
-"устройств, который вы будете использовать ТОЛЬКО локально и никогда удаленно."
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-"Введите это имя_хоста (или имя_хоста:порт) в установках Сервера/Регистратора "
-"тех SIP-устройств, которые вы будете использовать удаленно (локально они "
-"также будут работать)."
-
-msgid "External SIP Port"
-msgstr "Внешний порт SIP"
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr "Полное имя"
-
-msgid "General Settings"
-msgstr "Общие установки"
-
-msgid "Google Accounts"
-msgstr "Учетные записи Google"
-
-msgid "Google Talk Status"
-msgstr "Статус Google Talk"
-
-msgid "Google Talk Status Message"
-msgstr "Сообщение статуса Google Talk"
-
-msgid "Google Voice/Talk Accounts"
-msgstr "Учетные записи Google Voice/Talk"
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr "Входящие вызовы"
-
-msgid "Insert QoS Rules"
-msgstr "Вставить правила QoS"
-
-msgid "Makes Outgoing Calls"
-msgstr "Совершает исходящие вызовы"
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr "Нет"
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr "Outbound прокси сервер"
-
-msgid "Outgoing Calls"
-msgstr "Исходящие вызовы"
-
-msgid "PBX Main Page"
-msgstr "Главная страница АТС"
-
-#, fuzzy
-msgid "PBX Service Status"
-msgstr "Состояние службы АТС"
-
-msgid "PIN"
-msgstr "PIN"
-
-msgid "Password"
-msgstr "Пароль"
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr "Настройки порта устройств SIP"
-
-msgid "Providers Used for Outgoing Calls"
-msgstr "Провайдеры исходящих вызовов"
-
-msgid "QoS Settings"
-msgstr "Установки QoS"
-
-msgid "RTP Port Range End"
-msgstr "Конец диапазона портов RTP"
-
-msgid "RTP Port Range Start"
-msgstr "Начало диапазоно портов RTP"
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr "Принимает входящие вызовы"
-
-msgid "Remote Usage"
-msgstr "Удаленное использование"
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr "Учетные записи SIP"
-
-msgid "SIP Device/Softphone Accounts"
-msgstr "Учетные записи SIP устройства/программного телефона"
-
-msgid "SIP Provider Accounts"
-msgstr "Учетные записи SIP провайдера"
-
-msgid "SIP Realm (needed by some providers)"
-msgstr "SIP Realm (нужен для некоторых провайдеров)"
-
-msgid "SIP Server/Registrar"
-msgstr "SIP Сервер/Регистратор"
-
-msgid "SIP Server/Registrar Port"
-msgstr "Порт SIP Сервера/Регистратора"
-
-msgid "Server Setting"
-msgstr "Настройки сервера"
-
-msgid "Server Setting for Local SIP Devices"
-msgstr "Установки сервера для локальных SIP устройств"
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr "Настройки сервера для удаленных SIP устройств"
-
-msgid "Service Status"
-msgstr "Состояние сервиса"
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr "Черный список номеров (пробел между номерами для разделения)"
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-"Укажите отдельные номера. Нажмите enter, чтобы добавить больше номеров."
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-"Использовать эту учетную запись для исходящих вызовов в соответстии с "
-"наcтройками секции \"Маршрутизация вызовов\"."
-
-msgid "Use this account to make outgoing calls."
-msgstr "Использовать эту учетную запись для исходящих вызовов"
-
-msgid "User Accounts"
-msgstr "Учетные записи пользователя"
-
-msgid "User Agent String"
-msgstr "Строка агента пользователя"
-
-msgid "User Name"
-msgstr "Имя пользователя"
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr "Да"
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr "Здесь Вы можете указать имя для отображения вместо ID звонящего."
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-#~ msgid ""
-#~ "Designate numbers that are allowed to call through this system and which "
-#~ "user's privileges it will have."
-#~ msgstr ""
-#~ "Указать телефонные номера, которым разрешено осуществлять звонки через эту "
-#~ "систему, а также какими они будут обладать пользовательскими привилегиями."
-
-#~ msgid "Account Status"
-#~ msgstr "Статус учетной записи"
-
-#~ msgid "Account Status Message"
-#~ msgstr "Статус сообщение учетной записи"
-
-#~ msgid "Domain Name/Dynamic Domain Name"
-#~ msgstr "Имя домена/Динамическое имя домена"
-
-#~ msgid "Enable Incoming Calls (See Status, Message below)"
-#~ msgstr "Разрешить входящие вызовы (см. статус, сообщение ниже)"
-
-#~ msgid "Service Control and Connection Status"
-#~ msgstr "Управление сервисом и статус соединения"
diff --git a/applications/luci-app-pbx/po/sk/pbx.po b/applications/luci-app-pbx/po/sk/pbx.po
deleted file mode 100644
index 7b6d4a5c64..0000000000
--- a/applications/luci-app-pbx/po/sk/pbx.po
+++ /dev/null
@@ -1,484 +0,0 @@
-msgid ""
-msgstr ""
-"Content-Type: text/plain; charset=UTF-8\n"
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=3; plural=(n==1) ? 0 : (n>=2 && n<=4) ? 1 : 2;\n"
-
-msgid "Advanced Settings"
-msgstr ""
-
-msgid "Available"
-msgstr ""
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-
-msgid "Away"
-msgstr ""
-
-msgid "Blacklisted Numbers"
-msgstr ""
-
-msgid "Call Routing"
-msgstr ""
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr ""
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr ""
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Email"
-msgstr ""
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr ""
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr ""
-
-msgid "Enable Outgoing Calls"
-msgstr ""
-
-msgid "Enabled"
-msgstr ""
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr ""
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr ""
-
-msgid "General Settings"
-msgstr ""
-
-msgid "Google Accounts"
-msgstr ""
-
-msgid "Google Talk Status"
-msgstr ""
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr ""
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr ""
-
-msgid "Insert QoS Rules"
-msgstr ""
-
-msgid "Makes Outgoing Calls"
-msgstr ""
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr ""
-
-msgid "PBX Main Page"
-msgstr ""
-
-msgid "PBX Service Status"
-msgstr ""
-
-msgid "PIN"
-msgstr ""
-
-msgid "Password"
-msgstr ""
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr ""
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr ""
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr ""
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr ""
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr ""
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr ""
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
diff --git a/applications/luci-app-pbx/po/sv/pbx.po b/applications/luci-app-pbx/po/sv/pbx.po
deleted file mode 100644
index 400289b6bb..0000000000
--- a/applications/luci-app-pbx/po/sv/pbx.po
+++ /dev/null
@@ -1,506 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2014-04-28 06:11+0200\n"
-"Last-Translator: Umeaboy <kristoffer.grundstrom1983@gmail.com>\n"
-"Language-Team: none\n"
-"Language: sv\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=2; plural=(n != 1);\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Advanced Settings"
-msgstr "Avancerade inställningar"
-
-msgid "Available"
-msgstr "Tillgänglig"
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-"Undvik att använda allt förutom alfa-numeriska karaktärer, mellanslag, komma-"
-"tecken och punkt."
-
-msgid "Away"
-msgstr "Borta"
-
-msgid "Blacklisted Numbers"
-msgstr "Svartlistade nummer"
-
-msgid "Call Routing"
-msgstr ""
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr "Kopiera och klistra in ett stort antal nummer här."
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr "Ringer upp nummer som inte passar någon annanstans"
-
-msgid "Do Not Disturb"
-msgstr "Stör ej"
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr "Domän/IP-adress/Dynamisk domän"
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr "Dynamisk lista över svartlistade nummer"
-
-msgid "Email"
-msgstr "E-post"
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr "Aktivera inkommande samtal (Registrera via SIP)"
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr "Aktivera inkommande samtal (se status nedanför)"
-
-msgid "Enable Outgoing Calls"
-msgstr "Aktivera utgående samtal"
-
-msgid "Enabled"
-msgstr "Aktiverat"
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-"Ange telefonnummer som du vill neka samtal från automatiskt. Du borde "
-"förmodligen utesluta landskoden och eventuella inledande nollor, men "
-"experimentera gärna för att vara säker på att du lyckas blockera nummer från "
-"ditt önskade område."
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-"Ange den här IP:n (eller IP:port) i Server/Registrar-inställningarna för SIP-"
-"enheter som du endast kommer att använda LOKALT och aldrig från en "
-"fjärrstyrd anslutning."
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-"Ange det här värdnamnet (eller värdnamn:port) under Server/Registrar "
-"inställningen för SIP-enheten som du kommer att använda från en fjärrstyrd "
-"plats (de kommer att fungera lokalt också)."
-
-msgid "External SIP Port"
-msgstr "Extern SIP-port"
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr "Fullständigt namn"
-
-msgid "General Settings"
-msgstr "Allmänna inställningar"
-
-msgid "Google Accounts"
-msgstr "Google-konton"
-
-msgid "Google Talk Status"
-msgstr "Status för Google Talk"
-
-msgid "Google Talk Status Message"
-msgstr "Statusmeddelande för Google Talk"
-
-msgid "Google Voice/Talk Accounts"
-msgstr "Google Voice/Talk-konton"
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr "Inkommande samtal"
-
-msgid "Insert QoS Rules"
-msgstr "För in QoS-regler"
-
-msgid "Makes Outgoing Calls"
-msgstr "Gör utgående samtal"
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr "NOTERA: Det finns inga lokala användarkonton konfigurerade."
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-"NOTERA: Det finns inga lokala användar-konton aktiverade för utgående samtal."
-
-msgid "No"
-msgstr "Nej"
-
-msgid "Number of Seconds to Ring"
-msgstr "Antal sekunder att ringa"
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr "Utgående samtal"
-
-msgid "PBX Main Page"
-msgstr "Huvudsida för PBX"
-
-msgid "PBX Service Status"
-msgstr "Status för PBX-tjänsten"
-
-msgid "PIN"
-msgstr "PIN-kod"
-
-msgid "Password"
-msgstr "Lösenord"
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr "Port-inställning för SIP-enheter"
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr "QoS-inställningar"
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr "Tar emot inkommande samtal"
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr "Ringer användare som är aktiverade för inkommande samtal"
-
-msgid "SIP Accounts"
-msgstr "SIP-konton"
-
-msgid "SIP Device/Softphone Accounts"
-msgstr "SIP-enhet/Softphone-konton"
-
-msgid "SIP Provider Accounts"
-msgstr "SIP-operatörskonton"
-
-msgid "SIP Realm (needed by some providers)"
-msgstr "SIP-sfär (behövs av vissa operatörer)"
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr "Server-inställning"
-
-msgid "Server Setting for Local SIP Devices"
-msgstr "Server-inställning för lokala SIP-enheter"
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr "Server-inställning för fjärrstyrda SIP-enheter"
-
-msgid "Service Status"
-msgstr "Status för tjänst"
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-"Specificera nummer individuellt här. Tryck på enter-knappen för att lägga "
-"till fler nummer."
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-"Det här valet borde vara inställt på \"Ja\" om du har ett DID (riktigt "
-"telefonnummer) associerat med det här SIP-kontot eller om du vill ta emot "
-"SIP uri-samtal via den här operatören."
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr "Använd det här kontot för att göra utgående samtal."
-
-msgid "User Accounts"
-msgstr "Användar-konton"
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr "Användarnamn"
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr "Använder operatörer för utgående samtal"
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr "Ja"
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-"Du kan specifiera ett riktigt namn som visas i samband med nummret här."
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
diff --git a/applications/luci-app-pbx/po/templates/pbx.pot b/applications/luci-app-pbx/po/templates/pbx.pot
deleted file mode 100644
index 86dd2eb72d..0000000000
--- a/applications/luci-app-pbx/po/templates/pbx.pot
+++ /dev/null
@@ -1,477 +0,0 @@
-msgid ""
-msgstr "Content-Type: text/plain; charset=UTF-8"
-
-msgid "Advanced Settings"
-msgstr ""
-
-msgid "Available"
-msgstr ""
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-
-msgid "Away"
-msgstr ""
-
-msgid "Blacklisted Numbers"
-msgstr ""
-
-msgid "Call Routing"
-msgstr ""
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr ""
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr ""
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Email"
-msgstr ""
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr ""
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr ""
-
-msgid "Enable Outgoing Calls"
-msgstr ""
-
-msgid "Enabled"
-msgstr ""
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr ""
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr ""
-
-msgid "General Settings"
-msgstr ""
-
-msgid "Google Accounts"
-msgstr ""
-
-msgid "Google Talk Status"
-msgstr ""
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr ""
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr ""
-
-msgid "Insert QoS Rules"
-msgstr ""
-
-msgid "Makes Outgoing Calls"
-msgstr ""
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr ""
-
-msgid "PBX Main Page"
-msgstr ""
-
-msgid "PBX Service Status"
-msgstr ""
-
-msgid "PIN"
-msgstr ""
-
-msgid "Password"
-msgstr ""
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr ""
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr ""
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr ""
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr ""
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr ""
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr ""
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
diff --git a/applications/luci-app-pbx/po/tr/pbx.po b/applications/luci-app-pbx/po/tr/pbx.po
deleted file mode 100644
index 59af3e878d..0000000000
--- a/applications/luci-app-pbx/po/tr/pbx.po
+++ /dev/null
@@ -1,484 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=1; plural=0;\n"
-
-msgid "Advanced Settings"
-msgstr ""
-
-msgid "Available"
-msgstr ""
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-
-msgid "Away"
-msgstr ""
-
-msgid "Blacklisted Numbers"
-msgstr ""
-
-msgid "Call Routing"
-msgstr ""
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr ""
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr ""
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Email"
-msgstr ""
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr ""
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr ""
-
-msgid "Enable Outgoing Calls"
-msgstr ""
-
-msgid "Enabled"
-msgstr ""
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr ""
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr ""
-
-msgid "General Settings"
-msgstr ""
-
-msgid "Google Accounts"
-msgstr ""
-
-msgid "Google Talk Status"
-msgstr ""
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr ""
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr ""
-
-msgid "Insert QoS Rules"
-msgstr ""
-
-msgid "Makes Outgoing Calls"
-msgstr ""
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr ""
-
-msgid "PBX Main Page"
-msgstr ""
-
-msgid "PBX Service Status"
-msgstr ""
-
-msgid "PIN"
-msgstr ""
-
-msgid "Password"
-msgstr ""
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr ""
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr ""
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr ""
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr ""
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr ""
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr ""
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
diff --git a/applications/luci-app-pbx/po/uk/pbx.po b/applications/luci-app-pbx/po/uk/pbx.po
deleted file mode 100644
index d65a784435..0000000000
--- a/applications/luci-app-pbx/po/uk/pbx.po
+++ /dev/null
@@ -1,501 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2013-08-13 15:47+0200\n"
-"Last-Translator: zubr_139 <zubr139@ukr.net>\n"
-"Language-Team: none\n"
-"Language: uk\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=3; plural=(n%10==1 && n%100!=11 ? 0 : n%10>=2 && n%"
-"10<=4 && (n%100<10 || n%100>=20) ? 1 : 2);\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Advanced Settings"
-msgstr "Розширені налаштування"
-
-msgid "Available"
-msgstr "Доступний"
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-"Намагайтеся не використовувати нічого, крім алфавітно-цифрових символів, "
-"пропусків, ком і крапок."
-
-msgid "Away"
-msgstr ""
-
-msgid "Blacklisted Numbers"
-msgstr ""
-
-msgid "Call Routing"
-msgstr "Маршрутизація Викликів"
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr "Виклик через номери"
-
-msgid "Copy-paste large lists of numbers here."
-msgstr ""
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-#, fuzzy
-msgid "Do Not Disturb"
-msgstr "Не турбувати"
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-#, fuzzy
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr "Динамічний список небажаних дзвінків"
-
-msgid "Email"
-msgstr "Електронна скринька"
-
-#, fuzzy
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr "Активувати вхідні дзвінки (зареєструватися через SIP)"
-
-#, fuzzy
-msgid "Enable Incoming Calls (set Status below)"
-msgstr "Активувати вхідні дзвінки (Встановити низький статус)"
-
-msgid "Enable Outgoing Calls"
-msgstr "Активувати вихідні виклики"
-
-msgid "Enabled"
-msgstr "Активувати"
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-#, fuzzy
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-"Введіть цей IP (або IP:порт) Сервера/Реєстратор налаштування SIP пристрою ви "
-"будете використовувати тільки локально й ніколи з віддаленого місця."
-
-#, fuzzy
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-"Введіть це хост ім'я (або ім'я хоста:порт) сервер/Реєстратор налаштування "
-"SIP пристрою ви будете використовувати з віддаленого місця розташування "
-"(воно також буде працювати локально)."
