+++ /dev/null
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-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>.
+++ /dev/null
-#
-# 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
+++ /dev/null
---[[
- 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
+++ /dev/null
---[[
- 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
+++ /dev/null
-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."
+++ /dev/null
-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 ""
+++ /dev/null
-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."
+++ /dev/null
-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"
+++ /dev/null
-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."
+++ /dev/null
-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."
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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."
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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."
+++ /dev/null
-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."
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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."
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-#
-# 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。"
+++ /dev/null
-#
-# 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。"
+++ /dev/null
-config 'voicemail' 'global_voicemail'
-
-config 'voicemail' 'voicemail_smtp'
-
-config 'voicemail' 'voicemail_log'
-
+++ /dev/null
-#!/bin/sh
-
-if [ -f "/tmp/voicemail/greeting.gsm" ]
-then
- mv /tmp/voicemail/greeting.gsm /etc/pbx-voicemail/recordings/
-fi
+++ /dev/null
-user |USER|
-password |PASSWORD|
+++ /dev/null
-account default : defaultacct
-
+++ /dev/null
-account defaultacct
-host |HOST|
-port |PORT|
-from voicemail@pbx
-
+++ /dev/null
-defaults
-auth |AUTH|
-tls_certcheck off
-tls |TLS|
-
+++ /dev/null
-#!/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
-
+++ /dev/null
- 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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-
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+++ /dev/null
-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)
+++ /dev/null
-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
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-visit:
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-http://www.digium.com/ivr or http://www.theasteriskvoice.com
-
-LICENSE
--------
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+++ /dev/null
-#
-# 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
+++ /dev/null
---[[
- 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
+++ /dev/null
---[[
- 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
+++ /dev/null
---[[
- 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
+++ /dev/null
---[[
- 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
+++ /dev/null
---[[
- 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
+++ /dev/null
---[[
- 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
+++ /dev/null
---[[
- 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
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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)"
+++ /dev/null
-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"
+++ /dev/null
-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"
+++ /dev/null
-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."
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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"
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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"
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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"
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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 ""
+++ /dev/null
-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。"
+++ /dev/null
-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"
+++ /dev/null
-config 'main' 'connection_status'
+++ /dev/null
-config 'settings' 'advanced'
- option 'useragent' 'PBX'
- option 'ringtime' '30'
- option 'rtpstart' '19850'
- option 'rtpend' '19900'
+++ /dev/null
-config 'call_routing' 'outgoing_calls'
-
-config 'call_routing' 'incoming_calls'
-
-config 'call_routing' 'providers_user_can_use'
-
-config 'call_routing' 'blacklisting'
+++ /dev/null
-config 'user' 'server'
+++ /dev/null
-#!/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 ${MODULENAME}-advanced
- pbx_assemble_and_copy_config
-
- # Link sound files
- pbx_link_sounds
-
- # Enforce ownership of specified files and directories.
- pbx_fix_ownership
-}
+++ /dev/null
-[directories]
-astetcdir => /etc/asterisk
-astmoddir => /usr/lib/asterisk/modules
-astvarlibdir => /usr/lib/asterisk
-astdbdir => /usr/lib/asterisk
-astkeydir => /usr/lib/asterisk
-astdatadir => /usr/lib/asterisk
-astagidir => /usr/lib/asterisk/agi-bin
-astspooldir => /var/spool/asterisk
-astrundir => /var/run/asterisk
-astlogdir => /var/log/asterisk
-
-[options]
-languageprefix = yes
-dumpcore = no
-runuser = nobody
-rungroup = nogroup
+++ /dev/null
-#!/bin/sh
-
-# Check if there are more than one instance of this command
-# with the same command line running at the same time for some
-# reason, then quit. We are checking for the same
-# commandline in order to permit two different callback
-# attempts simultaneously.
