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diff --git a/applications/luci-app-pbx/COPYING b/applications/luci-app-pbx/COPYING deleted file mode 100644 index 94a9ed024d..0000000000 --- a/applications/luci-app-pbx/COPYING +++ /dev/null @@ -1,674 +0,0 @@ - GNU GENERAL PUBLIC LICENSE - Version 3, 29 June 2007 - - Copyright (C) 2007 Free Software Foundation, Inc. <http://fsf.org/> - Everyone is permitted to copy and distribute verbatim copies - of this license document, but changing it is not allowed. - - Preamble - - The GNU General Public License is a free, copyleft license for -software and other kinds of works. - - The licenses for most software and other practical works are designed -to take away your freedom to share and change the works. By contrast, -the GNU General Public License is intended to guarantee your freedom to -share and change all versions of a program--to make sure it remains free -software for all its users. 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Of course, your program's commands -might be different; for a GUI interface, you would use an "about box". - - You should also get your employer (if you work as a programmer) or school, -if any, to sign a "copyright disclaimer" for the program, if necessary. -For more information on this, and how to apply and follow the GNU GPL, see -<http://www.gnu.org/licenses/>. - - The GNU General Public License does not permit incorporating your program -into proprietary programs. If your program is a subroutine library, you -may consider it more useful to permit linking proprietary applications with -the library. If this is what you want to do, use the GNU Lesser General -Public License instead of this License. But first, please read -<http://www.gnu.org/philosophy/why-not-lgpl.html>. diff --git a/applications/luci-app-pbx/CREDITS-SOUNDS b/applications/luci-app-pbx/CREDITS-SOUNDS deleted file mode 100644 index 1fa64bc6cb..0000000000 --- a/applications/luci-app-pbx/CREDITS-SOUNDS +++ /dev/null @@ -1,7 +0,0 @@ -This file pertains to the sounds files included in root/etc/pbx-asterisk/sounds - -Recorded by: -Allison Smith (http://www.theivrvoice.com) - -Financial Contributions by: -Digium, Inc. (http://www.digium.com) diff --git a/applications/luci-app-pbx/LICENSE-SOUNDS b/applications/luci-app-pbx/LICENSE-SOUNDS deleted file mode 100644 index fe9c8221a2..0000000000 --- a/applications/luci-app-pbx/LICENSE-SOUNDS +++ /dev/null @@ -1,312 +0,0 @@ -This file pertains to the sounds files included in root/etc/pbx-asterisk/sounds - -LICENSE FOR VOICE PROMPT FILES ------------------------------- - -The voice prompt files distributed herewith are Copyright (C) 2003-2008 -Allison Smith, and provided under terms of the following License. For -more information, or to purchase custom voice prompt files, please -visit: - -http://www.digium.com/ivr or http://www.theasteriskvoice.com - -LICENSE -------- - -THE WORK (AS DEFINED BELOW) IS PROVIDED UNDER THE TERMS OF THIS -CREATIVE COMMONS PUBLIC LICENSE ("CCPL" OR "LICENSE"). THE WORK IS -PROTECTED BY COPYRIGHT AND/OR OTHER APPLICABLE LAW. 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"Derivative Work" means a work based upon the Work or upon the Work -and other pre-existing works, such as a translation, musical -arrangement, dramatization, fictionalization, motion picture version, -sound recording, art reproduction, abridgment, condensation, or any -other form in which the Work may be recast, transformed, or adapted, -except that a work that constitutes a Collective Work will not be -considered a Derivative Work for the purpose of this License. For the -avoidance of doubt, where the Work is a musical composition or sound -recording, the synchronization of the Work in timed-relation with a -moving image ("synching") will be considered a Derivative Work for the -purpose of this License. - -d. "License Elements" means the following high-level license -attributes as selected by Licensor and indicated in the title of this -License: Attribution, ShareAlike. - -e. 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This License may not be -modified without the mutual written agreement of the Licensor and You. diff --git a/applications/luci-app-pbx/Makefile b/applications/luci-app-pbx/Makefile deleted file mode 100644 index 772713b444..0000000000 --- a/applications/luci-app-pbx/Makefile +++ /dev/null @@ -1,19 +0,0 @@ -# -# Copyright (C) 2008-2014 The LuCI Team <luci@lists.subsignal.org> -# -# This is free software, licensed under the Apache License, Version 2.0 . -# - -include $(TOPDIR)/rules.mk - -LUCI_TITLE:=LuCI PBX Administration -LUCI_DEPENDS:= @BROKEN \ - +asterisk18 +asterisk18-app-authenticate +asterisk18-app-disa \ - +asterisk18-app-setcallerid +asterisk18-app-system +asterisk18-chan-gtalk \ - +asterisk18-codec-a-mu +asterisk18-codec-alaw +asterisk18-func-cut \ - +asterisk18-res-clioriginate +asterisk18-func-channel +asterisk18-chan-local \ - +asterisk18-app-record +asterisk18-app-senddtmf +asterisk18-res-crypto - -include ../../luci.mk - -# call BuildPackage - OpenWrt buildroot signature diff --git a/applications/luci-app-pbx/luasrc/controller/pbx.lua b/applications/luci-app-pbx/luasrc/controller/pbx.lua deleted file mode 100644 index b77814b150..0000000000 --- a/applications/luci-app-pbx/luasrc/controller/pbx.lua +++ /dev/null @@ -1,29 +0,0 @@ ---[[ - Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com> - - This file is part of luci-pbx. - - luci-pbx is free software: you can redistribute it and/or modify - it under the terms of the GNU General Public License as published by - the Free Software Foundation, either version 3 of the License, or - (at your option) any later version. - - luci-pbx is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License for more details. - - You should have received a copy of the GNU General Public License - along with luci-pbx. If not, see <http://www.gnu.org/licenses/>. -]]-- - -module("luci.controller.pbx", package.seeall) - -function index() - entry({"admin", "services", "pbx"}, cbi("pbx"), "PBX", 80) - entry({"admin", "services", "pbx", "pbx-google"}, cbi("pbx-google"), "Google Accounts", 1) - entry({"admin", "services", "pbx", "pbx-voip"}, cbi("pbx-voip"), "SIP Accounts", 2) - entry({"admin", "services", "pbx", "pbx-users"}, cbi("pbx-users"), "User Accounts", 3) - entry({"admin", "services", "pbx", "pbx-calls"}, cbi("pbx-calls"), "Call Routing", 4) - entry({"admin", "services", "pbx", "pbx-advanced"}, cbi("pbx-advanced"), "Advanced Settings", 6) -end diff --git a/applications/luci-app-pbx/luasrc/model/cbi/pbx-advanced.lua b/applications/luci-app-pbx/luasrc/model/cbi/pbx-advanced.lua deleted file mode 100644 index 34288c6632..0000000000 --- a/applications/luci-app-pbx/luasrc/model/cbi/pbx-advanced.lua +++ /dev/null @@ -1,293 +0,0 @@ ---[[ - Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com> - - This file is part of luci-pbx. - - luci-pbx is free software: you can redistribute it and/or modify - it under the terms of the GNU General Public License as published by - the Free Software Foundation, either version 3 of the License, or - (at your option) any later version. - - luci-pbx is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License for more details. - - You should have received a copy of the GNU General Public License - along with luci-pbx. If not, see <http://www.gnu.org/licenses/>. -]]-- - -if nixio.fs.access("/etc/init.d/asterisk") then - server = "asterisk" -elseif nixio.fs.access("/etc/init.d/freeswitch") then - server = "freeswitch" -else - server = "" -end - -appname = "PBX" -modulename = "pbx-advanced" -defaultbindport = 5060 -defaultrtpstart = 19850 -defaultrtpend = 19900 - --- Returns all the network related settings, including a constructed RTP range -function get_network_info() - externhost = m.uci:get(modulename, "advanced", "externhost") - ipaddr = m.uci:get("network", "lan", "ipaddr") - bindport = m.uci:get(modulename, "advanced", "bindport") - rtpstart = m.uci:get(modulename, "advanced", "rtpstart") - rtpend = m.uci:get(modulename, "advanced", "rtpend") - - if bindport == nil then bindport = defaultbindport end - if rtpstart == nil then rtpstart = defaultrtpstart end - if rtpend == nil then rtpend = defaultrtpend end - - if rtpstart == nil or rtpend == nil then - rtprange = nil - else - rtprange = rtpstart .. "-" .. rtpend - end - - return bindport, rtprange, ipaddr, externhost -end - --- If not present, insert empty rules in the given config & section named PBX-SIP and PBX-RTP -function insert_empty_sip_rtp_rules(config, section) - - -- Add rules named PBX-SIP and PBX-RTP if not existing - found_sip_rule = false - found_rtp_rule = false - m.uci:foreach(config, section, - function(s1) - if s1._name == 'PBX-SIP' then - found_sip_rule = true - elseif s1._name == 'PBX-RTP' then - found_rtp_rule = true - end - end) - - if found_sip_rule ~= true then - newrule=m.uci:add(config, section) - m.uci:set(config, newrule, '_name', 'PBX-SIP') - end - if found_rtp_rule ~= true then - newrule=m.uci:add(config, section) - m.uci:set(config, newrule, '_name', 'PBX-RTP') - end -end - --- Delete rules in the given config & section named PBX-SIP and PBX-RTP -function delete_sip_rtp_rules(config, section) - - -- Remove rules named PBX-SIP and PBX-RTP - commit = false - m.uci:foreach(config, section, - function(s1) - if s1._name == 'PBX-SIP' or s1._name == 'PBX-RTP' then - m.uci:delete(config, s1['.name']) - commit = true - end - end) - - -- If something changed, then we commit the config. - if commit == true then m.uci:commit(config) end -end - --- Deletes QoS rules associated with this PBX. -function delete_qos_rules() - delete_sip_rtp_rules ("qos", "classify") -end - - -function insert_qos_rules() - -- Insert empty PBX-SIP and PBX-RTP rules if not present. - insert_empty_sip_rtp_rules ("qos", "classify") - - -- Get the network information - bindport, rtprange, ipaddr, externhost = get_network_info() - - -- Iterate through the QoS rules, and if there is no other rule with the same port - -- range at the priority service level, insert this rule. - commit = false - m.uci:foreach("qos", "classify", - function(s1) - if s1._name == 'PBX-SIP' then - if s1.ports ~= bindport or s1.target ~= "Priority" or s1.proto ~= "udp" then - m.uci:set("qos", s1['.name'], "ports", bindport) - m.uci:set("qos", s1['.name'], "proto", "udp") - m.uci:set("qos", s1['.name'], "target", "Priority") - commit = true - end - elseif s1._name == 'PBX-RTP' then - if s1.ports ~= rtprange or s1.target ~= "Priority" or s1.proto ~= "udp" then - m.uci:set("qos", s1['.name'], "ports", rtprange) - m.uci:set("qos", s1['.name'], "proto", "udp") - m.uci:set("qos", s1['.name'], "target", "Priority") - commit = true - end - end - end) - - -- If something changed, then we commit the qos config. - if commit == true then m.uci:commit("qos") end -end - --- This function is a (so far) unsuccessful attempt to manipulate the firewall rules from here --- Need to do more testing and eventually move to this mode. -function maintain_firewall_rules() - -- Get the network information - bindport, rtprange, ipaddr, externhost = get_network_info() - - commit = false - -- Only if externhost is set, do we control firewall rules. - if externhost ~= nil and bindport ~= nil and rtprange ~= nil then - -- Insert empty PBX-SIP and PBX-RTP rules if not present. - insert_empty_sip_rtp_rules ("firewall", "rule") - - -- Iterate through the firewall rules, and if the dest_port and dest_ip setting of the\ - -- SIP and RTP rule do not match what we want configured, set all the entries in the rule\ - -- appropriately. - m.uci:foreach("firewall", "rule", - function(s1) - if s1._name == 'PBX-SIP' then - if s1.dest_port ~= bindport then - m.uci:set("firewall", s1['.name'], "dest_port", bindport) - m.uci:set("firewall", s1['.name'], "src", "wan") - m.uci:set("firewall", s1['.name'], "proto", "udp") - m.uci:set("firewall", s1['.name'], "target", "ACCEPT") - commit = true - end - elseif s1._name == 'PBX-RTP' then - if s1.dest_port ~= rtprange then - m.uci:set("firewall", s1['.name'], "dest_port", rtprange) - m.uci:set("firewall", s1['.name'], "src", "wan") - m.uci:set("firewall", s1['.name'], "proto", "udp") - m.uci:set("firewall", s1['.name'], "target", "ACCEPT") - commit = true - end - end - end) - else - -- We delete the firewall rules if one or more of the necessary parameters are not set. - sip_rule_name=nil - rtp_rule_name=nil - - -- First discover the configuration names of the rules. - m.uci:foreach("firewall", "rule", - function(s1) - if s1._name == 'PBX-SIP' then - sip_rule_name = s1['.name'] - elseif s1._name == 'PBX-RTP' then - rtp_rule_name = s1['.name'] - end - end) - - -- Then, using the names, actually delete the rules. - if sip_rule_name ~= nil then - m.uci:delete("firewall", sip_rule_name) - commit = true - end - if rtp_rule_name ~= nil then - m.uci:delete("firewall", rtp_rule_name) - commit = true - end - end - - -- If something changed, then we commit the firewall config. - if commit == true then m.uci:commit("firewall") end -end - -m = Map (modulename, translate("Advanced Settings"), - translate("This section contains settings that do not need to be changed under \ - normal circumstances. In addition, here you can configure your system \ - for use with remote SIP devices, and resolve call quality issues by enabling \ - the insertion of QoS rules.")) - --- Recreate the voip server config, and restart necessary services after changes are commited --- to the advanced configuration. The firewall must restart because of "Remote Usage". -function m.on_after_commit(self) - - -- Make sure firewall rules are in place - maintain_firewall_rules() - - -- If insertion of QoS rules is enabled - if m.uci:get(modulename, "advanced", "qos_enabled") == "yes" then - insert_qos_rules() - else - delete_qos_rules() - end - - luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null") - luci.sys.call("/etc/init.d/" .. server .. " restart 1\>/dev/null 2\>/dev/null") - luci.sys.call("/etc/init.d/firewall restart 1\>/dev/null 2\>/dev/null") -end - ------------------------------------------------------------------------------ -s = m:section(NamedSection, "advanced", "settings", translate("Advanced Settings")) -s.anonymous = true - -s:tab("general", translate("General Settings")) -s:tab("remote_usage", translate("Remote Usage"), - translatef("You can use your SIP devices/softphones with this system from a remote location \ - as well, as long as your Internet Service Provider gives you a public IP. \ - You will be able to call other local users for free (e.g. other Analog Telephone Adapters (ATAs)) \ - and use your VoIP providers to make calls as if you were local to the PBX. \ - After configuring this tab, go back to where users are configured and see the new \ - Server and Port setting you need to configure the remote SIP devices with. Please note that if this \ - PBX is not running on your router/gateway, you will need to configure port forwarding (NAT) on your \ - router/gateway. Please forward the ports below (SIP port and RTP range) to the IP address of the \ - device running this PBX.")) - -s:tab("qos", translate("QoS Settings"), - translate("If you experience jittery or high latency audio during heavy downloads, you may want \ - to enable QoS. QoS prioritizes traffic to and from your network for specified ports and IP \ - addresses, resulting in better latency and throughput for sound in our case. If enabled below, \ - a QoS rule for this service will be configured by the PBX automatically, but you must visit the \ - QoS configuration page (Network->QoS) to configure other critical QoS settings like Download \ - and Upload speed.")) - -ringtime = s:taboption("general", Value, "ringtime", translate("Number of Seconds to Ring"), - translate("Set the number of seconds to ring users upon incoming calls before hanging up \ - or going to voicemail, if the voicemail is installed and enabled.")) -ringtime.datatype = "port" -ringtime.default = 30 - -ua = s:taboption("general", Value, "useragent", translate("User