-
-msgid "External SIP Port"
-msgstr "Зовнішній порт SIP"
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr "Повне Ім'я"
-
-msgid "General Settings"
-msgstr "Загальні Налаштування"
-
-msgid "Google Accounts"
-msgstr "Облікові записи Google"
-
-msgid "Google Talk Status"
-msgstr ""
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr ""
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr ""
-
-msgid "Insert QoS Rules"
-msgstr ""
-
-msgid "Makes Outgoing Calls"
-msgstr ""
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr "Ні"
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr ""
-
-msgid "PBX Main Page"
-msgstr ""
-
-msgid "PBX Service Status"
-msgstr ""
-
-msgid "PIN"
-msgstr ""
-
-msgid "Password"
-msgstr ""
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr ""
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr ""
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr ""
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr ""
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr "Облікові записи користувачів"
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr "Ім'я користувача"
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr "Так"
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
diff --git a/applications/luci-app-pbx/po/vi/pbx.po b/applications/luci-app-pbx/po/vi/pbx.po
deleted file mode 100644
index 59af3e878d..0000000000
--- a/applications/luci-app-pbx/po/vi/pbx.po
+++ /dev/null
@@ -1,484 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"Last-Translator: Automatically generated\n"
-"Language-Team: none\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=1; plural=0;\n"
-
-msgid "Advanced Settings"
-msgstr ""
-
-msgid "Available"
-msgstr ""
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr ""
-
-msgid "Away"
-msgstr ""
-
-msgid "Blacklisted Numbers"
-msgstr ""
-
-msgid "Call Routing"
-msgstr ""
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr ""
-
-msgid "Copy-paste large lists of numbers here."
-msgstr ""
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr ""
-
-msgid "Do Not Disturb"
-msgstr ""
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr ""
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Email"
-msgstr ""
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr ""
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr ""
-
-msgid "Enable Outgoing Calls"
-msgstr ""
-
-msgid "Enabled"
-msgstr ""
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr ""
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr ""
-
-msgid "General Settings"
-msgstr ""
-
-msgid "Google Accounts"
-msgstr ""
-
-msgid "Google Talk Status"
-msgstr ""
-
-msgid "Google Talk Status Message"
-msgstr ""
-
-msgid "Google Voice/Talk Accounts"
-msgstr ""
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr ""
-
-msgid "Insert QoS Rules"
-msgstr ""
-
-msgid "Makes Outgoing Calls"
-msgstr ""
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr ""
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr ""
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr ""
-
-msgid "No"
-msgstr ""
-
-msgid "Number of Seconds to Ring"
-msgstr ""
-
-msgid "Outbound Proxy"
-msgstr ""
-
-msgid "Outgoing Calls"
-msgstr ""
-
-msgid "PBX Main Page"
-msgstr ""
-
-msgid "PBX Service Status"
-msgstr ""
-
-msgid "PIN"
-msgstr ""
-
-msgid "Password"
-msgstr ""
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr ""
-
-msgid "Providers Used for Outgoing Calls"
-msgstr ""
-
-msgid "QoS Settings"
-msgstr ""
-
-msgid "RTP Port Range End"
-msgstr ""
-
-msgid "RTP Port Range Start"
-msgstr ""
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr ""
-
-msgid "Remote Usage"
-msgstr ""
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr ""
-
-msgid "SIP Device/Softphone Accounts"
-msgstr ""
-
-msgid "SIP Provider Accounts"
-msgstr ""
-
-msgid "SIP Realm (needed by some providers)"
-msgstr ""
-
-msgid "SIP Server/Registrar"
-msgstr ""
-
-msgid "SIP Server/Registrar Port"
-msgstr ""
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr ""
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr ""
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr ""
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
diff --git a/applications/luci-app-pbx/po/zh-cn/pbx.po b/applications/luci-app-pbx/po/zh-cn/pbx.po
deleted file mode 100644
index 45325b99c1..0000000000
--- a/applications/luci-app-pbx/po/zh-cn/pbx.po
+++ /dev/null
@@ -1,495 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2014-07-15 16:11+0200\n"
-"Last-Translator: Tanyingyu <Tanyingyu@163.com>\n"
-"Language-Team: none\n"
-"Language: zh_CN\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=1; plural=0;\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Advanced Settings"
-msgstr "高级设置"
-
-msgid "Available"
-msgstr "可用"
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr "避免使用除字母，数字，空格，逗号和句号外的其他字符。"
-
-msgid "Away"
-msgstr "外"
-
-msgid "Blacklisted Numbers"
-msgstr "黑名单"
-
-msgid "Call Routing"
-msgstr "呼叫路由"
-
-msgid "Call-back Numbers"
-msgstr "回调数"
-
-msgid "Call-back Provider"
-msgstr "回呼提供者"
-
-msgid "Call-through Numbers"
-msgstr "通过数字呼叫"
-
-msgid "Copy-paste large lists of numbers here."
-msgstr "复制粘贴数字大名单。"
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr "其他地方无法匹配拨号号码"
-
-msgid "Do Not Disturb"
-msgstr "请勿打扰"
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr "域名/ IP地址/动态域名"
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr "动态黑名单号码列表"
-
-msgid "Email"
-msgstr "电子邮件"
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr "允许电话呼入（SIP注册者）"
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr "允许电话呼入（下面设置状态）"
-
-msgid "Enable Outgoing Calls"
-msgstr "允许电话外呼"
-
-msgid "Enabled"
-msgstr "允许"
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-"输入您想自动屏蔽的电话号码。您应该忽略国家代码和任何前导零,但请测试来确保您成"
-"功屏蔽了想要屏蔽的号码。"
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-"在SIP设备注册服务器中输入IP(或IP:端口)，仅在本地使用，不可以在远程使用。"
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-
-msgid "External SIP Port"
-msgstr "外部SIP端口"
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr "全名"
-
-msgid "General Settings"
-msgstr "通用设置"
-
-msgid "Google Accounts"
-msgstr "google账号"
-
-msgid "Google Talk Status"
-msgstr "google Talk状态"
-
-msgid "Google Talk Status Message"
-msgstr "google Talk状态消息"
-
-msgid "Google Voice/Talk Accounts"
-msgstr "Google Voice/Talk账号"
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr "呼入电话"
-
-msgid "Insert QoS Rules"
-msgstr "插入QoS规则"
-
-msgid "Makes Outgoing Calls"
-msgstr "安排外呼列表"
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr "注意:没有google或SIP提供者账户配置。"
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr "注意:没有google或SIP提供者账户允许呼入电话。"
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr "注意:没有google或SIP提供者账户允许外呼电话。"
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr "注意:没有本地用户设置。"
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr "注意:没有本地用户允许外呼电话。"
-
-msgid "No"
-msgstr "不"
-
-msgid "Number of Seconds to Ring"
-msgstr "多少秒振铃"
-
-msgid "Outbound Proxy"
-msgstr "外呼代理"
-
-msgid "Outgoing Calls"
-msgstr "外呼电话"
-
-msgid "PBX Main Page"
-msgstr "PBX主页"
-
-msgid "PBX Service Status"
-msgstr "PBX服务状态"
-
-msgid "PIN"
-msgstr "PIN"
-
-msgid "Password"
-msgstr "密码"
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr "SIP设备端口设置"
-
-msgid "Providers Used for Outgoing Calls"
-msgstr "用于外呼电话的提供者"
-
-msgid "QoS Settings"
-msgstr "QoS设置"
-
-msgid "RTP Port Range End"
-msgstr "RTP结束端口"
-
-msgid "RTP Port Range Start"
-msgstr "RTP起始端口"
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr "收到呼入电话"
-
-msgid "Remote Usage"
-msgstr "远程使用"
-
-msgid "Rings users enabled for incoming calls"
-msgstr ""
-
-msgid "SIP Accounts"
-msgstr "SIP账号"
-
-msgid "SIP Device/Softphone Accounts"
-msgstr "SIP 设备/软电话账号"
-
-msgid "SIP Provider Accounts"
-msgstr "SIP提供者账户"
-
-msgid "SIP Realm (needed by some providers)"
-msgstr "SIP Realm(一些供应商需要）"
-
-msgid "SIP Server/Registrar"
-msgstr "SIP注册服务器"
-
-msgid "SIP Server/Registrar Port"
-msgstr "SIP注册服务器端口"
-
-msgid "Server Setting"
-msgstr ""
-
-msgid "Server Setting for Local SIP Devices"
-msgstr ""
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr ""
-
-msgid "Service Status"
-msgstr ""
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr ""
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr ""
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr ""
-
-msgid "User Accounts"
-msgstr ""
-
-msgid "User Agent String"
-msgstr ""
-
-msgid "User Name"
-msgstr ""
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr ""
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr ""
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr ""
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-#~ msgid ""
-#~ "Designate numbers that are allowed to call through this system and which "
-#~ "user's privileges it will have."
-#~ msgstr "设定号码作为用户拥有使用交换机呼叫的权限。"
diff --git a/applications/luci-app-pbx/po/zh-tw/pbx.po b/applications/luci-app-pbx/po/zh-tw/pbx.po
deleted file mode 100644
index 603b9df585..0000000000
--- a/applications/luci-app-pbx/po/zh-tw/pbx.po
+++ /dev/null
@@ -1,507 +0,0 @@
-msgid ""
-msgstr ""
-"Project-Id-Version: PACKAGE VERSION\n"
-"PO-Revision-Date: 2014-05-16 13:59+0200\n"
-"Last-Translator: omnistack <omnistack@gmail.com>\n"
-"Language-Team: none\n"
-"Language: zh_TW\n"
-"MIME-Version: 1.0\n"
-"Content-Type: text/plain; charset=UTF-8\n"
-"Content-Transfer-Encoding: 8bit\n"
-"Plural-Forms: nplurals=1; plural=0;\n"
-"X-Generator: Pootle 2.0.6\n"
-
-msgid "Advanced Settings"
-msgstr "進階設定"
-
-msgid "Available"
-msgstr "可運用"
-
-msgid ""
-"Avoid using anything but alpha-numeric characters, space, comma, and period."
-msgstr "除了字母數字字符，空格，逗號和句號其它一概不用."
-
-msgid "Away"
-msgstr "離線"
-
-msgid "Blacklisted Numbers"
-msgstr "列入黑名單號碼"
-
-msgid "Call Routing"
-msgstr "路由呼叫"
-
-msgid "Call-back Numbers"
-msgstr ""
-
-msgid "Call-back Provider"
-msgstr ""
-
-msgid "Call-through Numbers"
-msgstr "通話接通號碼"
-
-msgid "Copy-paste large lists of numbers here."
-msgstr "號碼大型清單複製貼上此地"
-
-msgid ""
-"Designate numbers that are allowed to call through this system and which "
-"user's privileges they will have."
-msgstr ""
-
-msgid ""
-"Designate numbers to whom the system will hang up and call back, which "
-"provider will be used to call them, and which user's privileges will be "
-"granted to them."
-msgstr ""
-
-msgid "Dials numbers unmatched elsewhere"
-msgstr "撥號它處號碼不符"
-
-msgid "Do Not Disturb"
-msgstr "勿擾中"
-
-msgid "Domain/IP Address/Dynamic Domain"
-msgstr "網域/IP位址/動態網域"
-
-msgid "Dynamic List of Blacklisted Numbers"
-msgstr "黑名單動態列表"
-
-msgid "Email"
-msgstr "郵件信箱"
-
-msgid "Enable Incoming Calls (Register via SIP)"
-msgstr "啟用來話呼叫(透過SIP註冊)"
-
-msgid "Enable Incoming Calls (set Status below)"
-msgstr "啟用來話呼叫(在下面設定狀態)"
-
-msgid "Enable Outgoing Calls"
-msgstr "啟用外撥"
-
-msgid "Enabled"
-msgstr "已啟用"
-
-msgid ""
-"Enter a VoIP provider to use for call-back in the format username@some.host."
-"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste "
-"the providers from above. Invalid entries, including providers not enabled "
-"for outgoing calls, will be rejected silently."
-msgstr ""
-
-msgid ""
-"Enter phone numbers that you want to decline calls from automatically. You "
-"should probably omit the country code and any leading zeroes, but please "
-"experiment to make sure you are blocking numbers from your desired area "
-"successfully."
-msgstr ""
-"打入您允許自動通話的號碼. 您或許可以忽略國碼和0字開頭, 但僅供實驗以確保期望區"
-"的號碼被阻斷成功."
-
-msgid ""
-"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
-"you will use ONLY locally and never from a remote location."
-msgstr ""
-"要設定SIP設備在Server/Registrar內打入IP(或IP:埠號)您僅能本地端使用絕不要打入"
-"遠端位置"
-
-msgid ""
-"Enter this hostname (or hostname:port) in the Server/Registrar setting of "
-"SIP devices you will use from a remote location (they will work locally too)."
-msgstr ""
-"要設定SIP設備從遠端使用(在本地端也一樣能執行),在Server/Registrar內打入主機名"
-"稱(或主機名稱:埠號)"
-
-msgid "External SIP Port"
-msgstr "外部SIP埠號"
-
-msgid ""
-"For each provider enabled for incoming calls, here you can restrict which "
-"users to ring on incoming calls. If the list is empty, the system will "
-"indicate that all users enabled for incoming calls will ring. Invalid "
-"usernames will be rejected silently. Also, entering a username here "
-"overrides the user's setting to not receive incoming calls. This way, you "
-"can make certain users ring only for specific providers. Entries can be made "
-"in a space-separated list, and/or one per line by hitting enter after every "
-"one."
-msgstr ""
-
-msgid ""
-"For each user enabled for outgoing calls you can restrict what providers the "
-"user can use for outgoing calls. By default all users can use all providers. "
-"To show up in the list below the user should be allowed to make outgoing "
-"calls in the \"User Accounts\" page. Enter VoIP providers in the format "
-"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
-"to copy and paste the providers from above. Invalid entries, including "
-"providers not enabled for outgoing calls, will be rejected silently. Entries "
-"can be made in a space-separated list, and/or one per line by hitting enter "
-"after every one."
-msgstr ""
-
-msgid "Full Name"
-msgstr "全名"
-
-msgid "General Settings"
-msgstr "一般設定"
-
-msgid "Google Accounts"
-msgstr "Google帳戶"
-
-msgid "Google Talk Status"
-msgstr "Google Talk狀態"
-
-msgid "Google Talk Status Message"
-msgstr "Google Talk訊息狀態"
-
-msgid "Google Voice/Talk Accounts"
-msgstr "Google 語音/簡訊帳戶"
-
-msgid "Hang-up Delay"
-msgstr ""
-
-msgid ""
-"Here you must configure at least one SIP account, that you will use to "
-"register with this service. Use this account either in an Analog Telephony "
-"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
-"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
-"default, all SIP accounts will ring simultaneously if a call is made to one "
-"of your VoIP provider accounts or GV numbers."
-msgstr ""
-
-msgid ""
-"How long to wait before hanging up. If the provider you use to dial "
-"automatically forwards to voicemail, you can set this value to a delay that "
-"will allow you to hang up before your call gets forwarded and you get billed "
-"for it."
-msgstr ""
-
-msgid ""
-"If setting Server/Registrar to %s or %s does not work for you, try setting "
-"it to %s or %s and entering this port number in a separate field that "
-"specifies the Server/Registrar port number. Beware that some devices have a "
-"confusing setting that sets the port where SIP requests originate from on "
-"the SIP device itself (the bind port). The port specified on this page is "
-"NOT this bind port but the port this service listens on."
-msgstr ""
-
-msgid ""
-"If you experience jittery or high latency audio during heavy downloads, you "
-"may want to enable QoS. QoS prioritizes traffic to and from your network for "
-"specified ports and IP addresses, resulting in better latency and throughput "
-"for sound in our case. If enabled below, a QoS rule for this service will be "
-"configured by the PBX automatically, but you must visit the QoS "
-"configuration page (Network->QoS) to configure other critical QoS settings "
-"like Download and Upload speed."
-msgstr ""
-
-msgid ""
-"If you have more than one account that can make outgoing calls, you should "
-"enter a list of phone numbers and/or prefixes in the following fields for "
-"each provider listed. Invalid prefixes are removed silently, and only 0-9, "
-"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
-"matches 1-9, and N matches 2-9. For example to make calls to Germany through "
-"a provider, you can enter 49. To make calls to North America, you can enter "
-"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
-"code like New York's 646, you can enter 646NXXXXXX for that provider. You "
-"should leave one account with an empty list to make calls with it by "
-"default, if no other provider's prefixes match. The system will "
-"automatically replace an empty list with a message that the provider dials "
-"all numbers not matched by another provider's prefixes. Be as specific as "
-"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
-"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
-"space-separated list, and/or one per line by hitting enter after every one."
-msgstr ""
-
-msgid "Incoming Calls"
-msgstr "來電呼叫"
-
-msgid "Insert QoS Rules"
-msgstr "插入QoS規則"
-
-msgid "Makes Outgoing Calls"
-msgstr "開啟外撥"
-
-msgid "NOTE: There are no Google or SIP provider accounts configured."
-msgstr "注意:尚缺Google或者SIP提供者帳戶被設置"
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for incoming "
-"calls."
-msgstr "注意:尚缺Google或者SIP供應商帳戶被啟用才能接收來電呼叫"
-
-msgid ""
-"NOTE: There are no Google or SIP provider accounts enabled for outgoing "
-"calls."