-
-if ! grep -q "$@" /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: $@" >> /dev/shm/delayedcallback.err
-fi
+++ /dev/null
-[general]
-static = yes
-writeprotect = yes
-clearglobalvars = no
-
-[globals]
-RINGTIME => |RINGTIME|
-
-[default]
-
-[context-user-hangup-call-context]
-exten => s,1,Hangup()
-exten => _X.,1,Hangup()
-
-[context-catch-all]
-exten => _[!-~].,1,Dial(SIP/${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
+++ /dev/null
-exten => s,n,Gotoif($[ "${CALLERID(NUM)}" = "|BLACKLISTITEM|" ]?context-user-hangup,s,1)
+++ /dev/null
-exten => s,n,Goto(${SOURCECONTEXT},${SOURCEEXTEN},doneblacklist)
-
+++ /dev/null
-
-[blacklist-call-context]
-exten => s,1,Noop()
+++ /dev/null
-exten => s,n,Gotoif($[ "${CALLERID(NUM)}" =~ ".*|NUMBER|" ]?context-user-callback,|NUMBER|,1)
+++ /dev/null
-exten => s,n,Goto(${SOURCECONTEXT},${SOURCEEXTEN},donecallback)
-
+++ /dev/null
-
-[callback-check-call-context]
-exten => s,1,Noop()
+++ /dev/null
-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()
-
+++ /dev/null
-[context-user-callback]
+++ /dev/null
-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()
-
+++ /dev/null
-[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)
+++ /dev/null
-exten => |DEFAULTUSER|,1,Goto(default-incoming-call-context,s,1)
+++ /dev/null
-exten => s,n,Gotoif($[ "${CALLERID(NUM)}" =~ ".*|NUMBER|" ]?disa-call-context,|NUMBER|,1)
+++ /dev/null
-exten => s,n,Goto(${SOURCECONTEXT},${SOURCEEXTEN},donedisacheck)
+++ /dev/null
-[disa-check-call-context]
-exten => s,1,Noop()
+++ /dev/null
-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|)
-
+++ /dev/null
-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|)
-
+++ /dev/null
-[disa-call-context]
+++ /dev/null
-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|,${RINGTIME},D(:w11111111))
-same => n,Goto(context-voicemail,s,1)
-
+++ /dev/null
-[context-incoming-gtalk]
+++ /dev/null
-
-[context-incoming-|NAME|]
-exten => s,1,NoOp(${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|,${RINGTIME},r)
-exten => s,n,Goto(context-voicemail,s,1)
+++ /dev/null
-exten => |DEFAULTUSER|,1,Dial(SIP/|DEFAULTUSER|,${RINGTIME},r)
+++ /dev/null
-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)
-
+++ /dev/null
-exten => |PATTERN|,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN|SYMBOLSTOREMOVE|}@voice.google.com,60)
-
+++ /dev/null
-exten => |PATTERN|,1,Dial(SIP/${EXTEN|SYMBOLSTOREMOVE|}@peer-|NAME|,60,r)
+++ /dev/null
-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)
+++ /dev/null
-include => context-voicemail-record-greeting
-include => context-catch-all
+++ /dev/null
-
-[context-user-|DEFAULTUSER|]
-
+++ /dev/null
-[context-voicemail-record-greeting]
-
-[context-voicemail]
-exten => s,1,Hangup()
+++ /dev/null
-[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)}' &)
+++ /dev/null
-[general]
-context=context-incoming-gtalk
-allowguest=yes
-allowguests=yes
-bindaddr=0.0.0.0
-
-[guest]
-disallow=all
-allow=ulaw
-context=context-incoming-gtalk
+++ /dev/null
-; 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
+++ /dev/null
-[general]
-autoregister=yes
-
-#include jabber_users.conf
+++ /dev/null
-[gtalk-|NAME|]
-type=client
-serverhost=talk.google.com
-username=|USERNAME|/Talk
-secret=|SECRET|
-timeout=150
-status=|STATUS|
-statusmessage=|STATUSMESSAGE|
+++ /dev/null
-[general]
-queue_log = no
-event_log = no
-
-[logfiles]
-console => notice,warning,error
-messages => error
+++ /dev/null
-[general]
-enabled = no
-
-port = 5038
-bindaddr = 0.0.0.0
-
-
+++ /dev/null
-[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
+++ /dev/null
-[general]
-rtpstart=|RTPSTART|
-rtpend=|RTPEND|
-rtpchecksums=no
-dtmftimeout=3000
-rtcpinterval = 2000
+++ /dev/null
-[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
+++ /dev/null
-
-[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|
+++ /dev/null
-register => |DEFAULTUSER|:|SECRET|@peer-|NAME|
-
+++ /dev/null
-
-[|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
projects / project / luci.git / commitdiff