Agent String"), - translate("This is the name that the VoIP server will use to identify itself when \ - registering to VoIP (SIP) providers. Some providers require this to a specific \ - string matching a hardware SIP device.")) -ua.default = appname - -h = s:taboption("remote_usage", Value, "externhost", translate("Domain/IP Address/Dynamic Domain"), - translate("You can enter your domain name, external IP address, or dynamic domain name here. \ - The best thing to input is a static IP address. If your IP address is dynamic and it changes, \ - your configuration will become invalid. Hence, it's recommended to set up Dynamic DNS in this case. \ - and enter your Dynamic DNS hostname here. You can configure Dynamic DNS with the luci-app-ddns package.")) -h.datatype = "host(0)" - -p = s:taboption("remote_usage", Value, "bindport", translate("External SIP Port"), - translate("Pick a random port number between 6500 and 9500 for the service to listen on. \ - Do not pick the standard 5060, because it is often subject to brute-force attacks. \ - When finished, (1) click \"Save and Apply\", and (2) look in the \ - \"SIP Device/Softphone Accounts\" section for updated Server and Port settings \ - for your SIP Devices/Softphones.")) -p.datatype = "port" - -p = s:taboption("remote_usage", Value, "rtpstart", translate("RTP Port Range Start"), - translate("RTP traffic carries actual voice packets. This is the start of the port range \ - that will be used for setting up RTP communication. It's usually OK to leave this \ - at the default value.")) -p.datatype = "port" -p.default = defaultrtpstart - -p = s:taboption("remote_usage", Value, "rtpend", translate("RTP Port Range End")) -p.datatype = "port" -p.default = defaultrtpend - -p = s:taboption("qos", ListValue, "qos_enabled", translate("Insert QoS Rules")) -p:value("yes", translate("Yes")) -p:value("no", translate("No")) -p.default = "yes" - -return m diff --git a/applications/luci-app-pbx/luasrc/model/cbi/pbx-calls.lua b/applications/luci-app-pbx/luasrc/model/cbi/pbx-calls.lua deleted file mode 100644 index ca373d63a3..0000000000 --- a/applications/luci-app-pbx/luasrc/model/cbi/pbx-calls.lua +++ /dev/null @@ -1,424 +0,0 @@ ---[[ - Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com> - - This file is part of luci-pbx. - - luci-pbx is free software: you can redistribute it and/or modify - it under the terms of the GNU General Public License as published by - the Free Software Foundation, either version 3 of the License, or - (at your option) any later version. - - luci-pbx is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License for more details. - - You should have received a copy of the GNU General Public License - along with luci-pbx. If not, see <http://www.gnu.org/licenses/>. -]]-- - -if nixio.fs.access("/etc/init.d/asterisk") then - server = "asterisk" -elseif nixio.fs.access("/etc/init.d/freeswitch") then - server = "freeswitch" -else - server = "" -end - -modulename = "pbx-calls" -voipmodulename = "pbx-voip" -googlemodulename = "pbx-google" -usersmodulename = "pbx-users" -allvalidaccounts = {} -nallvalidaccounts = 0 -validoutaccounts = {} -nvalidoutaccounts = 0 -validinaccounts = {} -nvalidinaccounts = 0 -allvalidusers = {} -nallvalidusers = 0 -validoutusers = {} -nvalidoutusers = 0 - - --- Checks whether the entered extension is valid syntactically. -function is_valid_extension(exten) - return (exten:match("[#*+0-9NXZ]+$") ~= nil) -end - - -m = Map (modulename, translate("Call Routing"), - translate("This is where you indicate which Google/SIP accounts are used to call what \ - country/area codes, which users can use what SIP/Google accounts, how incoming \ - calls are routed, what numbers can get into this PBX with a password, and what \ - numbers are blacklisted.")) - --- Recreate the config, and restart services after changes are commited to the configuration. -function m.on_after_commit(self) - luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null") - luci.sys.call("/etc/init.d/" .. server .. " restart 1\>/dev/null 2\>/dev/null") -end - --- Add Google accounts to all valid accounts, and accounts valid for incoming and outgoing calls. -m.uci:foreach(googlemodulename, "gtalk_jabber", - function(s1) - -- Add this provider to list of valid accounts. - if s1.username ~= nil and s1.name ~= nil then - allvalidaccounts[s1.name] = s1.username - nallvalidaccounts = nallvalidaccounts + 1 - - if s1.make_outgoing_calls == "yes" then - -- Add provider to the associative array of valid outgoing accounts. - validoutaccounts[s1.name] = s1.username - nvalidoutaccounts = nvalidoutaccounts + 1 - end - - if s1.register == "yes" then - -- Add provider to the associative array of valid outgoing accounts. - validinaccounts[s1.name] = s1.username - nvalidinaccounts = nvalidinaccounts + 1 - end - end - end) - --- Add SIP accounts to all valid accounts, and accounts valid for incoming and outgoing calls. -m.uci:foreach(voipmodulename, "voip_provider", - function(s1) - -- Add this provider to list of valid accounts. - if s1.defaultuser ~= nil and s1.host ~= nil and s1.name ~= nil then - allvalidaccounts[s1.name] = s1.defaultuser .. "@" .. s1.host - nallvalidaccounts = nallvalidaccounts + 1 - - if s1.make_outgoing_calls == "yes" then - -- Add provider to the associative array of valid outgoing accounts. - validoutaccounts[s1.name] = s1.defaultuser .. "@" .. s1.host - nvalidoutaccounts = nvalidoutaccounts + 1 - end - - if s1.register == "yes" then - -- Add provider to the associative array of valid outgoing accounts. - validinaccounts[s1.name] = s1.defaultuser .. "@" .. s1.host - nvalidinaccounts = nvalidinaccounts + 1 - end - end - end) - --- Add Local User accounts to all valid users, and users allowed to make outgoing calls. -m.uci:foreach(usersmodulename, "local_user", - function(s1) - -- Add user to list of all valid users. - if s1.defaultuser ~= nil then - allvalidusers[s1.defaultuser] = true - nallvalidusers = nallvalidusers + 1 - - if s1.can_call == "yes" then - validoutusers[s1.defaultuser] = true - nvalidoutusers = nvalidoutusers + 1 - end - end - end) - - ----------------------------------------------------------------------------------------------------- --- If there are no accounts configured, or no accounts enabled for outgoing calls, display a warning. --- Otherwise, display the usual help text within the section. -if nallvalidaccounts == 0 then - text = translate("NOTE: There are no Google or SIP provider accounts configured.") -elseif nvalidoutaccounts == 0 then - text = translate("NOTE: There are no Google or SIP provider accounts enabled for outgoing calls.") -else - text = translate("If you have more than one account that can make outgoing calls, you \ - should enter a list of phone numbers and/or prefixes in the following fields for each \ - provider listed. Invalid prefixes are removed silently, and only 0-9, X, Z, N, #, *, \ - and + are valid characters. The letter X matches 0-9, Z matches 1-9, and N matches 2-9. \ - For example to make calls to Germany through a provider, you can enter 49. To make calls \ - to North America, you can enter 1NXXNXXXXXX. If one of your providers can make \"local\" \ - calls to an area code like New York's 646, you can enter 646NXXXXXX for that \ - provider. You should leave one account with an empty list to make calls with \ - it by default, if no other provider's prefixes match. The system will automatically \ - replace an empty list with a message that the provider dials all numbers not matched by another \ - provider's prefixes. Be as specific as possible (i.e. 1NXXNXXXXXX is better than 1). Please note \ - all international dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a \ - space-separated list, and/or one per line by hitting enter after every one.") -end - - -s = m:section(NamedSection, "outgoing_calls", "call_routing", translate("Outgoing Calls"), text) -s.anonymous = true - -for k,v in pairs(validoutaccounts) do - patterns = s:option(DynamicList, k, v) - - -- If