-msgstr "注意:尚缺Google或者SIP供應商帳戶被啟用才能外撥."
-
-msgid "NOTE: There are no local user accounts configured."
-msgstr "注意:尚未設置本地端帳戶"
-
-msgid "NOTE: There are no local user accounts enabled for outgoing calls."
-msgstr "注意:啟用本地端帳戶才能外撥"
-
-msgid "No"
-msgstr "不"
-
-msgid "Number of Seconds to Ring"
-msgstr "響鈴秒數"
-
-msgid "Outbound Proxy"
-msgstr "外連代理"
-
-msgid "Outgoing Calls"
-msgstr "去電外撥"
-
-msgid "PBX Main Page"
-msgstr "PBX總機主頁"
-
-msgid "PBX Service Status"
-msgstr "PBX服務狀態"
-
-msgid "PIN"
-msgstr "PIN碼"
-
-msgid "Password"
-msgstr "密碼"
-
-msgid ""
-"Pick a random port number between 6500 and 9500 for the service to listen "
-"on. Do not pick the standard 5060, because it is often subject to brute-"
-"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in "
-"the \"SIP Device/Softphone Accounts\" section for updated Server and Port "
-"settings for your SIP Devices/Softphones."
-msgstr ""
-
-msgid "Port Setting for SIP Devices"
-msgstr "SIP設備的埠號設置"
-
-msgid "Providers Used for Outgoing Calls"
-msgstr "已採用的外撥供應商"
-
-msgid "QoS Settings"
-msgstr "QoS語音品質設置"
-
-msgid "RTP Port Range End"
-msgstr "RTP協定埠域結束"
-
-msgid "RTP Port Range Start"
-msgstr "RTP協定埠域啟始"
-
-msgid ""
-"RTP traffic carries actual voice packets. This is the start of the port "
-"range that will be used for setting up RTP communication. It's usually OK to "
-"leave this at the default value."
-msgstr ""
-
-msgid "Receives Incoming Calls"
-msgstr "接受來電呼叫"
-
-msgid "Remote Usage"
-msgstr "遠端啟用"
-
-msgid "Rings users enabled for incoming calls"
-msgstr "來電呼叫時震鈴通知使用者"
-
-msgid "SIP Accounts"
-msgstr "SIP帳戶"
-
-msgid "SIP Device/Softphone Accounts"
-msgstr "SIP設備/軟體式手機帳戶"
-
-msgid "SIP Provider Accounts"
-msgstr "SIP供應商帳戶"
-
-msgid "SIP Realm (needed by some providers)"
-msgstr "SIP領域(某些供應商需用到)"
-
-msgid "SIP Server/Registrar"
-msgstr "SIP伺服器/登記處"
-
-msgid "SIP Server/Registrar Port"
-msgstr "SIP伺服器/登記埠"
-
-msgid "Server Setting"
-msgstr "伺服器設置"
-
-msgid "Server Setting for Local SIP Devices"
-msgstr "本地SIP設備的伺服器設置"
-
-msgid "Server Setting for Remote SIP Devices"
-msgstr "遠端SIP設備的伺服器設置"
-
-msgid "Service Status"
-msgstr "服務狀態"
-
-msgid ""
-"Set the number of seconds to ring users upon incoming calls before hanging "
-"up or going to voicemail, if the voicemail is installed and enabled."
-msgstr ""
-
-msgid "Space-Separated List of Blacklisted Numbers"
-msgstr "以空格分隔的黑名單號碼列表"
-
-msgid "Specify numbers individually here. Press enter to add more numbers."
-msgstr "在此指定獨立號碼. 按enter 可新增更多號碼"
-
-msgid ""
-"Specify numbers individually here. Press enter to add more numbers. You will "
-"have to experiment with what country and area codes you need to add to the "
-"number."
-msgstr ""
-
-msgid ""
-"The number(s) specified above will be able to dial out with this user's "
-"providers. Invalid usernames, including users not enabled for outgoing "
-"calls, are dropped silently. Please verify that the entry was accepted."
-msgstr ""
-
-msgid ""
-"This configuration page allows you to configure a phone system (PBX) service "
-"which permits making phone calls through multiple Google and SIP (like "
-"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
-"devices. Note that Google accounts, SIP accounts, and local user accounts "
-"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
-"Accounts\" sub-sections. You must add at least one User Account to this PBX, "
-"and then configure a SIP device or softphone to use the account, in order to "
-"make and receive calls with your Google/SIP accounts. Configuring multiple "
-"users will allow you to make free calls between all users, and share the "
-"configured Google and SIP accounts. If you have more than one Google and SIP "
-"accounts set up, you should probably configure how calls to and from them "
-"are routed in the \"Call Routing\" page. If you're interested in using your "
-"own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
-"in the \"Advanced Settings\" page."
-msgstr ""
-
-msgid ""
-"This is the name that the VoIP server will use to identify itself when "
-"registering to VoIP (SIP) providers. Some providers require this to a "
-"specific string matching a hardware SIP device."
-msgstr ""
-
-msgid ""
-"This is where you indicate which Google/SIP accounts are used to call what "
-"country/area codes, which users can use what SIP/Google accounts, how "
-"incoming calls are routed, what numbers can get into this PBX with a "
-"password, and what numbers are blacklisted."
-msgstr ""
-
-msgid ""
-"This is where you set up your Google (Talk and Voice) Accounts, in order to "
-"start using them for dialing and receiving calls (voice chat and real phone "
-"calls). Please make at least one voice call using the Google Talk plugin "
-"installable through the GMail interface, and then log out from your account "
-"everywhere. Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
-"SipSorcery, the popular Betamax providers, and any other providers with SIP "
-"settings in order to start using them for dialing and receiving calls (SIP "
-"uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
-msgstr ""
-
-msgid ""
-"This option should be set to \"Yes\" if you have a DID (real telephone "
-"number) associated with this SIP account or want to receive SIP uri calls "
-"through this provider."
-msgstr ""
-
-msgid ""
-"This section contains settings that do not need to be changed under normal "
-"circumstances. In addition, here you can configure your system for use with "
-"remote SIP devices, and resolve call quality issues by enabling the "
-"insertion of QoS rules."
-msgstr ""
-
-msgid ""
-"Use (four to five digit) numeric user name if you are connecting normal "
-"telephones with ATAs to this system (so they can dial user names)."
-msgstr ""
-
-msgid ""
-"Use this account to make outgoing calls as configured in the \"Call Routing"
-"\" section."
-msgstr ""
-
-msgid "Use this account to make outgoing calls."
-msgstr "使用這個帳號外撥."
-
-msgid "User Accounts"
-msgstr "使用者帳號"
-
-msgid "User Agent String"
-msgstr "用戶代理字串"
-
-msgid "User Name"
-msgstr "用戶名稱"
-
-msgid "Uses providers enabled for outgoing calls"
-msgstr "採用供應商啟用以便外撥"
-
-msgid ""
-"When somebody starts voice chat with your GTalk account or calls the GVoice, "
-"number (if you have Google Voice), the call will be forwarded to any users "
-"that are online (registered using a SIP device or softphone) and permitted "
-"to receive the call. If you have Google Voice, you must go to your GVoice "
-"settings and forward calls to Google chat in order to actually receive calls "
-"made to your GVoice number. If you have trouble receiving calls from GVoice, "
-"experiment with the Call Screening option in your GVoice Settings. Finally, "
-"make sure no other client is online with this account (browser in gmail, "
-"mobile/desktop Google Talk App) as it may interfere."
-msgstr ""
-
-msgid ""
-"When your password is saved, it disappears from this field and is not "
-"displayed for your protection. The previously saved password will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-
-msgid "Yes"
-msgstr "是"
-
-msgid ""
-"You can enter your domain name, external IP address, or dynamic domain name "
-"here. The best thing to input is a static IP address. If your IP address is "
-"dynamic and it changes, your configuration will become invalid. Hence, it's "
-"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS "
-"hostname here. You can configure Dynamic DNS with the luci-app-ddns package."
-msgstr ""
-
-msgid "You can specify a real name to show up in the Caller ID here."
-msgstr "您可以在此指定一個真實名稱以便顯示在來電ID"
-
-msgid ""
-"You can use your SIP devices/softphones with this system from a remote "
-"location as well, as long as your Internet Service Provider gives you a "
-"public IP. You will be able to call other local users for free (e.g. other "
-"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
-"as if you were local to the PBX. After configuring this tab, go back to "
-"where users are configured and see the new Server and Port setting you need "
-"to configure the remote SIP devices with. Please note that if this PBX is "
-"not running on your router/gateway, you will need to configure port "
-"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
-"port and RTP range) to the IP address of the device running this PBX."
-msgstr ""
-"不管多遠,只要能取得ISP提供的公眾合法IP,依然可以使用這個系統的SIP設備/PC電話表"
-"現一樣的好.您將能夠免費呼叫其他本地端用戶(e.g 其它類比電話機(ATAs)和使用您的"
-"VoIP供應商講電話就像您在使用本地的PBX總機電話一樣.在設定這個標籤後, 需回到用"
-"戶設定並且查看新的伺服器和埠設定以便能設定遠端SIP設備.請注意假如PBX總機若不在"
-"您的路由器/GW上執行,您將必須在您的路由器/GW上設定埠轉發(NAT).在PBX主機上請轉"
-"發下列(SIP埠+RTP所採用的範圍埠)埠號址定到跑PBX服務設備上的IP位址."
-
-msgid ""
-"Your PIN disappears when saved for your protection. It will be changed only "
-"when you enter a value different from the saved one. Leaving the PIN empty "
-"is possible, but please beware of the security implications."
-msgstr ""
-"當存檔時為保護起見您的PIN碼將不會顯示. 除非您打入不同於原始存檔的值它才會變"
-"更. 也可以把PIN碼保留空白, 但要提防會有安全的隱憂."
-
-msgid ""
-"Your password disappears when saved for your protection. It will be changed "
-"only when you enter a value different from the saved one."
-msgstr ""
-"當存檔時為保護起見您的密碼將不會顯示. 除非您打入不同於原始存檔的值它才會變更."
-
-#~ msgid ""
-#~ "Designate numbers that are allowed to call through this system and which "
-#~ "user's privileges it will have."
-#~ msgstr "依據系統和戶用的權限允許通話的指定號碼"
diff --git a/applications/luci-app-pbx/root/etc/config/pbx b/applications/luci-app-pbx/root/etc/config/pbx
deleted file mode 100644
index ca7c1669d0..0000000000
--- a/applications/luci-app-pbx/root/etc/config/pbx
+++ /dev/null
@@ -1 +0,0 @@
-config 'main' 'connection_status'
diff --git a/applications/luci-app-pbx/root/etc/config/pbx-advanced b/applications/luci-app-pbx/root/etc/config/pbx-advanced
deleted file mode 100644
index 39da6f880c..0000000000
--- a/applications/luci-app-pbx/root/etc/config/pbx-advanced
+++ /dev/null
@@ -1,5 +0,0 @@
-config 'settings' 'advanced'
-        option 'useragent' 'PBX'
-        option 'ringtime' '30'
-        option 'rtpstart' '19850'
-        option 'rtpend' '19900'
diff --git a/applications/luci-app-pbx/root/etc/config/pbx-calls b/applications/luci-app-pbx/root/etc/config/pbx-calls
deleted file mode 100644
index 822bd4a1be..0000000000
--- a/applications/luci-app-pbx/root/etc/config/pbx-calls
+++ /dev/null
@@ -1,7 +0,0 @@
-config 'call_routing' 'outgoing_calls'
-
-config 'call_routing' 'incoming_calls'
-
-config 'call_routing' 'providers_user_can_use'
-
-config 'call_routing' 'blacklisting'
diff --git a/applications/luci-app-pbx/root/etc/config/pbx-google b/applications/luci-app-pbx/root/etc/config/pbx-google
deleted file mode 100644
index e69de29bb2..0000000000
--- a/applications/luci-app-pbx/root/etc/config/pbx-google
+++ /dev/null
diff --git a/applications/luci-app-pbx/root/etc/config/pbx-users b/applications/luci-app-pbx/root/etc/config/pbx-users
deleted file mode 100644
index a4277b1bfe..0000000000
--- a/applications/luci-app-pbx/root/etc/config/pbx-users
+++ /dev/null
@@ -1 +0,0 @@
-config 'user' 'server'
diff --git a/applications/luci-app-pbx/root/etc/config/pbx-voip b/applications/luci-app-pbx/root/etc/config/pbx-voip
deleted file mode 100644
index e69de29bb2..0000000000
--- a/applications/luci-app-pbx/root/etc/config/pbx-voip
+++ /dev/null
diff --git a/applications/luci-app-pbx/root/etc/init.d/pbx-asterisk b/applications/luci-app-pbx/root/etc/init.d/pbx-asterisk
deleted file mode 100755
index e05ae11cd6..0000000000
--- a/applications/luci-app-pbx/root/etc/init.d/pbx-asterisk
+++ /dev/null
@@ -1,837 +0,0 @@
-#!/bin/sh /etc/rc.common
-#
-# Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com>
-#
-#    This file is part of luci-pbx.
-#
-#    luci-pbx is free software: you can redistribute it and/or modify
-#    it under the terms of the GNU General Public License as published by
-#    the Free Software Foundation, either version 3 of the License, or
-#    (at your option) any later version.
-#
-#    luci-pbx is distributed in the hope that it will be useful,
-#    but WITHOUT ANY WARRANTY; without even the implied warranty of
-#    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-#    GNU General Public License for more details.
-#
-#    You should have received a copy of the GNU General Public License
-#    along with luci-pbx.  If not, see <http://www.gnu.org/licenses/>.
-
-. /lib/functions.sh
-
-START=60
-
-# Some global variables
-MODULENAME=pbx
-USERAGENT="PBX"
-HANGUPCNTXT=hangup-call-context
-GTALKUNVL=unavailable
-
-ASTUSER=nobody
-ASTGROUP=nogroup
-ASTDIRSRECURSIVE="/var/run/asterisk /var/log/asterisk /var/spool/asterisk"
-ASTDIRS="/usr/lib/asterisk"
-ASTSOUNDSDIR="/usr/lib/asterisk/sounds"
-
-TEMPLATEDIR=/etc/${MODULENAME}-asterisk
-PBXSOUNDSDIR=$TEMPLATEDIR/sounds
-VMTEMPLATEDIR=/etc/${MODULENAME}-voicemail
-VMSOUNDSDIR=$VMTEMPLATEDIR/sounds
-ASTERISKDIR=/etc/asterisk
-WORKDIR=/tmp/$MODULENAME.$$
-MD5SUMSFILE=/tmp/$MODULENAME-sums.$$
-
-TMPL_ASTERISK=$TEMPLATEDIR/asterisk.conf.TEMPLATE
-TMPL_GTALK=$TEMPLATEDIR/gtalk.conf.TEMPLATE
-TMPL_INDICATIONS=$TEMPLATEDIR/indications.conf.TEMPLATE
-TMPL_LOGGER=$TEMPLATEDIR/logger.conf.TEMPLATE
-TMPL_MANAGER=$TEMPLATEDIR/manager.conf.TEMPLATE
-TMPL_MODULES=$TEMPLATEDIR/modules.conf.TEMPLATE
-TMPL_RTP=$TEMPLATEDIR/rtp.conf.TEMPLATE
-
-TMPL_EXTCTHRUCHECKHDR=$TEMPLATEDIR/extensions_disa-check_header.conf.TEMPLATE
-TMPL_EXTCTHRUCHECK=$TEMPLATEDIR/extensions_disa-check.conf.TEMPLATE
-TMPL_EXTCTHRUCHECKFTR=$TEMPLATEDIR/extensions_disa-check_footer.conf.TEMPLATE
-TMPL_EXTCTHRUHDR=$TEMPLATEDIR/extensions_disa_header.conf.TEMPLATE
-TMPL_EXTCTHRU=$TEMPLATEDIR/extensions_disa.conf.TEMPLATE
-TMPL_EXTCTHRUNOPIN=$TEMPLATEDIR/extensions_disa-nopin.conf.TEMPLATE
-
-TMPL_EXTCBACKCHECKHDR=$TEMPLATEDIR/extensions_callback-check_header.conf.TEMPLATE
-TMPL_EXTCBACKCHECK=$TEMPLATEDIR/extensions_callback-check.conf.TEMPLATE
-TMPL_EXTCBACKCHECKFTR=$TEMPLATEDIR/extensions_callback-check_footer.conf.TEMPLATE
-TMPL_EXTCBACKHDR=$TEMPLATEDIR/extensions_callback_header.conf.TEMPLATE
-TMPL_EXTCBACKSIP=$TEMPLATEDIR/extensions_callback_sip.conf.TEMPLATE
-TMPL_EXTCBACKGTALK=$TEMPLATEDIR/extensions_callback_gtalk.conf.TEMPLATE
-
-TMPL_EXTENSIONS=$TEMPLATEDIR/extensions.conf.TEMPLATE
-
-TMPL_EXTVMDISABLED=$TEMPLATEDIR/extensions_voicemail_disabled.conf.TEMPLATE
-TMPL_EXTVMENABLED=$TEMPLATEDIR/extensions_voicemail_enabled.conf.TEMPLATE
-
-TMPL_EXTBLKLIST=$TEMPLATEDIR/extensions_blacklist.conf.TEMPLATE
-TMPL_EXTBLKLISTFTR=$TEMPLATEDIR/extensions_blacklist_footer.conf.TEMPLATE
-TMPL_EXTBLKLISTHDR=$TEMPLATEDIR/extensions_blacklist_header.conf.TEMPLATE
-
-TMPL_EXTDEFAULT=$TEMPLATEDIR/extensions_default.conf.TEMPLATE
-TMPL_EXTDEFAULTUSER=$TEMPLATEDIR/extensions_default_user.conf.TEMPLATE
-
-TMPL_EXTINCNTXTSIP=$TEMPLATEDIR/extensions_incoming_context_sip.conf.TEMPLATE
-TMPL_EXTINCNTXTGTALKHDR=$TEMPLATEDIR/extensions_incoming_context_gtalk_header.conf.TEMPLATE
-TMPL_EXTINCNTXTGTALK=$TEMPLATEDIR/extensions_incoming_context_gtalk.conf.TEMPLATE
-
-TMPL_EXTUSERCNTXT=$TEMPLATEDIR/extensions_user_context.conf.TEMPLATE
-TMPL_EXTUSERCNTXTFTR=$TEMPLATEDIR/extensions_user_context_footer.conf.TEMPLATE
-TMPL_EXTUSERCNTXTHDR=$TEMPLATEDIR/extensions_user_context_header.conf.TEMPLATE
-
-TMPL_EXTOUTHDR=$TEMPLATEDIR/extensions_default_outgoing_header.conf.TEMPLATE
-TMPL_EXTOUTGTALK=$TEMPLATEDIR/extensions_outgoing_gtalk.conf.TEMPLATE
-TMPL_EXTOUTLOCAL=$TEMPLATEDIR/extensions_outgoing_dial_local_user.conf.TEMPLATE
-TMPL_EXTOUTSIP=$TEMPLATEDIR/extensions_outgoing_sip.conf.TEMPLATE
-
-TMPL_JABBER=$TEMPLATEDIR/jabber.conf.TEMPLATE
-TMPL_JABBERUSER=$TEMPLATEDIR/jabber_users.conf.TEMPLATE
-TMPL_SIP=$TEMPLATEDIR/sip.conf.TEMPLATE
-TMPL_SIPPEER=$TEMPLATEDIR/sip_peer.TEMPLATE
-TMPL_SIPREG=$TEMPLATEDIR/sip_registration.TEMPLATE
-TMPL_SIPUSR=$TEMPLATEDIR/sip_user.TEMPLATE
-
-TMPL_MSMTPDEFAULT=$VMTEMPLATEDIR/pbx-msmtprc-defaults.TEMPLATE
-TMPL_MSMTPACCOUNT=$VMTEMPLATEDIR/pbx-msmtprc-account.TEMPLATE
-TMPL_MSMTPAUTH=$VMTEMPLATEDIR/pbx-msmtprc-account-auth.TEMPLATE
-TMPL_MSMTPACCTDFLT=$VMTEMPLATEDIR/pbx-msmtprc-account-default.TEMPLATE
-
-
-INCLUDED_FILES="$WORKDIR/extensions_blacklist.conf $WORKDIR/extensions_callthrough.conf\
-                $WORKDIR/extensions_incoming.conf  $WORKDIR/extensions_incoming_gtalk.conf\
-                $WORKDIR/extensions_user.conf      $WORKDIR/jabber_users.conf\
-                $WORKDIR/sip_peers.conf            $WORKDIR/sip_registrations.conf\
-                $WORKDIR/sip_users.conf            $WORKDIR/extensions_voicemail.conf\
-                $WORKDIR/extensions_default.conf"
-
-
-# In this string, we concatenate all local users enabled to receive calls
-# readily formatted for the Dial command.
-localusers_to_ring=""
-
-# In this string, we keep a list of all users that are enabled for outgoing
-# calls. It is used at the end to create the user contexts.
-localusers_can_dial=""
-
-# In this string, we put together a space-separated list of provider names
-# (alphanumeric, with all non-alpha characters replaced with underscores),
-# which will be used to dial out by default (whose outgoing contexts will
-# be included in users' contexts by default.
-outbound_providers=""
-sip_outbound_providers=""
-gtalk_outbound_providers=""
-
-# Function which escapes non-alpha-numeric characters in a string
-escape_non_alpha() {
-    echo $@ | sed 's/\([^a-zA-Z0-9]\)/\\\1/g'
-}
-
-# Function which replaces non-alpha-numeric characters with an underscore
-sub_underscore_for_non_alpha() {
-    echo $@ | sed 's/[^a-zA-Z0-9]/_/g'
-}
-
-# Copies the template files which we don't edit.
-copy_unedited_templates_over()
-{
-    cp $TMPL_ASTERISK      $WORKDIR/asterisk.conf
-    cp $TMPL_GTALK         $WORKDIR/gtalk.conf
-    cp $TMPL_INDICATIONS   $WORKDIR/indications.conf
-    cp $TMPL_LOGGER        $WORKDIR/logger.conf
-    cp $TMPL_MANAGER       $WORKDIR/manager.conf
-    cp $TMPL_MODULES       $WORKDIR/modules.conf
-    # If this file isn't present at this stage, voicemail is disabled.
-    [ ! -f $WORKDIR/extensions_voicemail.conf ] && \
-      cp $TMPL_EXTVMDISABLED $WORKDIR/extensions_voicemail.conf
-}
-
-# Touches all the included files, to prevent asterisk from refusing to
-# start if a config item is missing and an included config file isn't created.
-create_included_files()
-{
-    touch $INCLUDED_FILES
-}
-
-# Puts together all the extensions.conf related configuration.
-pbx_create_extensions_config()
-{
-    local ringtime
-    config_get ringtime advanced ringtime
-
-    sed "s/|RINGTIME|/$ringtime/" $TMPL_EXTENSIONS > $WORKDIR/extensions.conf
-    mv  $WORKDIR/inext.TMP                           $WORKDIR/extensions_incoming.conf
-    cp  $TMPL_EXTINCNTXTGTALKHDR                     $WORKDIR/extensions_incoming_gtalk.conf
-    cat $WORKDIR/outextgtalk.TMP                  >> $WORKDIR/extensions_incoming_gtalk.conf 2>/dev/null
-    rm -f $WORKDIR/outextgtalk.TMP
-    mv  $WORKDIR/blacklist.TMP                       $WORKDIR/extensions_blacklist.conf
-    mv  $WORKDIR/userext.TMP                         $WORKDIR/extensions_user.conf
-
-    cp  $TMPL_EXTCTHRUHDR                            $WORKDIR/extensions_callthrough.conf
-    cat $WORKDIR/callthrough.TMP                  >> $WORKDIR/extensions_callthrough.conf 2>/dev/null
-    rm -f $WORKDIR/callthrough.TMP
-    cat $TMPL_EXTCTHRUCHECKHDR                    >> $WORKDIR/extensions_callthrough.conf 2>/dev/null
-    cat $WORKDIR/callthroughcheck.TMP             >> $WORKDIR/extensions_callthrough.conf 2>/dev/null
-    rm -f $WORKDIR/callthroughcheck.TMP
-    cat $TMPL_EXTCTHRUCHECKFTR                    >> $WORKDIR/extensions_callthrough.conf 2>/dev/null
-
-    cp  $TMPL_EXTCBACKHDR                            $WORKDIR/extensions_callback.conf
-    cat $WORKDIR/callback.TMP                     >> $WORKDIR/extensions_callback.conf 2>/dev/null
-    rm -f $WORKDIR/callback.TMP
-    cat $TMPL_EXTCBACKCHECKHDR                    >> $WORKDIR/extensions_callback.conf 2>/dev/null
-    cat $WORKDIR/callbackcheck.TMP                >> $WORKDIR/extensions_callback.conf 2>/dev/null
-    rm -f $WORKDIR/callbackcheck.TMP
-    cat $TMPL_EXTCBACKCHECKFTR                    >> $WORKDIR/extensions_callback.conf 2>/dev/null
-
-    rm -f $WORKDIR/outext-*.TMP
-    rm -f $WORKDIR/localext.TMP
-    sed "s/|LOCALUSERS|/$localusers_to_ring/g" $TMPL_EXTDEFAULT \
-                                                   > $WORKDIR/extensions_default.conf
-    cat $WORKDIR/inextuser.TMP                    >> $WORKDIR/extensions_default.conf
-    rm -f $WORKDIR/inextuser.TMP
-}
-
-# Puts together all the sip.conf related configuration.
-pbx_create_sip_config()
-{
-    mv $WORKDIR/sip_regs.TMP    $WORKDIR/sip_registrations.conf
-    mv $WORKDIR/sip_peers.TMP   $WORKDIR/sip_peers.conf
-    mv $WORKDIR/sip_users.TMP   $WORKDIR/sip_users.conf
-}
-
-# Creates the jabber.conf related config
-pbx_create_jabber_config()
-{
-    cp  $TMPL_JABBER        $WORKDIR/jabber.conf
-    mv  $WORKDIR/jabber.TMP $WORKDIR/jabber_users.conf
-}
-
-# Gets rid of any config files from $ASTERISKDIR not found in $WORKDIR.
-clean_up_asterisk_config_dir()
-{
-    for f in $ASTERISKDIR/* ; do
-        basef="`basename $f`"
-        if [ ! -e "$WORKDIR/$basef" ] ; then
-            rm -rf "$f"
-        fi
-    done
-}
-
-# Compares md5sums of the config files in $WORKDIR to those
-# in $ASTERISKDIR, and copies only changed files over to reduce
-# wear on flash in embedded devices.
-compare_configs_and_copy_changed()
-{
-    # First, compute md5sums of the config files in $WORKDIR.
-    cd $WORKDIR/
-    md5sum * > $MD5SUMSFILE
-
-    # Now, check the files in $ASTERISKDIR against the md5sums.
-    cd $ASTERISKDIR/
-    changed_files="`md5sum -c $MD5SUMSFILE 2>/dev/null | fgrep ": FAILED" | awk -F: '{print $1}'`"
-
-    rm -f $MD5SUMSFILE
-
-    [ -z "$changed_files" ] && return
-
-    # Now copy over the changed files.
-    for f in $changed_files ; do
-        cp "$WORKDIR/$f" "$ASTERISKDIR/$f"
-    done
-}
-
-# Calls the functions that create the final config files
-# Calls the function which compares which files have changed
-# Puts the final touches on $ASTERISKDIR
-# Gets rid of $WORKDIR
-pbx_assemble_and_copy_config()
-{
-    mkdir -p $ASTERISKDIR
-
-    copy_unedited_templates_over
-    create_included_files
-    pbx_create_extensions_config
-    pbx_create_sip_config
-    pbx_create_jabber_config
-
-    touch    $WORKDIR/features.conf
-
-    # At this point, $WORKDIR should contain a complete, working config.
-    clean_up_asterisk_config_dir
-
-    compare_configs_and_copy_changed
-
-    [ ! -d $ASTERISKDIR/manager.d ] && mkdir -p $ASTERISKDIR/manager.d/
-
-    # Get rid of the working directory
-    rm -rf $WORKDIR/
-}
-
-# Creates configuration for a user and adds it to the temporary file that holds
-# all users configured so far.
-pbx_add_user()
-{
-    local fullname
-    local defaultuser
-    local rawdefaultuser
-    local secret
-    local ring
-    local can_call
-
-    config_get fullname    $1 fullname
-    fullname=`escape_non_alpha $fullname`
-    config_get rawdefaultuser $1 defaultuser
-    defaultuser=`escape_non_alpha $rawdefaultuser`
-    config_get secret      $1 secret
-    secret=`escape_non_alpha $secret`
-    config_get ring        $1 ring
-    config_get can_call    $1 can_call
-
-    [ -z "$defaultuser" -o -z "$secret" ] && return
-    [ -z "$fullname" ] && fullname="$defaultuser"
-
-    sed "s/|DEFAULTUSER|/$defaultuser/g" $TMPL_SIPUSR > $WORKDIR/sip_user.tmp
-
-    if [ "$can_call" = "yes" ] ; then
-        # Add user to list of all users that are allowed to make calls.
-        localusers_can_dial="$localusers_can_dial $rawdefaultuser"
-        sed -i "s/|CONTEXTNAME|/$defaultuser/g"     $WORKDIR/sip_user.tmp
-    else
-        sed -i "s/|CONTEXTNAME|/$HANGUPCNTXT/g"     $WORKDIR/sip_user.tmp
-    fi
-
-    # Add this user's configuration to the temp file containing all user configs.
-    sed "s/|FULLNAME|/$fullname/" $WORKDIR/sip_user.tmp |\
-             sed "s/|SECRET|/$secret/g"                       >> $WORKDIR/sip_users.TMP
-
-    if [ "$ring" = "yes" ] ; then
-        if [ -z "$localusers_to_ring" ] ; then
-            localusers_to_ring="SIP\/$defaultuser"
-        else
-            localusers_to_ring="$localusers_to_ring\&SIP\/$defaultuser"
-        fi
-    fi
-
-    # Add configuration which allows local users to call each other.
-    sed "s/|DEFAULTUSER|/$defaultuser/g" $TMPL_EXTOUTLOCAL    >> $WORKDIR/localext.TMP
-
-    # Add configuration which puts calls to users through the default
-    # context, so that blacklists and voicemail take effect for this user.
-    sed "s/|DEFAULTUSER|/$defaultuser/g" $TMPL_EXTDEFAULTUSER >> $WORKDIR/inextuser.TMP
-
-    rm -f $WORKDIR/sip_user.tmp
-}
-
-# Creates configuration for a Google account, and adds it to the temporary file that holds
-# all accounts configured so far.
-# Also creates the outgoing extensions which are used in users' outgoing contexts.
-pbx_add_jabber()
-{
-    local username
-    local secret
-    local numprefix
-    local register
-    local make_outgoing_calls
-    local name
-    local users_to_ring
-    local status
-    local statusmessage
-
-    config_get username $1 username
-    username=`escape_non_alpha $username`
-    config_get secret $1 secret
-    secret=`escape_non_alpha $secret`
-    #TODO: Is this really necessary here? Numprefix is retrieved below.
-    config_get numprefix $1 numprefix
-    config_get register $1 register
-    config_get make_outgoing_calls $1 make_outgoing_calls
-    config_get name $1 name
-    config_get status $1 status
-    status=`escape_non_alpha $status`
-    config_get statusmessage $1 statusmessage
-    statusmessage=`escape_non_alpha $statusmessage`
-
-    [ -z "$username" -o -z "$secret" ] && return
-
-    # Construct a jabber entry for this provider.
-    sed     "s/|USERNAME|/$username/g" $TMPL_JABBERUSER |\
-            sed "s/|NAME|/$name/g"                            > $WORKDIR/jabber.tmp
-
-    if [ "$register" = yes ] ; then
-        # If this provider is enabled for incoming calls, we need to set the
-        # status of the user to something other than unavailable in order to receive calls.
-        sed -i "s/|STATUS|/$status/g"                   $WORKDIR/jabber.tmp
-        sed -i "s/|STATUSMESSAGE|/\"$statusmessage\"/g" $WORKDIR/jabber.tmp
-
-        users_to_ring="`uci -q get ${MODULENAME}-calls.incoming_calls.$name`"
-        # If no users have been specified to ring, we ring all users enabled for incoming calls.
-        if [ -z "$users_to_ring" ] ; then
-            users_to_ring=$localusers_to_ring
-        else
-            # Else, we cook up a string formatted for the Dial command
-            # with the specified users (SIP/user1&SIP/user2&...). We do it
-            # with set, shift and a loop in order to be more tolerant of ugly whitespace
-            # messes entered by users.
-            set $users_to_ring
-            users_to_ring="SIP\/$1" && shift
-            for u in $@ ; do u=`escape_non_alpha $u` ; users_to_ring=$users_to_ring\\\&SIP\\\/$u ; done
-        fi
-
-        # Now, we add this account to the gtalk incoming context.
-        sed    "s/|USERNAME|/$username/g" $TMPL_EXTINCNTXTGTALK |\
-                   sed "s/|LOCALUSERS|/$users_to_ring/g"    >> $WORKDIR/outextgtalk.TMP
-    else
-        sed -i "s/|STATUS|/$GTALKUNVL/g"               $WORKDIR/jabber.tmp
-        sed -i "s/|STATUSMESSAGE|/\"\"/g"              $WORKDIR/jabber.tmp
-    fi
-
-    # Add this account's configuration to the temp file containing all account configs.
-    sed "s/|SECRET|/$secret/g" $WORKDIR/jabber.tmp      >> $WORKDIR/jabber.TMP
-
-    # If this provider is enabled for outgoing calls.