the saved field is empty, we return a string - -- telling the user that this provider would dial any exten. - function patterns.cfgvalue(self, section) - value = self.map:get(section, self.option) - - if value == nil then - return {translate("Dials numbers unmatched elsewhere")} - else - return value - end - end - - -- Write only valid extensions into the config file. - function patterns.write(self, section, value) - newvalue = {} - nindex = 1 - for index, field in ipairs(value) do - val = luci.util.trim(value[index]) - if is_valid_extension(val) == true then - newvalue[nindex] = val - nindex = nindex + 1 - end - end - DynamicList.write(self, section, newvalue) - end -end - ----------------------------------------------------------------------------------------------------- --- If there are no accounts configured, or no accounts enabled for incoming calls, display a warning. --- Otherwise, display the usual help text within the section. -if nallvalidaccounts == 0 then - text = translate("NOTE: There are no Google or SIP provider accounts configured.") -elseif nvalidinaccounts == 0 then - text = translate("NOTE: There are no Google or SIP provider accounts enabled for incoming calls.") -else - text = translate("For each provider enabled for incoming calls, here you can restrict which users to\ - ring on incoming calls. If the list is empty, the system will indicate that all users \ - enabled for incoming calls will ring. Invalid usernames will be rejected \ - silently. Also, entering a username here overrides the user's setting to not receive \ - incoming calls. This way, you can make certain users ring only for specific providers. \ - Entries can be made in a space-separated list, and/or one per line by hitting enter after \ - every one.") -end - - -s = m:section(NamedSection, "incoming_calls", "call_routing", translate("Incoming Calls"), text) -s.anonymous = true - -for k,v in pairs(validinaccounts) do - users = s:option(DynamicList, k, v) - - -- If the saved field is empty, we return a string telling the user that - -- this provider would ring all users configured for incoming calls. - function users.cfgvalue(self, section) - value = self.map:get(section, self.option) - - if value == nil then - return {translate("Rings users enabled for incoming calls")} - else - return value - end - end - - -- Write only valid user names. - function users.write(self, section, value) - newvalue = {} - nindex = 1 - for index, field in ipairs(value) do - trimuser = luci.util.trim(value[index]) - if allvalidusers[trimuser] == true then - newvalue[nindex] = trimuser - nindex = nindex + 1 - end - end - DynamicList.write(self, section, newvalue) - end -end - - ----------------------------------------------------------------------------------------------------- --- If there are no user accounts configured, no user accounts enabled for outgoing calls, --- display a warning. Otherwise, display the usual help text within the section. -if nallvalidusers == 0 then - text = translate("NOTE: There are no local user accounts configured.") -elseif nvalidoutusers == 0 then - text = translate("NOTE: There are no local user accounts enabled for outgoing calls.") -else - text = translate("For each user enabled for outgoing calls you can restrict what providers the user \ - can use for outgoing calls. By default all users can use all providers. To show up in the list \ - below the user should be allowed to make outgoing calls in the \"User Accounts\" page. Enter VoIP \ - providers in the format username@some.host.name, as listed in \"Outgoing Calls\" above. It's \ - easiest to copy and paste the providers from above. Invalid entries, including providers not \ - enabled for outgoing calls, will be rejected silently. Entries can be made in a space-separated \ - list, and/or one per line by hitting enter after every one.") -end - - -s = m:section(NamedSection, "providers_user_can_use", "call_routing", - translate("Providers Used for Outgoing Calls"), text) -s.anonymous = true - -for k,v in pairs(validoutusers) do - providers = s:option(DynamicList, k, k) - - -- If the saved field is empty, we return a string telling the user - -- that this user uses all providers enavled for outgoing calls. - function providers.cfgvalue(self, section) - value = self.map:get(section, self.option) - - if value == nil then - return {translate("Uses providers enabled for outgoing calls")} - else - newvalue = {} - -- Convert internal names to user@host values. - for i,v in ipairs(value) do - newvalue[i] = validoutaccounts[v] - end - return newvalue - end - end - - -- Cook the new values prior to entering them into the config file. - -- Also, enter them only if they are valid. - function providers.write(self, section, value) - cookedvalue = {} - cindex = 1 - for index, field in ipairs(value) do - cooked = string.gsub(luci.util.trim(value[index]), "%W", "_") - if validoutaccounts[cooked] ~= nil then - cookedvalue[cindex] = cooked - cindex = cindex + 1 - end - end - DynamicList.write(self, section, cookedvalue) - end -end - ----------------------------------------------------------------------------------------------------- -s = m:section(TypedSection, "callthrough_numbers", translate("Call-through Numbers"), - translate("Designate numbers that are allowed to call through this system and which user's \ - privileges they will have.")) -s.anonymous = true -s.addremove = true - -num = s:option(DynamicList, "callthrough_number_list", translate("Call-through Numbers"), - translate("Specify numbers individually here. Press enter to add more numbers. \ - You will have to experiment with what country and area codes you need to add \ - to the number.")) -num.datatype = "uinteger" - -p = s:option(ListValue, "enabled", translate("Enabled")) -p:value("yes", translate("Yes")) -p:value("no", translate("No")) -p.default = "yes" - -user = s:option(Value, "defaultuser", translate("User Name"), - translate("The number(s) specified above will be able to dial out with this user's providers. \ - Invalid usernames, including users not enabled for outgoing calls, are dropped silently. \ - Please verify that the entry was accepted.")) -function user.write(self, section, value) - trimuser = luci.util.trim(value) - if allvalidusers[trimuser] == true then - Value.write(self, section, trimuser) - end -end - -pwd = s:option(Value, "pin", translate("PIN"), - translate("Your PIN disappears when saved for your protection. It will be changed \ - only when you enter a value different from the saved one. Leaving the PIN \ - empty is possible, but please beware of the security implications.")) -pwd.password = true -pwd.rmempty = false - --- We skip reading off the saved value and return nothing. -function pwd.cfgvalue(self, section) - return "" -end - --- We check the entered value against the saved one, and only write if the entered value is --- something other than the empty string, and it differes from the saved value. -function pwd.write(self, section, value) - local orig_pwd = m:get(section, self.option) - if value and #value > 0 and orig_pwd ~= value then - Value.write(self, section, value) - end -end - ----------------------------------------------------------------------------------------------------- -s = m:section(TypedSection, "callback_numbers", translate("Call-back Numbers"), - translate("Designate numbers to whom the system will hang up and call back, which provider will \ - be used to call them, and which user's privileges will be granted to them.")) -s.anonymous = true -s.addremove = true - -num = s:option(DynamicList, "callback_number_list", translate("Call-back Numbers"), - translate("Specify numbers individually here. Press enter to add more numbers. \ - You will have to experiment with what country and area codes you need to add \ - to the number.")) -num.datatype = "uinteger" - -p = s:option(ListValue, "enabled", translate("Enabled")) -p:value("yes", translate("Yes")) -p:value("no", translate("No")) -p.default = "yes" - -delay = s:option(Value, "callback_hangup_delay", translate("Hang-up Delay"), - translate("How long to wait before hanging up. If the provider you use to dial automatically forwards \ - to voicemail, you can set this value to a delay that will allow you to hang up before your call gets \ - forwarded and you get billed for it.")) -delay.datatype = "uinteger" -delay.default = 0 - -user = s:option(Value, "defaultuser", translate("User Name"), - translate("The number(s) specified above will be able to dial out with this user's providers. \ - Invalid usernames, including users not enabled for outgoing calls, are dropped silently. \ - Please verify that the entry was accepted.")) -function user.write(self, section, value) - trimuser = luci.util.trim(value) - if allvalidusers[trimuser] == true then - Value.write(self, section, trimuser) - end -end - -pwd = s:option(Value, "pin", translate("PIN"), - translate("Your PIN disappears when saved for your protection. It will be changed \ - only when you enter a value different from the saved one. Leaving the PIN \ - empty is possible, but please beware of the security implications.")) -pwd.password = true -pwd.rmempty = false - --- We skip reading off the saved value and return nothing. -function pwd.cfgvalue(self, section) - return "" -end - --- We check the entered value against the saved one, and only write if the entered value is --- something other than the empty string, and it differes from the saved value. -function pwd.write(self, section, value) - local orig_pwd = m:get(section, self.option) - if value and #value > 0 and orig_pwd ~= value then - Value.write(self, section, value) - end -end - -provider = s:option(Value, "callback_provider", translate("Call-back Provider"), - translate("Enter a VoIP provider to use for call-back in the format username@some.host.name, as listed in \ - \"Outgoing Calls\" above. It's easiest to copy and paste the providers from above. Invalid entries, including \ - providers not enabled for outgoing calls, will be rejected silently.")) -function provider.write(self, section, value) - cooked = string.gsub(luci.util.trim(value), "%W", "_") - if validoutaccounts[cooked] ~= nil then - Value.write(self, section, value) - end -end - ----------------------------------------------------------------------------------------------------- -s = m:section(NamedSection, "blacklisting", "call_routing", translate("Blacklisted Numbers"), - translate("Enter phone numbers that you want to decline calls from automatically. \ - You should probably omit the country code and any leading zeroes, but please \ - experiment to make sure you are blocking numbers from your desired area successfully.")) -s.anonymous = true - -b = s:option(DynamicList, "blacklist1", translate("Dynamic List of Blacklisted Numbers"), - translate("Specify numbers individually here. Press enter to add more numbers.")) -b.cast = "string" -b.datatype = "uinteger" - -b = s:option(Value, "blacklist2", translate("Space-Separated List of Blacklisted Numbers"), - translate("Copy-paste large lists of numbers here.")) -b.template = "cbi/tvalue" -b.rows = 3 - -return m diff --git a/applications/luci-app-pbx/luasrc/model/cbi/pbx-google.lua b/applications/luci-app-pbx/luasrc/model/cbi/pbx-google.lua deleted file mode 100644 index 3c36a168d9..0000000000 --- a/applications/luci-app-pbx/luasrc/model/cbi/pbx-google.lua +++ /dev/null @@ -1,122 +0,0 @@ ---[[ - Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com> - - This file is part of luci-pbx. - - luci-pbx is free software: you can redistribute it and/or modify - it under the terms of the GNU General Public License as published by - the Free Software Foundation, either version 3 of the License, or - (at your option) any later version. - - luci-pbx is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License for more details. - - You should have received a copy of the GNU General Public License - along with luci-pbx. If not, see <http://www.gnu.org/licenses/>. -]]-- - -if nixio.fs.access("/etc/init.d/asterisk") then - server = "asterisk" -elseif nixio.fs.access("/etc/init.d/freeswitch") then - server = "freeswitch" -else - server = "" -end - -modulename = "pbx-google" -googlemodulename = "pbx-google" -defaultstatus = "dnd" -defaultstatusmessage = "PBX online, may lose messages" - -m = Map (modulename, translate("Google Accounts"), - translate("This is where you set up your Google (Talk and Voice) Accounts, in order to start \ - using them for dialing and receiving calls (voice chat and real phone calls). Please \ - make at least one voice call using the Google Talk plugin installable through the \ - GMail interface, and then log out from your account everywhere. Click \"Add\" \ - to add as many accounts as you wish.")) - --- Recreate the config, and restart services after changes are commited to the configuration. -function m.on_after_commit(self) - -- Create a field "name" for each account that identifies the account in the backend. - commit = false - m.uci:foreach(modulename, "gtalk_jabber", - function(s1) - if s1.username ~= nil then - name=string.gsub(s1.username, "%W", "_") - if s1.name ~= name then - m.uci:set(modulename, s1['.name'], "name", name) - commit = true - end - end - end) - if commit == true then m.uci:commit(modulename) end - - luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null") - luci.sys.call("/etc/init.d/asterisk restart 1\>/dev/null 2\>/dev/null") -end - ------------------------------------------------------------------------------ -s = m:section(TypedSection, "gtalk_jabber", translate("Google Voice/Talk Accounts")) -s.anonymous = true -s.addremove = true - -s:option(Value, "username", translate("Email")) - -pwd = s:option(Value, "secret", translate("Password"), - translate("When your password is saved, it disappears from this field and is not displayed \ - for your protection. The previously saved password will be changed only when you \ - enter a value different from the saved one.")) -pwd.password = true -pwd.rmempty = false - --- We skip reading off the saved value and return nothing. -function pwd.cfgvalue(self, section) - return "" -end - --- We check the entered value against the saved one, and only write if the entered value is --- something other than the empty string, and it differes from the saved value. -function pwd.write(self, section, value) - local orig_pwd = m:get(section, self.option) - if value and #value > 0 and orig_pwd ~= value then - Value.write(self, section, value) - end -end - - -p = s:option(ListValue, "register", - translate("Enable Incoming Calls (set Status below)"), - translate("When somebody starts voice chat with your GTalk account or calls the GVoice, \ - number (if you have Google Voice), the call will be forwarded to any users \ - that are online (registered using a SIP device or softphone) and permitted to \ - receive the call. If you have Google Voice, you must go to your GVoice settings and \ - forward calls to Google chat in order to actually receive calls made to your \ - GVoice number. If you have trouble receiving calls from GVoice, experiment \ - with the Call Screening option in your GVoice Settings. Finally, make sure no other \ - client is online with this account (browser in gmail, mobile/desktop Google Talk \ - App) as it may interfere.")) -p:value("yes", translate("Yes")) -p:value("no", translate("No")) -p.default = "yes" - -p = s:option(ListValue, "make_outgoing_calls", translate("Enable Outgoing Calls"), - translate("Use this account to make outgoing calls as configured in the \"Call Routing\" section.")) -p:value("yes", translate("Yes")) -p:value("no", translate("No")) -p.default = "yes" - -st = s:option(ListValue, "status", translate("Google Talk Status")) -st:depends("register", "yes") -st:value("dnd", translate("Do Not Disturb")) -st:value("away", translate("Away")) -st:value("available", translate("Available")) -st.default = defaultstatus - -stm = s:option(Value, "statusmessage", translate("Google Talk Status Message"), - translate("Avoid using anything but