-    if [ "$make_outgoing_calls" = "yes" ] ; then
-
-        numprefix="`uci -q get ${MODULENAME}-calls.outgoing_calls.$name`"
-
-        # If no prefixes are specified, then we use "X" which matches any prefix.
-        [ -z "$numprefix" ]  && numprefix="X"
-
-        for p in $numprefix ; do
-            p=`escape_non_alpha $p`
-            sed "s/|NUMPREFIX|/$p/g" $TMPL_EXTOUTGTALK |\
-                            sed "s/|NAME|/$name/g"            >> $WORKDIR/outext-$name.TMP
-        done
-
-        # Add this provider to the list of enabled outbound providers.
-        if [ -z "$outbound_providers" ] ; then
-            outbound_providers="$name"
-        else
-            outbound_providers="$outbound_providers $name"
-        fi
-
-        # Add this provider to the list of enabled gtalk outbound providers.
-        if [ -z "$gtalk_outbound_providers" ] ; then
-            gtalk_outbound_providers="$name"
-        else
-            gtalk_outbound_providers="$gtalk_outbound_providers $name"
-        fi
-    fi
-
-    rm -f $WORKDIR/jabber.tmp
-}
-
-# Creates configuration for a SIP provider account, and adds it to the temporary file that holds
-# all accounts configured so far.
-# Also creates the outgoing extensions which are used in users' outgoing contexts.
-pbx_add_peer()
-{
-    local defaultuser
-    local secret
-    local host
-    local fromdomain
-    local register
-    local numprefix
-    local make_outgoing_calls
-    local name
-    local users_to_ring
-    local port
-    local outboundproxy
-
-    config_get defaultuser $1 defaultuser
-    defaultuser=`escape_non_alpha $defaultuser`
-    config_get secret $1 secret
-    secret=`escape_non_alpha $secret`
-    config_get host $1 host
-    host=`escape_non_alpha $host`
-    config_get port $1 port
-    config_get outbountproxy $1 outboundproxy
-    outbountproxy=`escape_non_alpha $outbountproxy`
-    config_get fromdomain $1 fromdomain
-    fromdomain=`escape_non_alpha $fromdomain`
-    config_get register $1 register
-    config_get numprefix $1 numprefix
-    config_get make_outgoing_calls $1 make_outgoing_calls
-    config_get name $1 name
-
-    [ -z "$defaultuser" -o -z "$secret" -o -z "$host" ] && return
-    [ -z "$fromdomain" ]    && fromdomain=$host
-    [ -n "$port" ]          && port="port=$port"
-    [ -n "$outboundproxy" ] && outboundproxy="outboundproxy=$outboundproxy"
-
-    # Construct a sip peer entry for this provider.
-    sed    "s/|DEFAULTUSER|/$defaultuser/" $TMPL_SIPPEER  >  $WORKDIR/sip_peer.tmp
-    sed -i "s/|NAME|/$name/"                                 $WORKDIR/sip_peer.tmp
-    sed -i "s/|FROMUSER|/$defaultuser/"                      $WORKDIR/sip_peer.tmp
-    sed -i "s/|SECRET|/$secret/"                             $WORKDIR/sip_peer.tmp
-    sed -i "s/|HOST|/$host/"                                 $WORKDIR/sip_peer.tmp
-    sed -i "s/|PORT|/$port/"                                 $WORKDIR/sip_peer.tmp
-    sed -i "s/|OUTBOUNDPROXY|/$outboundproxy/"               $WORKDIR/sip_peer.tmp
-    # Add this account's configuration to the temp file containing all account configs.
-    sed    "s/|FROMDOMAIN|/$host/" $WORKDIR/sip_peer.tmp >>  $WORKDIR/sip_peers.TMP
-
-    # If this provider is enabled for incoming calls.
-    if [ "$register" = "yes" ] ; then
-        # Then we create a registration string for this provider.
-        sed    "s/|DEFAULTUSER|/$defaultuser/g" $TMPL_SIPREG         > $WORKDIR/sip_reg.tmp
-        sed -i "s/|SECRET|/$secret/g"                                  $WORKDIR/sip_reg.tmp
-        sed    "s/|NAME|/$name/g" $WORKDIR/sip_reg.tmp              >> $WORKDIR/sip_regs.TMP
-
-        users_to_ring="`uci -q get ${MODULENAME}-calls.incoming_calls.$name`"
-        # If no users have been specified to ring, we ring all users enabled for incoming calls.
-        if [ -z "$users_to_ring" ] ; then
-            users_to_ring=$localusers_to_ring
-        else
-            # Else, we cook up a string formatted for the Dial command
-            # with the specified users (SIP/user1&SIP/user2&...). We do it
-            # with set, shift and a loop in order to be more tolerant of ugly whitespace
-            # messes entered by users.
-            set $users_to_ring
-            users_to_ring="SIP\/$1" && shift
-            for u in $@ ; do users_to_ring=$users_to_ring\\\&SIP\\\/$u ; done
-        fi
-
-        # And we create an incoming calls context for this provider.
-        sed    "s/|NAME|/$name/g"                          $TMPL_EXTINCNTXTSIP |\
-                    sed "s/|LOCALUSERS|/$users_to_ring/g"       >> $WORKDIR/inext.TMP
-    fi
-
-    # If this provider is enabled for outgoing calls.
-    if [ "$make_outgoing_calls" = "yes" ] ; then
-
-        numprefix="`uci -q get ${MODULENAME}-calls.outgoing_calls.$name`"
-        # If no prefixes are specified, then we use "X" which matches any prefix.
-        [ -z "$numprefix" ]  && numprefix="X"
-        for p in $numprefix ; do
-            p=`escape_non_alpha $p`
-            sed "s/|NUMPREFIX|/$p/g" $TMPL_EXTOUTSIP |\
-                            sed "s/|NAME|/$name/g"              >> $WORKDIR/outext-$name.TMP
-        done
-
-        # Add this provider to the list of enabled outbound providers.
-        if [ -z "$outbound_providers" ] ; then
-            outbound_providers="$name"
-        else
-            outbound_providers="$outbound_providers $name"
-        fi
-
-        # Add this provider to the list of enabled sip outbound providers.
-        if [ -z "$sip_outbound_providers" ] ; then
-            sip_outbound_providers="$name"
-        else
-            sip_outbound_providers="$sip_outbound_providers $name"
-        fi
-    fi
-
-    rm -f $WORKDIR/sip_peer.tmp
-    rm -f $WORKDIR/sip_reg.tmp
-}
-
-# For all local users enabled for outbound calls, creates a context
-# containing the extensions for Google and SIP accounts this user is
-# allowed to use.
-pbx_create_user_contexts()
-{
-    local providers
-
-    for u in $localusers_can_dial ; do
-        u=`escape_non_alpha $u`
-        sed "s/|DEFAULTUSER|/$u/g" $TMPL_EXTUSERCNTXTHDR                >> $WORKDIR/userext.TMP
-        cat $WORKDIR/localext.TMP                                       >> $WORKDIR/userext.TMP
-        providers="`uci -q get ${MODULENAME}-calls.providers_user_can_use.$u`"
-        [ -z "$providers" ] && providers="$outbound_providers"
-
-        # For each provider, cat the contents of outext-$name.TMP into the user's outgoing calls extension
-        for p in $providers ; do
-            [ -f $WORKDIR/outext-$p.TMP ] && cat $WORKDIR/outext-$p.TMP >> $WORKDIR/userext.TMP
-        done
-        cat $TMPL_EXTUSERCNTXTFTR                                       >> $WORKDIR/userext.TMP
-    done
-}
-
-# Creates the blacklist context which hangs up on blacklisted numbers.
-pbx_add_blacklist()
-{
-    local blacklist1
-    local blacklist2
-
-    config_get blacklist1 blacklisting blacklist1
-    config_get blacklist2 blacklisting blacklist2
-
-    # We create the blacklist context no matter whether the blacklist
-    # actually contains entries or not, since the PBX will send calls
-    # to the context for a check against the list anyway.
-    cp $TMPL_EXTBLKLISTHDR                                   $WORKDIR/blacklist.TMP
-    for n in $blacklist1 $blacklist2 ; do
-        n=`escape_non_alpha $n`
-        sed "s/|BLACKLISTITEM|/$n/g" $TMPL_EXTBLKLIST >> $WORKDIR/blacklist.TMP 
-    done
-    cat $TMPL_EXTBLKLISTFTR                               >> $WORKDIR/blacklist.TMP
-}
-
-# Creates the callthrough context which allows specified numbers to get
-# into the PBX and dial out as the configured user.
-pbx_add_callthrough()
-{
-    local callthrough_number_list
-    local defaultuser
-    local pin
-    local enabled
-    local F
-
-    config_get callthrough_number_list $1 callthrough_number_list
-    config_get defaultuser $1 defaultuser
-    defaultuser=`escape_non_alpha $defaultuser`
-    config_get pin $1 pin
-    pin=`escape_non_alpha $pin`
-    config_get enabled $1 enabled
-
-    [ "$enabled" = "no" ]   && return
-    [ "$defaultuser" = "" ] && return
-
-    for callthrough_number in $callthrough_number_list ; do
-        sed "s/|NUMBER|/$callthrough_number/g" $TMPL_EXTCTHRUCHECK >> $WORKDIR/callthroughcheck.TMP
-
-	if [ -n "$pin" ] ; then F=$TMPL_EXTCTHRU ; else F=$TMPL_EXTCTHRUNOPIN ; fi
-        sed "s/|NUMBER|/$callthrough_number/g" $F             |\
-                    sed "s/|DEFAULTUSER|/$defaultuser/"       |\
-                        sed "s/|PIN|/$pin/"                        >> $WORKDIR/callthrough.TMP
-    done
-}
-
-
-# Creates the callback context which allows specified numbers to get
-# a callback into the PBX and dial out as the configured user.
-pbx_add_callback()
-{
-    local callback_number_list
-    local defaultuser
-    local pin
-    local enabled
-    local callback_provider
-    local callback_hangup_delay
-    local FB
-    local FT
-
-    config_get callback_number_list $1 callback_number_list
-    config_get defaultuser $1 defaultuser
-    defaultuser=`escape_non_alpha $defaultuser`
-    config_get pin $1 pin
-    pin=`escape_non_alpha $pin`
-    config_get enabled $1 enabled
-    config_get callback_provider $1 callback_provider
-    callback_provider=`sub_underscore_for_non_alpha $callback_provider`
-    config_get callback_hangup_delay $1 callback_hangup_delay
-
-    [ "$enabled" = "no" ]   && return
-    [ "$defaultuser" = "" ] && return
-
-    # If the provider is a SIP provider, set the file to use to $TMPL_EXTCBACKSIP
-    # otherwise, set it to $TMPL_EXTCBACKGTALK
-    if echo $sip_outbound_providers | grep -q $callback_provider 2>/dev/null
-    then
-        FB=$TMPL_EXTCBACKSIP
-    else
-        FB=$TMPL_EXTCBACKGTALK
-    fi
-
-    for callback_number in $callback_number_list ; do
-        sed "s/|NUMBER|/$callback_number/g" $TMPL_EXTCBACKCHECK    >> $WORKDIR/callbackcheck.TMP
-
-        sed "s/|NUMBER|/$callback_number/g" $FB            |\
-            sed "s/|CALLBACKPROVIDER|/$callback_provider/" |\
-                sed "s/|CALLBACKHUPDELAY|/$callback_hangup_delay/" >> $WORKDIR/callback.TMP
-
-        # Perhaps a bit confusingly, we create "callthrough" configuration for callback
-        # numbers, because we use the same DISA construct as for callthrough.
-        if [ -n "$pin" ] ; then FT=$TMPL_EXTCTHRU ; else FT=$TMPL_EXTCTHRUNOPIN ; fi
-        sed "s/|NUMBER|/$callback_number/g" $FT            |\
-            sed "s/|DEFAULTUSER|/$defaultuser/"            |\
-                sed "s/|PIN|/$pin/"                                >> $WORKDIR/callthrough.TMP
-    done
-}
-
-
-# Creates sip.conf from its template.
-pbx_cook_sip_template()
-{
-    local useragent
-    local externhost
-    local bindport
-
-    config_get useragent advanced useragent
-    useragent=`escape_non_alpha $useragent`
-    config_get externhost advanced externhost
-    config_get bindport advanced bindport
-
-    [ -z "$useragent" ] && useragent="$USERAGENT"
-
-    sed "s/|USERAGENT|/$useragent/g" $TMPL_SIP  > $WORKDIR/sip.conf
-
-    if [ -z "$externhost" ] ; then
-        sed -i "s/externhost=|EXTERNHOST|//g" $WORKDIR/sip.conf
-    else
-        sed -i "s/|EXTERNHOST|/$externhost/g" $WORKDIR/sip.conf
-    fi
-
-    if [ -z "$bindport" ] ; then
-        sed -i "s/bindport=|BINDPORT|//g"     $WORKDIR/sip.conf
-    else
-        sed -i "s/|BINDPORT|/$bindport/g"     $WORKDIR/sip.conf
-    fi
-
-
-}
-
-# Creates rtp.conf from its template.
-pbx_cook_rtp_template()
-{
-    local rtpstart
-    local rtpend
-
-    config_get rtpstart advanced rtpstart
-    config_get rtpend   advanced rtpend
-
-    sed "s/|RTPSTART|/$rtpstart/" $TMPL_RTP |\
-            sed "s/|RTPEND|/$rtpend/"               > $WORKDIR/rtp.conf
-}
-
-# Links any sound files found in $PBXSOUNDSDIR and $VMSOUNDSDIR
-# into $ASTSOUNDSDIR for use by Asterisk. Does not overwrite files.
-pbx_link_sounds()
-{
-    mkdir -p $ASTSOUNDSDIR
-
-    for dir in $PBXSOUNDSDIR $VMSOUNDSDIR ; do
-        if [ -d $dir ] ; then
-            for f in $dir/* ; do
-                ln -s $f $ASTSOUNDSDIR 2>/dev/null
-            done
-        fi
-    done
-}
-
-
-# Makes sure the ownership of specified directories is proper.
-pbx_fix_ownership()
-{
-    chown $ASTUSER:$ASTGROUP    $ASTDIRS
-    chown $ASTUSER:$ASTGROUP -R $ASTDIRSRECURSIVE
-}
-
-
-# Creates voicemail config if installed and enabled.
-pbx_configure_voicemail()
-{
-    local enabled
-    local global_timeout
-    local global_email_addresses
-
-    local smtp_tls
-    local smtp_server
-    local smtp_port
-    local smtp_auth
-    local smtp_user
-    local smtp_password
-
-    config_get enabled global_voicemail enabled
-
-    # First check if voicemail is enabled.
-    [ "$enabled" != "yes" ] && return
-
-    config_get global_timeout         global_voicemail global_timeout
-    #config_get global_email_addresses global_voicemail global_email_addresses
-    config_get smtp_auth              voicemail_smtp   smtp_auth
-    config_get smtp_tls               voicemail_smtp   smtp_tls
-    config_get smtp_server            voicemail_smtp   smtp_server
-    config_get smtp_port              voicemail_smtp   smtp_port
-    config_get smtp_user              voicemail_smtp   smtp_user
-    smtp_user=`escape_non_alpha $smtp_user`
-    config_get smtp_password          voicemail_smtp   smtp_password
-    smtp_password=`escape_non_alpha $smtp_password`
-
-    sed "s/|AUTH|/$smtp_auth/"   $TMPL_MSMTPDEFAULT |\
-             sed "s/|TLS|/$smtp_tls/"               >  $WORKDIR/pbx-msmtprc
-
-    sed "s/|HOST|/$smtp_server/" $TMPL_MSMTPACCOUNT |\
-             sed "s/|PORT|/$smtp_port/"             >> $WORKDIR/pbx-msmtprc
-
-    if [ "$smtp_auth" = "on" ] ; then
-        sed "s/|USER|/$smtp_user/"  $TMPL_MSMTPAUTH |\
-             sed "s/|PASSWORD|/$smtp_password/"     >> $WORKDIR/pbx-msmtprc
-    fi
-
-    cat $TMPL_MSMTPACCTDFLT                         >> $WORKDIR/pbx-msmtprc
-
-    [ ! -f /etc/pbx-msmtprc ] && cp $WORKDIR/pbx-msmtprc /etc/pbx-msmtprc
-    cmp -s  $WORKDIR/pbx-msmtprc /etc/pbx-msmtprc 1>/dev/null \
-                                || mv $WORKDIR/pbx-msmtprc /etc/pbx-msmtprc
-    chmod 600                                              /etc/pbx-msmtprc
-    chown nobody                                           /etc/pbx-msmtprc
-
-    # Copy over the extensions file which has voicemail enabled.