alpha-numeric characters, space, comma, and period.")) -stm:depends("register", "yes") -stm.default = defaultstatusmessage - -return m diff --git a/applications/luci-app-pbx/luasrc/model/cbi/pbx-users.lua b/applications/luci-app-pbx/luasrc/model/cbi/pbx-users.lua deleted file mode 100644 index c7c8b4d8bb..0000000000 --- a/applications/luci-app-pbx/luasrc/model/cbi/pbx-users.lua +++ /dev/null @@ -1,133 +0,0 @@ ---[[ - Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com> - - This file is part of luci-pbx. - - luci-pbx is free software: you can redistribute it and/or modify - it under the terms of the GNU General Public License as published by - the Free Software Foundation, either version 3 of the License, or - (at your option) any later version. - - luci-pbx is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License for more details. - - You should have received a copy of the GNU General Public License - along with luci-pbx. If not, see <http://www.gnu.org/licenses/>. -]]-- - -if nixio.fs.access("/etc/init.d/asterisk") then - server = "asterisk" -elseif nixio.fs.access("/etc/init.d/freeswitch") then - server = "freeswitch" -else - server = "" -end - -modulename = "pbx-users" -modulenamecalls = "pbx-calls" -modulenameadvanced = "pbx-advanced" - - -m = Map (modulename, translate("User Accounts"), - translate("Here you must configure at least one SIP account, that you \ - will use to register with this service. Use this account either in an Analog Telephony \ - Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid on your \ - smartphone, or Ekiga, Linphone, or X-Lite on your computer. By default, all SIP accounts \ - will ring simultaneously if a call is made to one of your VoIP provider accounts or GV \ - numbers.")) - --- Recreate the config, and restart services after changes are commited to the configuration. -function m.on_after_commit(self) - luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null") - luci.sys.call("/etc/init.d/" .. server .. " restart 1\>/dev/null 2\>/dev/null") -end - -externhost = m.uci:get(modulenameadvanced, "advanced", "externhost") -bindport = m.uci:get(modulenameadvanced, "advanced", "bindport") -ipaddr = m.uci:get("network", "lan", "ipaddr") - ------------------------------------------------------------------------------ -s = m:section(NamedSection, "server", "user", translate("Server Setting")) -s.anonymous = true - -if ipaddr == nil or ipaddr == "" then - ipaddr = "(IP address not static)" -end - -if bindport ~= nil then - just_ipaddr = ipaddr - ipaddr = ipaddr .. ":" .. bindport -end - -s:option(DummyValue, "ipaddr", translate("Server Setting for Local SIP Devices"), - translate("Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices you will \ - use ONLY locally and never from a remote location.")).default = ipaddr - -if externhost ~= nil then - if bindport ~= nil then - just_externhost = externhost - externhost = externhost .. ":" .. bindport - end - s:option(DummyValue, "externhost", translate("Server Setting for Remote SIP Devices"), - translate("Enter this hostname (or hostname:port) in the Server/Registrar setting of SIP \ - devices you will use from a remote location (they will work locally too).") - ).default = externhost -end - -if bindport ~= nil then - s:option(DummyValue, "bindport", translate("Port Setting for SIP Devices"), - translatef("If setting Server/Registrar to %s or %s does not work for you, try setting \ - it to %s or %s and entering this port number in a separate field that specifies the \ - Server/Registrar port number. Beware that some devices have a confusing \ - setting that sets the port where SIP requests originate from on the SIP \ - device itself (the bind port). The port specified on this page is NOT this bind port \ - but the port this service listens on.", - ipaddr, externhost, just_ipaddr, just_externhost)).default = bindport -end - ------------------------------------------------------------------------------ -s = m:section(TypedSection, "local_user", translate("SIP Device/Softphone Accounts")) -s.anonymous = true -s.addremove = true - -s:option(Value, "fullname", translate("Full Name"), - translate("You can specify a real name to show up in the Caller ID here.")) - -du = s:option(Value, "defaultuser", translate("User Name"), - translate("Use (four to five digit) numeric user name if you are connecting normal telephones \ - with ATAs to this system (so they can dial user names).")) -du.datatype = "uciname" - -pwd = s:option(Value, "secret", translate("Password"), - translate("Your password disappears when saved for your protection. It will be changed \ - only when you enter a value different from the saved one.")) -pwd.password = true -pwd.rmempty = false - --- We skip reading off the saved value and return nothing. -function pwd.cfgvalue(self, section) - return "" -end - --- We check the entered value against the saved one, and only write if the entered value is --- something other than the empty string, and it differes from the saved value. -function pwd.write(self, section, value) - local orig_pwd = m:get(section, self.option) - if value and #value > 0 and orig_pwd ~= value then - Value.write(self, section, value) - end -end - -p = s:option(ListValue, "ring", translate("Receives Incoming Calls")) -p:value("yes", translate("Yes")) -p:value("no", translate("No")) -p.default = "yes" - -p = s:option(ListValue, "can_call", translate("Makes Outgoing Calls")) -p:value("yes", translate("Yes")) -p:value("no", translate("No")) -p.default = "yes" - -return m diff --git a/applications/luci-app-pbx/luasrc/model/cbi/pbx-voip.lua b/applications/luci-app-pbx/luasrc/model/cbi/pbx-voip.lua deleted file mode 100644 index 9b46202855..0000000000 --- a/applications/luci-app-pbx/luasrc/model/cbi/pbx-voip.lua +++ /dev/null @@ -1,116 +0,0 @@ ---[[ - Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com> - - This file is part of luci-pbx. - - luci-pbx is free software: you can redistribute it and/or modify - it under the terms of the GNU General Public License as published by - the Free Software Foundation, either version 3 of the License, or - (at your option) any later version. - - luci-pbx is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License for more details. - - You should have received a copy of the GNU General Public License - along with luci-pbx. If not, see <http://www.gnu.org/licenses/>. -]]-- - -if nixio.fs.access("/etc/init.d/asterisk") then - server = "asterisk" -elseif nixio.fs.access("/etc/init.d/freeswitch") then - server = "freeswitch" -else - server = "" -end - -modulename = "pbx-voip" - -m = Map (modulename, translate("SIP Accounts"), - translate("This is where you set up your SIP (VoIP) accounts ts like Sipgate, SipSorcery, \ - the popular Betamax providers, and any other providers with SIP settings in order to start \ - using them for dialing and receiving calls (SIP uri and real phone calls). Click \"Add\" to \ - add as many accounts as you wish.")) - --- Recreate the config, and restart services after changes are commited to the configuration. -function m.on_after_commit(self) - commit = false - -- Create a field "name" for each account that identifies the account in the backend. - m.uci:foreach(modulename, "voip_provider", - function(s1) - if s1.defaultuser ~= nil and s1.host ~= nil then - name=string.gsub(s1.defaultuser.."_"..s1.host, "%W", "_") - if s1.name ~= name then - m.uci:set(modulename, s1['.name'], "name", name) - commit = true - end - end - end) - if commit == true then m.uci:commit(modulename) end - - luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null") - luci.sys.call("/etc/init.d/" .. server .. " restart 1\>/dev/null 2\>/dev/null") -end - ------------------------------------------------------------------------------ -s = m:section(TypedSection, "voip_provider", translate("SIP Provider Accounts")) -s.anonymous = true -s.addremove = true - -s:option(Value, "defaultuser", translate("User Name")) -pwd = s:option(Value, "secret", translate("Password"), - translate("When your password is saved, it disappears from this field and is not displayed \ - for your protection. The previously saved password will be changed only when you \ - enter a value different from the saved one.")) - - - -pwd.password = true -pwd.rmempty = false - --- We skip reading off the saved value and return nothing. -function pwd.cfgvalue(self, section) - return "" -end - --- We check the entered value against the saved one, and only write if the entered value is --- something other than the empty string, and it differes from the saved value. -function pwd.write(self, section, value) - local orig_pwd = m:get(section, self.option) - if value and #value > 0 and orig_pwd ~= value then - Value.write(self, section, value) - end -end - -h = s:option(Value, "host", translate("SIP Server/Registrar")) -h.datatype = "host(0)" - -p = s:option(ListValue, "register", translate("Enable Incoming Calls (Register via SIP)"), - translate("This option should be set to \"Yes\" if you have a DID \(real telephone number\) \ - associated with this SIP account or want to receive SIP uri calls through this \ - provider.")) -p:value("yes", translate("Yes")) -p:value("no", translate("No")) -p.default = "yes" - -p = s:option(ListValue, "make_outgoing_calls", translate("Enable Outgoing Calls"), - translate("Use this account to make outgoing calls.")) -p:value("yes", translate("Yes")) -p:value("no", translate("No")) -p.default = "yes" - -from = s:option(Value, "fromdomain", - translate("SIP Realm (needed by some providers)")) -from.optional = true -from.datatype = "host(0)" - -port = s:option(Value, "port", translate("SIP Server/Registrar Port")) -port.optional = true -port.datatype = "port" - -op = s:option(Value, "outboundproxy", translate("Outbound Proxy")) -op.optional = true -op.datatype = "host(0)" - -return m diff --git a/applications/luci-app-pbx/luasrc/model/cbi/pbx.lua b/applications/luci-app-pbx/luasrc/model/cbi/pbx.lua deleted file mode 100644 index 4c5fcbdecd..0000000000 --- a/applications/luci-app-pbx/luasrc/model/cbi/pbx.lua +++ /dev/null @@ -1,115 +0,0 @@ ---[[ - Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com> - - This file is part of luci-pbx. - - luci-pbx is free software: you can redistribute it and/or modify - it under the terms of the GNU General Public License as published by - the Free Software Foundation, either version 3 of the License, or - (at your option) any later version. - - luci-pbx is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License for more details. - - You should have received a copy of the GNU General Public License - along with luci-pbx. If not, see <http://www.gnu.org/licenses/>. -]]-- - -modulename = "pbx" - - -if nixio.fs.access("/etc/init.d/asterisk") then - server = "asterisk" -elseif nixio.fs.access("/etc/init.d/freeswitch") then - server = "freeswitch" -else - server = "" -end - - --- Returns formatted output of string containing only the words at the indices --- specified in the table "indices". -function format_indices(string, indices) - if indices == nil then - return "Error: No indices to format specified.\n" - end - - -- Split input into separate lines. - lines = luci.util.split(luci.util.trim(string), "\n") - - -- Split lines into separate words. - splitlines = {} - for lpos,line in ipairs(lines) do - splitlines[lpos] = luci.util.split(luci.util.trim(line), "%s+", nil, true) - end - - -- For each split line, if the word at all indices specified - -- to be formatted are not null, add the formatted line to the - -- gathered output. - output = "" - for lpos,splitline in ipairs(splitlines) do - loutput = "" - for ipos,index in ipairs(indices) do - if splitline[index] ~= nil then - loutput = loutput .. string.format("%-40s", splitline[index]) - else - loutput = nil - break - end - end - - if loutput ~= nil then - output = output .. loutput .. "\n" - end - end - return output -end - - -m = Map (modulename, translate("PBX Main Page"), - translate("This configuration page allows you to configure a phone system (PBX) service which \ - permits making phone calls through multiple Google and SIP (like Sipgate, \ - SipSorcery, and Betamax) accounts and sharing them among many SIP devices. \ - Note that Google accounts, SIP accounts, and local user accounts are configured in the \ - \"Google Accounts\", \"SIP Accounts\", and \"User Accounts\" sub-sections. \ - You must add at least one User Account to this PBX, and then configure a SIP device or \ - softphone to use the account, in order to make and receive calls with your Google/SIP \ - accounts. Configuring multiple users will allow you to make free calls between all users, \ - and share the configured Google and SIP accounts. If you have more than one Google and SIP \ - accounts set up, you should probably configure how calls to and from them are routed in \ - the \"Call Routing\" page. If you're interested in using your own PBX from anywhere in the \ - world, then visit the \"Remote Usage\" section in the \"Advanced Settings\" page.")) - ------------------------------------------------------------------------------------------ -s = m:section(NamedSection, "connection_status", "main", - translate("PBX Service Status")) -s.anonymous = true - -s:option (DummyValue, "status", translate("Service Status")) - -sts = s:option(DummyValue, "_sts") -sts.template = "cbi/tvalue" -sts.rows = 20 - -function sts.cfgvalue(self, section) - - if server == "asterisk" then - regs = luci.sys.exec("asterisk -rx 'sip show registry' | sed 's/peer-//'") - jabs = luci.sys.exec("asterisk -rx 'jabber show connections' | grep onnected") - usrs = luci.sys.exec("asterisk -rx 'sip show users'") - chan = luci.sys.exec("asterisk -rx 'core show channels'") - - return format_indices(regs, {1, 5}) .. - format_indices(jabs, {2, 4}) .. "\n" .. - format_indices(usrs, {1} ) .. "\n" .. chan - - elseif server == "freeswitch" then - return "Freeswitch is not supported yet.\n" - else - return "Neither Asterisk nor FreeSwitch discovered, please install Asterisk, as Freeswitch is not supported yet.\n" - end -end - -return m diff --git a/applications/luci-app-pbx/po/ca/pbx.po b/applications/luci-app-pbx/po/ca/pbx.po deleted file mode 100644 index c8a0a9967e..0000000000 --- a/applications/luci-app-pbx/po/ca/pbx.po +++ /dev/null @@ -1,509 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"PO-Revision-Date: 2014-07-01 05:14+0200\n" -"Last-Translator: Alex <alexhenrie24@gmail.com>\n" -"Language-Team: none\n" -"Language: ca\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=2; plural=(n != 1);\n" -"X-Generator: Pootle 2.0.6\n" - -msgid "Advanced Settings" -msgstr "Ajusts avançats" - -msgid "Available" -msgstr "Disponible" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" -"Eviteu utilitzar res excepte caràcters alfanumèrics, espai, coma, i punt." - -msgid "Away" -msgstr "Fora" - -msgid "Blacklisted Numbers" -msgstr "Nombres prohibits" - -msgid "Call Routing" -msgstr "Encaminament de trucades" - -msgid "Call-back Numbers" -msgstr "Nombres de trucada de tornada" - -msgid "Call-back Provider" -msgstr "Proveïdor de trucada de tornada" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "Copieu i enganxeu llistes grans de nombres aquí." - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" -"Designeu els nombres que es permeten trucar a través d'aquest sistema i els " -"privilegis de qual usuari tindran." - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" -"Designeu els nombres als quals el sistema penjarà i trucarà de tornada, qual " -"proveïdor s'emprarà per a trucar-los, i els privilegis de qual usuari se " -"lis concedirà." - -msgid "Dials numbers unmatched elsewhere" -msgstr "Truca els nombres que no coincideixen d'altra manera" - -msgid "Do Not Disturb" -msgstr "No molestis" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "" - -msgid "Email" -msgstr "" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "Habilita trucades entrants (registreu via SIP)" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "Habilita trucades entrants (establiu