-    cp $TMPL_EXTVMENABLED                $WORKDIR/extensions_voicemail.conf
-
-    # Create the voicemail directory in /tmp
-    mkdir -p     /tmp/voicemail
-    chown nobody /tmp/voicemail
-
-    # Create the recordings directory
-    mkdir -p     /etc/pbx-voicemail/recordings
-    chown nobody /etc/pbx-voicemail/recordings
-
-    # Working around a bug in OpenWRT 12.09-rc1
-    # TODO: REMOVE AS SOON AS POSSIBLE
-    chmod ugo+w /tmp
-}
-
-
-start() {
-    mkdir -p $WORKDIR
-
-    # Create the users.
-    config_load ${MODULENAME}-users
-    config_foreach pbx_add_user local_user
-
-    # Create configuration for each google account.
-    config_unset
-    config_load ${MODULENAME}-google
-    config_foreach pbx_add_jabber gtalk_jabber
-
-    # Create configuration for each voip provider.
-    config_unset
-    config_load ${MODULENAME}-voip
-    config_foreach pbx_add_peer voip_provider
-
-    # Create the user contexts, callthroug/back, and phone blacklist.
-    config_unset
-    config_load ${MODULENAME}-calls
-    pbx_create_user_contexts
-    pbx_add_blacklist
-    config_foreach pbx_add_callthrough callthrough_numbers
-    config_foreach pbx_add_callback callback_numbers
-
-    # Prepare sip.conf using settings from the "advanced" section.
-    config_unset
-    config_load ${MODULENAME}-advanced
-    pbx_cook_sip_template
-    pbx_cook_rtp_template
-
-    # Prepare voicemail config.
-    config_unset
-    config_load ${MODULENAME}-voicemail
-    pbx_configure_voicemail
-
-    # Assemble the configuration, and copy changed files over.
-    config_unset
-    config_load ${MODULENAME}-advanced
-    pbx_assemble_and_copy_config
-
-    # Link sound files
-    pbx_link_sounds
-
-    # Enforce ownership of specified files and directories.
-    pbx_fix_ownership
-}
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE
deleted file mode 100644
index ac5439615b..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE
+++ /dev/null
@@ -1,17 +0,0 @@
-[directories]
-astetcdir => /etc/asterisk
-astmoddir => /usr/lib/asterisk/modules
-astvarlibdir => /usr/lib/asterisk
-astdbdir => /usr/lib/asterisk
-astkeydir => /usr/lib/asterisk
-astdatadir => /usr/lib/asterisk
-astagidir => /usr/lib/asterisk/agi-bin
-astspooldir => /var/spool/asterisk
-astrundir => /var/run/asterisk
-astlogdir => /var/log/asterisk
-
-[options]
-languageprefix = yes
-dumpcore = no
-runuser = nobody
-rungroup = nogroup
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/delayedcallback b/applications/luci-app-pbx/root/etc/pbx-asterisk/delayedcallback
deleted file mode 100755
index 903efe9ad9..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/delayedcallback
+++ /dev/null
@@ -1,18 +0,0 @@
-#!/bin/sh
-
-# Check if there are more than one instance of this command
-# with the same command line running at the same time for some
-# reason, then quit. We are checking for the same
-# commandline in order to permit two different callback
-# attempts simultaneously.
-
-if ! grep -q "$@" /dev/shm/delayedcallback.[0-9]* 2>/dev/null
-then
-  echo "$@" > /dev/shm/delayedcallback.$$
-  sleep 25
-  asterisk -r -x "$1 $2 \"$3\" $4 $5 $6"
-  rm /dev/shm/delayedcallback.$$
-#  echo "`date` $@": >> /dev/shm/delayedcallback.log
-#else
-#  echo "`date` ERROR: There appears to be a callback attempt in progress to: $@" >> /dev/shm/delayedcallback.err
-fi
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE
deleted file mode 100644
index c8966edd87..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE
+++ /dev/null
@@ -1,25 +0,0 @@
-[general]
-static = yes
-writeprotect = yes
-clearglobalvars = no
-
-[globals]
-RINGTIME => |RINGTIME|
-
-[default]
-
-[context-user-hangup-call-context]
-exten =>   s,1,Hangup()
-exten => _X.,1,Hangup()
-
-[context-catch-all]
-exten   => _[!-~].,1,Dial(SIP/${EXTEN},60,r)
-
-#include extensions_default.conf
-#include extensions_voicemail.conf
-#include extensions_incoming.conf
-#include extensions_incoming_gtalk.conf
-#include extensions_blacklist.conf
-#include extensions_callthrough.conf
-#include extensions_callback.conf
-#include extensions_user.conf
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist.conf.TEMPLATE
deleted file mode 100644
index 54ee989b0f..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist.conf.TEMPLATE
+++ /dev/null
@@ -1 +0,0 @@
-exten => s,n,Gotoif($[ "${CALLERID(NUM)}" = "|BLACKLISTITEM|" ]?context-user-hangup,s,1)
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist_footer.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist_footer.conf.TEMPLATE
deleted file mode 100644
index da964f2388..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist_footer.conf.TEMPLATE
+++ /dev/null
@@ -1,2 +0,0 @@
-exten => s,n,Goto(${SOURCECONTEXT},${SOURCEEXTEN},doneblacklist)
-
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist_header.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist_header.conf.TEMPLATE
deleted file mode 100644
index de0e984652..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist_header.conf.TEMPLATE
+++ /dev/null
@@ -1,3 +0,0 @@
-
-[blacklist-call-context]
-exten => s,1,Noop()
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check.conf.TEMPLATE
deleted file mode 100644
index 06b1a4b6b9..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check.conf.TEMPLATE
+++ /dev/null
@@ -1 +0,0 @@
-exten => s,n,Gotoif($[ "${CALLERID(NUM)}" =~ ".*|NUMBER|" ]?context-user-callback,|NUMBER|,1)
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check_footer.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check_footer.conf.TEMPLATE
deleted file mode 100644
index 282fe9e8ff..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check_footer.conf.TEMPLATE
+++ /dev/null
@@ -1,2 +0,0 @@
-exten => s,n,Goto(${SOURCECONTEXT},${SOURCEEXTEN},donecallback)
-
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check_header.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check_header.conf.TEMPLATE
deleted file mode 100644
index be289c4d33..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check_header.conf.TEMPLATE
+++ /dev/null
@@ -1,3 +0,0 @@
-
-[callback-check-call-context]
-exten => s,1,Noop()
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_gtalk.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_gtalk.conf.TEMPLATE
deleted file mode 100644
index 43eec788f3..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_gtalk.conf.TEMPLATE
+++ /dev/null
@@ -1,4 +0,0 @@
-exten => |NUMBER|,1,System(/etc/pbx-asterisk/delayedcallback "channel originate Gtalk/gtalk-|CALLBACKPROVIDER|/|NUMBER|@voice.google.com extension |NUMBER|@disa-call-context" &)
-exten => |NUMBER|,n,Wait(|CALLBACKHUPDELAY|)
-exten => |NUMBER|,n,Hangup()
-
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_header.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_header.conf.TEMPLATE
deleted file mode 100644
index 0b8fb4c23f..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_header.conf.TEMPLATE
+++ /dev/null
@@ -1 +0,0 @@
-[context-user-callback]
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_sip.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_sip.conf.TEMPLATE
deleted file mode 100644
index 300e9fa0e8..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_sip.conf.TEMPLATE
+++ /dev/null
@@ -1,4 +0,0 @@
-exten => |NUMBER|,1,System(/etc/pbx-asterisk/delayedcallback "channel originate SIP/|NUMBER|@peer-|CALLBACKPROVIDER| extension |NUMBER|@disa-call-context" &)
-exten => |NUMBER|,n,Wait(|CALLBACKHUPDELAY|)
-exten => |NUMBER|,n,Hangup()
-
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_default.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_default.conf.TEMPLATE
deleted file mode 100644
index 35836e290a..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_default.conf.TEMPLATE
+++ /dev/null
@@ -1,11 +0,0 @@
-[default-incoming-call-context]
-exten => s,1,NoOp(${CALLERID})
-exten => s,n,Set(SOURCECONTEXT=default-incoming-call-context)
-exten => s,n,Set(SOURCEEXTEN=s)
-exten => s,n,Goto(blacklist-call-context,s,1)
-exten => s,n(doneblacklist),NoOp()
-exten => s,n,Goto(callback-check-call-context,s,1)
-exten => s,n(donecallback),NoOp()
-exten => s,n,Goto(disa-check-call-context,s,1)
-exten => s,n(donedisacheck),Dial(|LOCALUSERS|,${RINGTIME},r)
-exten => s,n,Goto(context-voicemail,s,1)
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_default_user.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_default_user.conf.TEMPLATE
deleted file mode 100644
index 1910ff4d96..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_default_user.conf.TEMPLATE
+++ /dev/null
@@ -1 +0,0 @@
-exten => |DEFAULTUSER|,1,Goto(default-incoming-call-context,s,1)
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check.conf.TEMPLATE
deleted file mode 100644
index ba2379b738..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check.conf.TEMPLATE
+++ /dev/null
@@ -1 +0,0 @@
-exten => s,n,Gotoif($[ "${CALLERID(NUM)}" =~ ".*|NUMBER|" ]?disa-call-context,|NUMBER|,1)
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check_footer.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check_footer.conf.TEMPLATE
deleted file mode 100644
index 74056fa01d..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check_footer.conf.TEMPLATE
+++ /dev/null
@@ -1 +0,0 @@
-exten => s,n,Goto(${SOURCECONTEXT},${SOURCEEXTEN},donedisacheck)
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check_header.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check_header.conf.TEMPLATE
deleted file mode 100644
index e0d67b8025..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check_header.conf.TEMPLATE
+++ /dev/null
@@ -1,2 +0,0 @@
-[disa-check-call-context]
-exten => s,1,Noop()
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-nopin.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-nopin.conf.TEMPLATE
deleted file mode 100644
index 74e48de8c1..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-nopin.conf.TEMPLATE
+++ /dev/null
@@ -1,5 +0,0 @@
-exten => |NUMBER|,1,Noop()
-exten => |NUMBER|,n,Set(TIMEOUT(digit)=15)
-exten => |NUMBER|,n,Set(TIMEOUT(response)=40)
-exten => |NUMBER|,n,DISA(no-password,context-user-|DEFAULTUSER|)
-
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa.conf.TEMPLATE
deleted file mode 100644
index 3dd8fa35c9..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa.conf.TEMPLATE
+++ /dev/null
@@ -1,6 +0,0 @@
-exten => |NUMBER|,1,Noop()
-exten => |NUMBER|,n,Set(TIMEOUT(digit)=7)
-exten => |NUMBER|,n,Set(TIMEOUT(response)=21)
-exten => |NUMBER|,n,Authenticate(|PIN|)
-exten => |NUMBER|,n,DISA(no-password,context-user-|DEFAULTUSER|)
-
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa_header.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa_header.conf.TEMPLATE
deleted file mode 100644
index a742271146..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa_header.conf.TEMPLATE
+++ /dev/null
@@ -1 +0,0 @@
-[disa-call-context]
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk.conf.TEMPLATE
deleted file mode 100644
index 3f9cf4c7d9..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk.conf.TEMPLATE
+++ /dev/null
@@ -1,15 +0,0 @@
-exten => |USERNAME|,1,NoOp(${CALLERID})
-same => n,Set(CALLERID(name)=${CUT(CALLERID(name),@,1)})
-same => n,GotoIf($["${CALLERID(name):0:2}" != "+1"]?notrim)
-same => n,Set(CALLERID(name)=${CALLERID(name):2})
-same => n(notrim),Set(CALLERID(number)=${CALLERID(name)})
-same => n,Set(SOURCECONTEXT=context-incoming-gtalk)
-same => n,Set(SOURCEEXTEN=|USERNAME|)
-same => n,Goto(blacklist-call-context,s,1)
-same => n(doneblacklist),NoOp()
-same => n,Goto(callback-check-call-context,s,1)
-same => n(donecallback),NoOp()
-same => n,Goto(disa-check-call-context,s,1)
-same => n(donedisacheck),Dial(|LOCALUSERS|,${RINGTIME},D(:w11111111))
-same => n,Goto(context-voicemail,s,1)
-
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk_header.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk_header.conf.TEMPLATE
deleted file mode 100644
index f6e44a5bf0..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk_header.conf.TEMPLATE
+++ /dev/null
@@ -1 +0,0 @@
-[context-incoming-gtalk]
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE
deleted file mode 100644
index b2c3716bf4..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE
+++ /dev/null
@@ -1,12 +0,0 @@
-
-[context-incoming-|NAME|]
-exten => s,1,NoOp(${CALLERID})
-exten => s,n,Set(SOURCECONTEXT=context-incoming-|NAME|)
-exten => s,n,Set(SOURCEEXTEN=s)
-exten => s,n,Goto(blacklist-call-context,s,1)
-exten => s,n(doneblacklist),NoOp()
-exten => s,n,Goto(callback-check-call-context,s,1)
-exten => s,n(donecallback),NoOp()
-exten => s,n,Goto(disa-check-call-context,s,1)
-exten => s,n(donedisacheck),Dial(|LOCALUSERS|,${RINGTIME},r)
-exten => s,n,Goto(context-voicemail,s,1)
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_dial_local_user.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_dial_local_user.conf.TEMPLATE
deleted file mode 100644
index 45e8758846..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_dial_local_user.conf.TEMPLATE
+++ /dev/null
@@ -1 +0,0 @@
-exten =>     |DEFAULTUSER|,1,Dial(SIP/|DEFAULTUSER|,${RINGTIME},r)
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_gtalk.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_gtalk.conf.TEMPLATE
deleted file mode 100644
index 259c2ceaa1..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_gtalk.conf.TEMPLATE
+++ /dev/null
@@ -1,9 +0,0 @@
-exten =>      _|NUMPREFIX|,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN}@voice.google.com,60)
-exten =>     _+|NUMPREFIX|,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:1}@voice.google.com,60)
-exten =>     _|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN}@voice.google.com,60)
-exten =>    _+|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:1}@voice.google.com,60)
-exten =>   _00|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:2}@voice.google.com,60)
-exten =>  _011|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:3}@voice.google.com,60)
-exten =>  _010|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:3}@voice.google.com,60)
-exten => _0011|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:4}@voice.google.com,60)
-
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_gtalk.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_gtalk.conf.TEMPLATE
deleted file mode 100644
index 1fa7713e23..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_gtalk.conf.TEMPLATE
+++ /dev/null
@@ -1,2 +0,0 @@
-exten => |PATTERN|,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN|SYMBOLSTOREMOVE|}@voice.google.com,60)
-
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_sip.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_sip.conf.TEMPLATE
deleted file mode 100644
index 178b6deaa6..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_sip.conf.TEMPLATE
+++ /dev/null
@@ -1 +0,0 @@
-exten => |PATTERN|,1,Dial(SIP/${EXTEN|SYMBOLSTOREMOVE|}@peer-|NAME|,60,r)
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_sip.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_sip.conf.TEMPLATE
deleted file mode 100644
index 9b1d9addc9..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_sip.conf.TEMPLATE
+++ /dev/null
@@ -1,8 +0,0 @@
-exten =>      _|NUMPREFIX|,1,Dial(SIP/${EXTEN}@peer-|NAME|,60,r)
-exten =>     _+|NUMPREFIX|,1,Dial(SIP/${EXTEN}@peer-|NAME|,60,r)
-exten =>     _|NUMPREFIX|.,1,Dial(SIP/${EXTEN}@peer-|NAME|,60,r)
-exten =>    _+|NUMPREFIX|.,1,Dial(SIP/${EXTEN:1}@peer-|NAME|,60,r)
-exten =>   _00|NUMPREFIX|.,1,Dial(SIP/${EXTEN:2}@peer-|NAME|,60,r)
-exten =>  _011|NUMPREFIX|.,1,Dial(SIP/${EXTEN:3}@peer-|NAME|,60,r)
-exten =>  _010|NUMPREFIX|.,1,Dial(SIP/${EXTEN:3}@peer-|NAME|,60,r)
-exten => _0011|NUMPREFIX|.,1,Dial(SIP/${EXTEN:4}@peer-|NAME|,60,r)
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_user_context_footer.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_user_context_footer.conf.TEMPLATE
deleted file mode 100644
index a2ba28c055..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_user_context_footer.conf.TEMPLATE
+++ /dev/null
@@ -1,2 +0,0 @@
-include => context-voicemail-record-greeting
-include => context-catch-all
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_user_context_header.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_user_context_header.conf.TEMPLATE
deleted file mode 100644
index 5931eaf28b..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_user_context_header.conf.TEMPLATE
+++ /dev/null
@@ -1,3 +0,0 @@
-
-[context-user-|DEFAULTUSER|]
-
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_voicemail_disabled.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_voicemail_disabled.conf.TEMPLATE
deleted file mode 100644
index be23c294df..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_voicemail_disabled.conf.TEMPLATE
+++ /dev/null
@@ -1,4 +0,0 @@
-[context-voicemail-record-greeting]
-
-[context-voicemail]
-exten =>     s,1,Hangup()
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_voicemail_enabled.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_voicemail_enabled.conf.TEMPLATE
deleted file mode 100644
index 4edd9cb426..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_voicemail_enabled.conf.TEMPLATE
+++ /dev/null
@@ -1,27 +0,0 @@
-[context-voicemail-record-greeting]
-exten =>     *789,1,Wait(1)
-exten =>     *789,n,Playback(/etc/pbx-voicemail/recordings/greeting)
-exten =>     *789,n,Wait(1)
-exten =>     *789,n,Playback(beep)
-exten =>     *789,n,Playback(beep)
-exten =>     *789,n,WaitExten(30)
-
-exten =>     t,1,Playback(vm-goodbye)
-exten =>     t,n,Wait(2)
-exten =>     t,n,Hangup()
-
-exten =>     *,1,Playback(beep)
-exten =>     *,n,Playback(beep)
-exten =>     *,n,Record(/tmp/voicemail/greeting:gsm,20,120,k)
-exten =>     *,n,Wait(1)
-exten =>     *,n,Playback(/tmp/voicemail/greeting)
-
-exten =>     h,1,System(/etc/pbx-voicemail/pbx-move-greeting &)
-
-[context-voicemail]
-exten =>     s,1,Wait(2)
-exten =>     s,2,Playback(/etc/pbx-voicemail/recordings/greeting)
-exten =>     s,3,Wait(2)
-exten =>     s,n,Record(/tmp/voicemail/voicemail%d:WAV,20,180,k)
-
-exten =>     h,1,System(/etc/pbx-voicemail/pbx-send-voicemail '${RECORDED_FILE}.WAV' '${CALLERID(all)}' &)
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/gtalk.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/gtalk.conf.TEMPLATE
deleted file mode 100644
index 4f07a71660..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/gtalk.conf.TEMPLATE
+++ /dev/null
@@ -1,10 +0,0 @@
-[general]
-context=context-incoming-gtalk
-allowguest=yes
-allowguests=yes
-bindaddr=0.0.0.0
-
-[guest]
-disallow=all
-allow=ulaw
-context=context-incoming-gtalk
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/indications.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/indications.conf.TEMPLATE
deleted file mode 100644
index d7088db7c4..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/indications.conf.TEMPLATE
+++ /dev/null
@@ -1,733 +0,0 @@
-; indications.conf
-; Configuration file for location specific tone indications
-; used by the pbx_indications module.
-;
-; NOTE:
-;    When adding countries to this file, please keep them in alphabetical
-;    order according to the 2-character country codes!
-;
-; The [general] category is for certain global variables.
-; All other categories are interpreted as location specific indications
-;
-;
-[general]
-country=us		; default location
-
-
-; [example]
-; description = string
-;      The full name of your country, in English.
-; alias = iso[,iso]*
-;      List of other countries 2-letter iso codes, which have the same
-;      tone indications.
-; ringcadence = num[,num]*
-;      List of durations the physical bell rings.
-; dial = tonelist
-;      Set of tones to be played when one picks up the hook.
-; busy = tonelist
-;      Set of tones played when the receiving end is busy.
-; congestion = tonelist
-;      Set of tones played when there is some congestion (on the network?)
-; callwaiting = tonelist
-;      Set of tones played when there is a call waiting in the background.
-; dialrecall = tonelist
-;      Not well defined; many phone systems play a recall dial tone after hook
-;      flash.
-; record = tonelist
-;      Set of tones played when call recording is in progress.
-; info = tonelist
-;      Set of tones played with special information messages (e.g., "number is
-;      out of service")
-; 'name' = tonelist
-;      Every other variable will be available as a shortcut for the "PlayList" command
-;      but will not be used automatically by Asterisk.
-;
-;
-; The tonelist itself is defined by a comma-separated sequence of elements.
-; Each element consist of a frequency (f) with an optional duration (in ms)
-; attached to it (f/duration). The frequency component may be a mixture of two
-; frequencies (f1+f2) or a frequency modulated by another frequency (f1*f2).
-; The implicit modulation depth is fixed at 90%, though.
-; If the list element starts with a !, that element is NOT repeated,
-; therefore, only if all elements start with !, the tonelist is time-limited,
-; all others will repeat indefinitely.
-;
-; concisely:
-;   element = [!]freq[+|*freq2][/duration]
-;   tonelist = element[,element]*
-;
-; Please note that SPACES ARE NOT ALLOWED in tone lists!
-;
-
-[at]
-description = Austria
-ringcadence = 1000,5000
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-dial = 420
-busy = 420/400,0/400
-ring = 420/1000,0/5000
-congestion = 420/200,0/200
-callwaiting = 420/40,0/1960
-dialrecall = 420
-; RECORDTONE - not specified
-record = 1400/80,0/14920
-info = 950/330,1450/330,1850/330,0/1000
-stutter = 380+420
-
-[au]
-description = Australia
-; Reference http://www.acif.org.au/__data/page/3303/S002_2001.pdf
-; Normal Ring
-ringcadence = 400,200,400,2000
-; Distinctive Ring 1 - Forwarded Calls
-; 400,400,200,200,400,1400
-; Distinctive Ring 2 - Selective Ring 2 + Operator + Recall
-; 400,400,200,2000
-; Distinctive Ring 3 - Multiple Subscriber Number 1
-; 200,200,400,2200
-; Distinctive Ring 4 - Selective Ring 1 + Centrex
-; 400,2600
-; Distinctive Ring 5 - Selective Ring 3
-; 400,400,200,400,200,1400
-; Distinctive Ring 6 - Multiple Subscriber Number 2
-; 200,400,200,200,400,1600
-; Distinctive Ring 7 - Multiple Subscriber Number 3 + Data Privacy
-; 200,400,200,400,200,1600
-; Tones
-dial = 413+438
-busy = 425/375,0/375
-ring = 413+438/400,0/200,413+438/400,0/2000
-; XXX Congestion: Should reduce by 10 db every other cadence XXX
-congestion = 425/375,0/375,420/375,0/375
-callwaiting = 425/200,0/200,425/200,0/4400
-dialrecall = 413+438
-; Record tone used for Call Intrusion/Recording or Conference
-record = !425/1000,!0/15000,425/360,0/15000
-info = 425/2500,0/500
-; Other Australian Tones
-; The STD "pips" indicate the call is not an untimed local call
-std = !525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100
-; Facility confirmation tone (eg. Call Forward Activated)
-facility = 425
-; Message Waiting "stutter" dialtone
-stutter = 413+438/100,0/40
-; Ringtone for calls to Telstra mobiles
-ringmobile = 400+450/400,0/200,400+450/400,0/2000
-
-[bg]
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-description = Bulgaria
-ringcadence = 1000,4000
-;
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/4000
-congestion = 425/250,0/250
-callwaiting = 425/150,0/150,425/150,0/4000
-dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-record = 1400/425,0/15000
-info = 950/330,1400/330,1800/330,0/1000
-stutter = 425/1500,0/100
-
-[br]
-description = Brazil
-ringcadence = 1000,4000
-dial = 425
-busy = 425/250,0/250
-ring = 425/1000,0/4000
-congestion = 425/250,0/250,425/750,0/250
-callwaiting = 425/50,0/1000
-; Dialrecall not used in Brazil standard (using UK standard)
-dialrecall = 350+440
-; Record tone is not used in Brazil, use busy tone
-record = 425/250,0/250
-; Info not used in Brazil standard (using UK standard)
-info = 950/330,1400/330,1800/330
-stutter = 350+440
-
-[be]
-description = Belgium
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,3000
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/3000
-congestion = 425/167,0/167
-callwaiting = 1400/175,0/175,1400/175,0/3500
-; DIALRECALL - not specified
-dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440"
-; RECORDTONE - not specified
-record = 1400/500,0/15000
-info = 900/330,1400/330,1800/330,0/1000
-stutter = 425/1000,0/250
-
-[ch]
-description = Switzerland
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,4000
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/4000
-congestion = 425/200,0/200
-callwaiting = 425/200,0/200,425/200,0/4000
-; DIALRECALL - not specified
-dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-; RECORDTONE - not specified
-record = 1400/80,0/15000
-info = 950/330,1400/330,1800/330,0/1000
-stutter = 425+340/1100,0/1100
-
-[cl]
-description = Chile
-; According to specs from Telefonica CTC Chile
-ringcadence = 1000,3000
-dial = 400
-busy = 400/500,0/500
-ring = 400/1000,0/3000
-congestion = 400/200,0/200
-callwaiting = 400/250,0/8750
-dialrecall = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
-record = 1400/500,0/15000
-info = 950/333,1400/333,1800/333,0/1000
-stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
-
-[cn]
-description = China
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,4000
-dial = 450
-busy = 450/350,0/350
-ring = 450/1000,0/4000
-congestion = 450/700,0/700
-callwaiting = 450/400,0/4000
-dialrecall = 450
-record = 950/400,0/10000
-info = 450/100,0/100,450/100,0/100,450/100,0/100,450/400,0/400
-; STUTTER - not specified
-stutter = 450+425
-
-[cz]
-description = Czech Republic
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,4000
-dial = 425/330,0/330,425/660,0/660
-busy = 425/330,0/330
-ring = 425/1000,0/4000
-congestion = 425/165,0/165
-callwaiting = 425/330,0/9000
-; DIALRECALL - not specified
-dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425/330,0/330,425/660,0/660
-; RECORDTONE - not specified
-record = 1400/500,0/14000
-info = 950/330,0/30,1400/330,0/30,1800/330,0/1000
-; STUTTER - not specified
-stutter = 425/450,0/50
-
-[de]
-description = Germany
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,4000
-dial = 425
-busy = 425/480,0/480
-ring = 425/1000,0/4000
-congestion = 425/240,0/240
-callwaiting = !425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,0
-; DIALRECALL - not specified
-dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-; RECORDTONE - not specified
-record = 1400/80,0/15000
-info = 950/330,1400/330,1800/330,0/1000
-stutter = 425+400
-
-[dk]
-description = Denmark
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,4000
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/4000
-congestion = 425/200,0/200
-callwaiting = !425/200,!0/600,!425/200,!0/3000,!425/200,!0/200,!425/200,0
-; DIALRECALL - not specified
-dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-; RECORDTONE - not specified
-record = 1400/80,0/15000
-info = 950/330,1400/330,1800/330,0/1000
-; STUTTER - not specified
-stutter = 425/450,0/50
-
-[ee]
-description = Estonia
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,4000
-dial = 425
-busy = 425/300,0/300
-ring = 425/1000,0/4000
-congestion = 425/200,0/200
-; CALLWAIT not in accordance to ITU
-callwaiting = 950/650,0/325,950/325,0/30,1400/1300,0/2600
-; DIALRECALL - not specified
-dialrecall = 425/650,0/25
-; RECORDTONE - not specified
-record = 1400/500,0/15000
-; INFO not in accordance to ITU
-info = 950/650,0/325,950/325,0/30,1400/1300,0/2600
-; STUTTER not specified
-stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-
-[es]
-description = Spain
-ringcadence = 1500,3000
-dial = 425
-busy = 425/200,0/200
-ring = 425/1500,0/3000
-congestion = 425/200,0/200,425/200,0/200,425/200,0/600
-callwaiting = 425/175,0/175,425/175,0/3500
-dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425
-record = 1400/500,0/15000
-info = 950/330,0/1000
-dialout = 500
-
-
-[fi]
-description = Finland
-ringcadence = 1000,4000
-dial = 425
-busy = 425/300,0/300
-ring = 425/1000,0/4000
-congestion = 425/200,0/200
-callwaiting = 425/150,0/150,425/150,0/8000
-dialrecall = 425/650,0/25
-record = 1400/500,0/15000
-info = 950/650,0/325,950/325,0/30,1400/1300,0/2600
-stutter = 425/650,0/25
-
-[fr]
-description = France
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1500,3500
-; Dialtone can also be 440+330
-dial = 440
-busy = 440/500,0/500
-ring = 440/1500,0/3500
-; CONGESTION - not specified
-congestion = 440/250,0/250
-callwait = 440/300,0/10000
-; DIALRECALL - not specified
-dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-; RECORDTONE - not specified
-record = 1400/500,0/15000
-info = !950/330,!1400/330,!1800/330
-stutter = !440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,440
-
-[gr]
-description = Greece
-ringcadence = 1000,4000
-dial = 425/200,0/300,425/700,0/800
-busy = 425/300,0/300
-ring = 425/1000,0/4000
-congestion = 425/200,0/200
-callwaiting = 425/150,0/150,425/150,0/8000
-dialrecall = 425/650,0/25
-record = 1400/400,0/15000
-info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
-stutter = 425/650,0/25
-
-[hu]
-description = Hungary
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1250,3750
-dial = 425
-busy = 425/300,0/300
-ring = 425/1250,0/3750
-congestion = 425/300,0/300
-callwaiting = 425/40,0/1960
-dialrecall = 425+450
-; RECORDTONE - not specified
-record = 1400/400,0/15000
-info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
-stutter = 350+375+400
-
-[il]
-description = Israel
-ringcadence = 1000,3000
-dial = 414
-busy = 414/500,0/500
-ring = 414/1000,0/3000
-congestion = 414/250,0/250
-callwaiting = 414/100,0/100,414/100,0/100,414/600,0/3000
-dialrecall = !414/100,!0/100,!414/100,!0/100,!414/100,!0/100,414
-record = 1400/500,0/15000
-info = 1000/330,1400/330,1800/330,0/1000
-stutter = !414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,414
-
-
-[in]
-description = India
-ringcadence = 400,200,400,2000
-dial = 400*25
-busy = 400/750,0/750
-ring = 400*25/400,0/200,400*25/400,0/2000
-congestion = 400/250,0/250
-callwaiting = 400/200,0/100,400/200,0/7500
-dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-record = 1400/500,0/15000
-info = !950/330,!1400/330,!1800/330,0/1000
-stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-
-[it]
-description = Italy
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,4000
-dial = 425/200,0/200,425/600,0/1000
-busy = 425/500,0/500
-ring = 425/1000,0/4000
-congestion = 425/200,0/200
-callwaiting = 425/400,0/100,425/250,0/100,425/150,0/14000
-dialrecall = 470/400,425/400
-record = 1400/400,0/15000
-info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
-stutter = 470/400,425/400
-
-[lt]
-description = Lithuania
-ringcadence = 1000,4000
-dial = 425
-busy = 425/350,0/350
-ring = 425/1000,0/4000
-congestion = 425/200,0/200
-callwaiting = 425/150,0/150,425/150,0/4000
-; DIALRECALL - not specified
-dialrecall = 425/500,0/50
-; RECORDTONE - not specified
-record = 1400/500,0/15000
-info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
-; STUTTER - not specified
-stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-
-[jp]
-description = Japan
-ringcadence = 1000,2000
-dial = 400
-busy = 400/500,0/500
-ring = 400+15/1000,0/2000
-congestion = 400/500,0/500
-callwaiting = 400+16/500,0/8000
-dialrecall = !400/200,!0/200,!400/200,!0/200,!400/200,!0/200,400
-record = 1400/500,0/15000
-info = !950/330,!1400/330,!1800/330,0
-stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
-
-[mx]
-description = Mexico
-ringcadence = 2000,4000
-dial = 425
-busy = 425/250,0/250
-ring = 425/1000,0/4000
-congestion = 425/250,0/250
-callwaiting = 425/200,0/600,425/200,0/10000
-dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-record = 1400/500,0/15000
-info = 950/330,0/30,1400/330,0/30,1800/330,0/1000
-stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-
-[my]
-description = Malaysia
-ringcadence = 2000,4000
-dial = 425
-busy = 425/500,0/500
-ring = 425/400,0/200
-congestion = 425/500,0/500
-
-[nl]
-description = Netherlands
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-ringcadence = 1000,4000
-; Most of these 425's can also be 450's
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/4000
-congestion = 425/250,0/250
-callwaiting = 425/500,0/9500
-; DIALRECALL - not specified
-dialrecall = 425/500,0/50
-; RECORDTONE - not specified
-record = 1400/500,0/15000
-info = 950/330,1400/330,1800/330,0/1000
-stutter = 425/500,0/50
-
-[no]
-description = Norway
-ringcadence = 1000,4000
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/4000
-congestion = 425/200,0/200
-callwaiting = 425/200,0/600,425/200,0/10000
-dialrecall = 470/400,425/400
-record = 1400/400,0/15000
-info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
-stutter = 470/400,425/400
-
-[nz]
-description = New Zealand
-;NOTE - the ITU has different tonesets for NZ, but according to some residents there,
-;      this is, indeed, the correct way to do it.
-ringcadence = 400,200,400,2000
-dial = 400
-busy = 400/250,0/250
-ring = 400+450/400,0/200,400+450/400,0/2000
-congestion = 400/375,0/375
-callwaiting = !400/200,!0/3000,!400/200,!0/3000,!400/200,!0/3000,!400/200
-dialrecall = !400/100!0/100,!400/100,!0/100,!400/100,!0/100,400
-record = 1400/425,0/15000
-info = 400/750,0/100,400/750,0/100,400/750,0/100,400/750,0/400
-stutter = !400/100!0/100,!400/100,!0/100,!400/100,!0/100,!400/100!0/100,!400/100,!0/100,!400/100,!0/100,400
-unobtainable = 400/75,0/100,400/75,0/100,400/75,0/100,400/75,0/400
-
-[ph]
-
-; reference http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-
-description = Philippines
-ringcadence = 1000,4000
-dial = 425
-busy = 480+620/500,0/500
-ring = 425+480/1000,0/4000
-congestion = 480+620/250,0/250
-callwaiting = 440/300,0/10000
-; DIALRECALL - not specified
-dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-; RECORDTONE - not specified
-record = 1400/500,0/15000
-; INFO - not specified
-info = !950/330,!1400/330,!1800/330,0
-; STUTTER - not specified
-stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-
-
-[pl]
-description = Poland
-ringcadence = 1000,4000
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/4000
-congestion = 425/500,0/500
-callwaiting = 425/150,0/150,425/150,0/4000
-; DIALRECALL - not specified
-dialrecall = 425/500,0/50
-; RECORDTONE - not specified
-record = 1400/500,0/15000
-; 950/1400/1800 3x0.33 on 1.0 off  repeated 3 times 
-info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000
-; STUTTER - not specified
-stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-
-[pt]
-description = Portugal
-ringcadence = 1000,5000
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/5000
-congestion = 425/200,0/200
-callwaiting = 440/300,0/10000
-dialrecall = 425/1000,0/200
-record = 1400/500,0/15000
-info = 950/330,1400/330,1800/330,0/1000
-stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-
-[ru]
-; References:
-;	http://www.minsvyaz.ru/site.shtml?id=1806
-;	http://www.aboutphone.info/lib/gost/45-223-2001.html
-description = Russian Federation / ex Soviet Union
-ringcadence = 1000,4000
-dial = 425
-busy = 425/350,0/350
-ring = 425/1000,0/4000
-congestion = 425/175,0/175
-callwaiting = 425/200,0/5000
-record = 1400/400,0/15000
-info = 950/330,1400/330,1800/330,0/1000
-dialrecall = 425/400,0/40
-stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-
-[se]
-description = Sweden
-ringcadence = 1000,5000
-dial = 425
-busy = 425/250,0/250
-ring = 425/1000,0/5000
-congestion = 425/250,0/750
-callwaiting = 425/200,0/500,425/200,0/9100
-dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-record = 1400/500,0/15000
-info = !950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,0
-stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
-; stutter = 425/320,0/20        ; Real swedish standard, not used for now
-
-[sg]
-description = Singapore
-; Singapore
-; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf 
-; Frequency specs are:   425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz
-ringcadence = 400,200,400,2000
-dial        = 425
-ring        = 425*24/400,0/200,425*24/400,0/2000     ; modulation should be 100%, not 90%
-busy        = 425/750,0/750
-congestion  = 425/250,0/250
-callwaiting = 425*24/300,0/200,425*24/300,0/3200
-stutter     = !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425
-info        = 950/330,1400/330,1800/330,0/1000       ; not currently in use acc. to reference
-dialrecall  = 425*24/500,0/500,425/500,0/2500        ; unspecified in IDA reference, use repeating Holding Tone A,B
-record      = 1400/500,0/15000                       ; unspecified in IDA reference, use 0.5s tone every 15s
-; additionally defined in reference
-nutone      = 425/2500,0/500
-intrusion   = 425/250,0/2000
-warning     = 425/624,0/4376                         ; end of period tone, warning
-acceptance  = 425/125,0/125
-holdinga    = !425*24/500,!0/500                     ; followed by holdingb
-holdingb    = !425/500,!0/2500
-
-[th]
-description = Thailand
-ringcadence = 1000,4000
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-dial = 400*50
-busy = 400/500,0/500
-ring = 420/1000,0/5000
-congestion = 400/300,0/300
-callwaiting = 1000/400,10000/400,1000/400
-; DIALRECALL - not specified - use special dial tone instead.
-dialrecall = 400*50/400,0/100,400*50/400,0/100
-; RECORDTONE - not specified
-record = 1400/500,0/15000
-; INFO - specified as an announcement - use special information tones instead
-info = 950/330,1400/330,1800/330
-; STUTTER - not specified
-stutter = !400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,400
-
-[uk]
-description = United Kingdom
-ringcadence = 400,200,400,2000
-; These are the official tones taken from BT SIN350. The actual tones
-; used by BT include some volume differences so sound slightly different
-; from Asterisk-generated ones.
-dial = 350+440
-; Special dial is the intermittent dial tone heard when, for example,
-; you have a divert active on the line
-specialdial = 350+440/750,440/750
-; Busy is also called "Engaged"
-busy = 400/375,0/375
-; "Congestion" is the Beep-bip engaged tone
-congestion = 400/400,0/350,400/225,0/525
-; "Special Congestion" is not used by BT very often if at all
-specialcongestion = 400/200,1004/300
-unobtainable = 400
-ring = 400+450/400,0/200,400+450/400,0/2000
-callwaiting = 400/100,0/4000
-; BT seem to use "Special Call Waiting" rather than just "Call Waiting" tones
-specialcallwaiting = 400/250,0/250,400/250,0/250,400/250,0/5000
-; "Pips" used by BT on payphones. (Sounds wrong, but this is what BT claim it
-; is and I've not used a payphone for years)
-creditexpired = 400/125,0/125
-; These two are used to confirm/reject service requests on exchanges that
-; don't do voice announcements.
-confirm = 1400
-switching = 400/200,0/400,400/2000,0/400
-; This is the three rising tones Doo-dah-dee "Special Information Tone",
-; usually followed by the BT woman saying an appropriate message.
-info = 950/330,0/15,1400/330,0/15,1800/330,0/1000
-; Not listed in SIN350
-record = 1400/500,0/60000
-stutter = 350+440/750,440/750
-
-[us]
-description = United States / North America
-ringcadence = 2000,4000
-dial = 350+440
-busy = 480+620/500,0/500
-ring = 440+480/2000,0/4000
-congestion = 480+620/250,0/250
-callwaiting = 440/300,0/10000
-dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-record = 1400/500,0/15000
-info = !950/330,!1400/330,!1800/330,0
-stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-
-[us-old]
-description = United States Circa 1950/ North America
-ringcadence = 2000,4000
-dial = 600*120
-busy = 500*100/500,0/500
-ring = 420*40/2000,0/4000
-congestion = 500*100/250,0/250
-callwaiting = 440/300,0/10000
-dialrecall = !600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,600*120
-record = 1400/500,0/15000
-info = !950/330,!1400/330,!1800/330,0
-stutter = !600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,600*120
-
-[tw]
-description = Taiwan
-; http://nemesis.lonestar.org/reference/telecom/signaling/dialtone.html
-; http://nemesis.lonestar.org/reference/telecom/signaling/busy.html
-; http://www.iproducts.com.tw/ee/kylink/06ky-1000a.htm
-; http://www.pbx-manufacturer.com/ky120dx.htm
-; http://www.nettwerked.net/tones.txt
-; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/taiw_sup/taiw2.htm
-;
-; busy tone 480+620Hz 0.5 sec. on ,0.5 sec. off
-; reorder tone 480+620Hz 0.25 sec. on,0.25 sec. off
-; ringing tone 440+480Hz 1 sec. on ,2 sec. off
-;
-ringcadence = 1000,4000
-dial = 350+440
-busy = 480+620/500,0/500
-ring = 440+480/1000,0/2000
-congestion = 480+620/250,0/250
-callwaiting = 350+440/250,0/250,350+440/250,0/3250
-dialrecall = 300/1500,0/500
-record = 1400/500,0/15000
-info = !950/330,!1400/330,!1800/330,0
-stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
-
-[ve]
-; Tone definition source for ve found on 
-; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
-description = Venezuela / South America
-ringcadence = 1000,4000
-dial = 425
-busy = 425/500,0/500
-ring = 425/1000,0/4000
-congestion = 425/250,0/250
-callwaiting = 400+450/300,0/6000
-dialrecall = 425
-record = 1400/500,0/15000
-info = !950/330,!1440/330,!1800/330,0/1000
-
-
-[za]
-description = South Africa
-; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/safr_sup/saf02.htm
-; (definitions for other countries can also be found there)
-; Note, though, that South Africa uses two switch types in their network --
-; Alcatel switches -- mainly in the Western Cape, and Siemens elsewhere.
-; The former use 383+417 in dial, ringback etc.  The latter use 400*33
-; I've provided both, uncomment the ones you prefer
-ringcadence = 400,200,400,2000
-; dial/ring/callwaiting for the Siemens switches:
-dial = 400*33
-ring = 400*33/400,0/200,400*33/400,0/2000
-callwaiting = 400*33/250,0/250,400*33/250,0/250,400*33/250,0/250,400*33/250,0/250
-; dial/ring/callwaiting for the Alcatel switches:
-; dial = 383+417
-; ring = 383+417/400,0/200,383+417/400,0/2000
-; callwaiting = 383+417/250,0/250,383+417/250,0/250,383+417/250,0/250,383+417/250,0/250
-congestion = 400/250,0/250
-busy = 400/500,0/500
-dialrecall = 350+440
-; XXX Not sure about the RECORDTONE
-record = 1400/500,0/10000
-info = 950/330,1400/330,1800/330,0/330
-stutter = !400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,400*33
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/jabber.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/jabber.conf.TEMPLATE
deleted file mode 100644
index cf71e1f8f4..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/jabber.conf.TEMPLATE
+++ /dev/null
@@ -1,4 +0,0 @@
-[general]
-autoregister=yes
-
-#include jabber_users.conf
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/jabber_users.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/jabber_users.conf.TEMPLATE
deleted file mode 100644
index 3ee2463ed2..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/jabber_users.conf.TEMPLATE
+++ /dev/null
@@ -1,8 +0,0 @@
-[gtalk-|NAME|]
-type=client
-serverhost=talk.google.com
-username=|USERNAME|/Talk
-secret=|SECRET|
-timeout=150
-status=|STATUS|
-statusmessage=|STATUSMESSAGE|
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/logger.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/logger.conf.TEMPLATE
deleted file mode 100644
index e57325013a..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/logger.conf.TEMPLATE
+++ /dev/null
@@ -1,7 +0,0 @@
-[general]
-queue_log = no
-event_log = no
-
-[logfiles]
-console => notice,warning,error
-messages => error
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/manager.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/manager.conf.TEMPLATE
deleted file mode 100644
index 2ac2f0033f..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/manager.conf.TEMPLATE
+++ /dev/null
@@ -1,7 +0,0 @@
-[general]
-enabled = no
-
-port = 5038
-bindaddr = 0.0.0.0
-
-
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/modules.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/modules.conf.TEMPLATE
deleted file mode 100644
index 93c74336d1..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/modules.conf.TEMPLATE
+++ /dev/null
@@ -1,34 +0,0 @@
-[modules]
-autoload=no
-load => res_jabber.so ; Used for Gtalk
-load => res_clioriginate.so ; originate calls from commandline
-load => res_rtp_asterisk.so ; rtp "engine" is now a loadable module in asterisk 1.8
-load => pbx_config.so ; Text Extension Configuration Requires N/A
-load => func_callerid.so ; Gets or sets Caller*ID data on the channel. - Requires ?
-load => func_channel.so
-load => func_logic.so ; Logic functions (if, etc.)
-load => func_strings.so ; string manipulation functions
-load => cdr_manager.so ; Asterisk Call Manager CDR Backend - Requires N/A
-load => chan_local.so ; Show status of local channels- Requires N/A
-load => chan_gtalk.so ; Use gtalk
-load => chan_sip.so ; Session Initiation Protocol (SIP) - Requires res_features.so
-load => codec_alaw.so ; A-law Coder/Decoder - Requires N/A
-load => codec_a_mu.so ; A-law and Mulaw direct Coder/Decoder - Requires N/A
-load => codec_gsm.so ; GSM/PCM16 (signed linear) Codec Translat - Requires N/A
-load => codec_ulaw.so ; Mu-law Coder/Decoder - Requires N/A
-load => format_gsm.so ; Raw GSM data - Requires N/A
-load => format_pcm.so ; Raw uLaw 8khz Audio support (PCM) - Requires N/A
-load => format_wav_gsm.so
-load => app_dial.so ; Dialing Application - Requires res_features.so, res_musiconhold.so
-load => app_parkandannounce.so ; Call Parking and Announce Application - Requires res_features.so
-load => app_playback.so ; Sound File Playback Application - Requires N/A
-load => app_record.so ; Sound File Record Application - Requires N/A
-load => app_system.so ; Execute a system command - Requires N/A
-load => app_disa.so ; Direct Inward System Access
-load => app_authenticate.so ; Authenticate via pin
-load => app_senddtmf.so ; Ability to send DTMF tones on the line.
-load => func_cut.so ; To manipulate strings
-load => func_timeout.so ; Used for DISA timeouts
-
-[global]
-chan_modem.so=no
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/rtp.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/rtp.conf.TEMPLATE
deleted file mode 100644
index 10d577d3a2..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/rtp.conf.TEMPLATE
+++ /dev/null
@@ -1,6 +0,0 @@
-[general]
-rtpstart=|RTPSTART|
-rtpend=|RTPEND|
-rtpchecksums=no
-dtmftimeout=3000
-rtcpinterval = 2000
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sip.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/sip.conf.TEMPLATE
deleted file mode 100644
index 8f3b112ff6..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sip.conf.TEMPLATE
+++ /dev/null
@@ -1,39 +0,0 @@
-[general]
-transport=udp
-context=default-incoming-call-context
-allowoverlap=yes
-allowtransfer=yes
-realm=asterisk
-bindaddr=0.0.0.0
-srvlookup=yes
-maxexpiry=600
-minexpiry=60
-defaultexpiry=300
-qualifyfreq=55
-disallow=all
-allow=ulaw
-allow=alaw
-dtmfmode = inband
-alwaysauthreject = yes
-t1min=100
-timert1=500
-timerb=16000
-rtptimeout=600
-rtpkeepalive=30
-useragent=|USERAGENT|
-localnet=192.168.0.0/16
-localnet=10.0.0.0/8
-localnet=172.16.0.0/12
-nat=yes
-directmedia=no
-sipdebug=no
-bindport=|BINDPORT|
-externhost=|EXTERNHOST|
-externrefresh=60
-
-#include sip_registrations.conf
-
-[authentication]
-
-#include sip_peers.conf
-#include sip_users.conf
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_peer.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_peer.TEMPLATE
deleted file mode 100644
index 30abaadd58..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_peer.TEMPLATE
+++ /dev/null
@@ -1,13 +0,0 @@
-
-[peer-|NAME|]
-type          = peer
-defaultuser   = |DEFAULTUSER|
-fromuser      = |FROMUSER|
-secret        = |SECRET|
-host          = |HOST|
-fromdomain    = |FROMDOMAIN|
-context       = context-incoming-|NAME|
-insecure      = port,invite
-qualify       = 2000
-|PORT|
-|OUTBOUNDPROXY|
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_registration.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_registration.TEMPLATE
deleted file mode 100644
index e139d43f03..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_registration.TEMPLATE
+++ /dev/null
@@ -1,2 +0,0 @@
-register => |DEFAULTUSER|:|SECRET|@peer-|NAME|
-
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_user.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_user.TEMPLATE
deleted file mode 100644
index 61a8b0b86b..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_user.TEMPLATE
+++ /dev/null
@@ -1,11 +0,0 @@
-
-[|DEFAULTUSER|]
-fullname     = |FULLNAME|
-defaultuser  = |DEFAULTUSER|
-secret       = |SECRET|
-hassip       = yes
-hasvoicemail = no
-host         = dynamic
-type         = friend
-context      = context-user-|CONTEXTNAME|
-qualify      = no
\ No newline at end of file
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-alreadyon.gsm b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-alreadyon.gsm
deleted file mode 100644
index 83fe27ecfa..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-alreadyon.gsm
+++ /dev/null
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-incorrect.gsm b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-incorrect.gsm
deleted file mode 100644
index 27d934beb0..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-incorrect.gsm
+++ /dev/null
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-pass.gsm b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-pass.gsm
deleted file mode 100644
index f95637bb32..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-pass.gsm
+++ /dev/null
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/auth-incorrect.gsm b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/auth-incorrect.gsm
deleted file mode 100644
index 12fec25d56..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/auth-incorrect.gsm
+++ /dev/null
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/auth-thankyou.gsm b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/auth-thankyou.gsm
deleted file mode 100644
index 93f936d1a0..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/auth-thankyou.gsm
+++ /dev/null
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/beep.gsm b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/beep.gsm
deleted file mode 100644
index d38eda9cc5..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/beep.gsm
+++ /dev/null
diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/vm-goodbye.gsm b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/vm-goodbye.gsm
deleted file mode 100644
index 735b281c8e..0000000000
--- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/vm-goodbye.gsm
+++ /dev/null