l'Estat a baix)" - -msgid "Enable Outgoing Calls" -msgstr "Habilita trucades sortints" - -msgid "Enabled" -msgstr "Habilitat" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "Port SIP extern" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "Nom complet" - -msgid "General Settings" -msgstr "Ajusts generals" - -msgid "Google Accounts" -msgstr "Comptes de Google" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "" - -msgid "Hang-up Delay" -msgstr "Retard de penja" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" -"Quant temps per a esperar abans de penjar. Si el proveïdor que empreu per a " -"trucar automàticament redirigeix al correu de veu, podeu estableix aquest " -"valor a un retard que us permet penjar abans que la teva trucada es " -"redirigeixi i s'us cobri per ella." - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "Trucades entrants" - -msgid "Insert QoS Rules" -msgstr "Insereix regles QoS" - -msgid "Makes Outgoing Calls" -msgstr "Fa trucades sortints" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "NOTA: No hi ha cap compte configurat ni del Google ni de proveïdor SIP." - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" -"NOTA: No hi ha cap compte habilitat ni del Google ni de proveïdor SIP per " -"als trucades entrants." - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" -"NOTA: No hi ha cap compte habilitat ni del Google ni de proveïdor SIP per " -"als trucades sortints." - -msgid "NOTE: There are no local user accounts configured." -msgstr "NOTA: No hi ha cap compte d'usuari local configurat." - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" -"NOTA: No hi ha cap compte d'usuari local habilitat per als trucades " -"sortints." - -msgid "No" -msgstr "No" - -msgid "Number of Seconds to Ring" -msgstr "Nombre de segons a sonar" - -msgid "Outbound Proxy" -msgstr "Servidor intermediari de sortida" - -msgid "Outgoing Calls" -msgstr "Trucades sortints" - -msgid "PBX Main Page" -msgstr "Pàgina principal PBX" - -msgid "PBX Service Status" -msgstr "Estat del servei PBX" - -msgid "PIN" -msgstr "PIN" - -msgid "Password" -msgstr "Contrasenya" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "Ajust de port per als dispositius SIP" - -msgid "Providers Used for Outgoing Calls" -msgstr "Proveïdors utilitzats per als trucades sortints" - -msgid "QoS Settings" -msgstr "Ajusts QoS" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "Rep trucades entrants" - -msgid "Remote Usage" -msgstr "Ús remot" - -msgid "Rings users enabled for incoming calls" -msgstr "Truca als usuaris habilitats per a rebre trucades" - -msgid "SIP Accounts" -msgstr "Comptes SIP" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "Comptes de proveïdor SIP" - -msgid "SIP Realm (needed by some providers)" -msgstr "Regne SIP (necessitat per alguns proveïdors)" - -msgid "SIP Server/Registrar" -msgstr "Servidor/Registrador SIP" - -msgid "SIP Server/Registrar Port" -msgstr "Port del Servidor/Registrador SIP" - -msgid "Server Setting" -msgstr "Ajust de servidor" - -msgid "Server Setting for Local SIP Devices" -msgstr "Ajust de servidor pels dispositius SIP locals" - -msgid "Server Setting for Remote SIP Devices" -msgstr "Ajust de servidor pels dispositius SIP remots" - -msgid "Service Status" -msgstr "Estat de servei" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" -"Estableix el nombre de segons per a sonar als usuaris abans de penjar o anar " -"al correu de veu, si el correu de veu està instal·lat i habilitat." - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "Llista de nombres prohibits separats per espai" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" -"Especifiqueu els nombres individualment aquí. Premeu Enter per afegir més " -"nombres." - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" -"Utilitza aquest compte per fer trucades sortints com configurat en la secció " -"\"Encaminament de trucades\"." - -msgid "Use this account to make outgoing calls." -msgstr "Utilitza aquest compte per fer trucades sortints." - -msgid "User Accounts" -msgstr "Comptes d'usuari" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "Nom d'usuari" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "Sí" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" diff --git a/applications/luci-app-pbx/po/cs/pbx.po b/applications/luci-app-pbx/po/cs/pbx.po deleted file mode 100644 index 8b69ef15d8..0000000000 --- a/applications/luci-app-pbx/po/cs/pbx.po +++ /dev/null @@ -1,487 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"PO-Revision-Date: 2014-07-12 20:19+0200\n" -"Last-Translator: koli <lukas.koluch@gmail.com>\n" -"Language-Team: none\n" -"Language: cs\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=3; plural=(n==1) ? 0 : (n>=2 && n<=4) ? 1 : 2;\n" -"X-Generator: Pootle 2.0.6\n" - -msgid "Advanced Settings" -msgstr "Pokročilé nastavení" - -msgid "Available" -msgstr "Dostupné" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" - -msgid "Away" -msgstr "Pryč" - -msgid "Blacklisted Numbers" -msgstr "" - -msgid "Call Routing" -msgstr "" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "Nevyrušovat" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "" - -msgid "Email" -msgstr "Email" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "Povolit příchozí hovory (Registrace přes SIP)" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "" - -msgid "Enable Outgoing Calls" -msgstr "Povolit odchozí hovory" - -msgid "Enabled" -msgstr "Povoleno" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "Externí SIP port" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "Celé jméno (jméno a příjmení)" - -msgid "General Settings" -msgstr "Obecné nastavení" - -msgid "Google Accounts" -msgstr "Google účty" - -msgid "Google Talk Status" -msgstr "Stav Google Talk" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "Google Voice/Talk účty" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "Příchozí volání" - -msgid "Insert QoS Rules" -msgstr "Vložte QoS pravidla" - -msgid "Makes Outgoing Calls" -msgstr "" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "Ne" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "Odchozí volání" - -msgid "PBX Main Page" -msgstr "Hlavní stránka PBX" - -msgid "PBX Service Status" -msgstr "Stav PBX služby" - -msgid "PIN" -msgstr "PIN" - -msgid "Password" -msgstr "Heslo" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "Nastavení QoS" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "SIP účty" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "Stav služby" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "Uživatelské účty" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "Uživatelské jméno" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "Ano" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" diff --git a/applications/luci-app-pbx/po/de/pbx.po b/applications/luci-app-pbx/po/de/pbx.po deleted file mode 100644 index 3bc4bd428e..0000000000 --- a/applications/luci-app-pbx/po/de/pbx.po +++ /dev/null @@ -1,699 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"PO-Revision-Date: 2013-01-30 18:17+0200\n" -"Last-Translator: DAC324 <gerd_roethig@web.de>\n" -"Language-Team: none\n" -"Language: de\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=2; plural=(n != 1);\n" -"X-Generator: Pootle 2.0.6\n" - -msgid "Advanced Settings" -msgstr "Erweiterte Einstellungen" - -msgid "Available" -msgstr "Verfügbar" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "Nur alphanumerische Zeichen, Komma, Punkt und Leerzeichen verwenden" - -msgid "Away" -msgstr "Abwesend" - -msgid "Blacklisted Numbers" -msgstr "Nicht erlaubte Nummern (Blacklist)" - -msgid "Call Routing" -msgstr "Anrufweiterleitung" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "Durchwahl Nummern" - -msgid "Copy-paste large lists of numbers here." -msgstr "Hier können per Copy & Paste größere Nummernlisten eingefügt werden." - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "Wählt Nummern an, für die es keine andere Übereinstimmung gibt" - -msgid "Do Not Disturb" -msgstr "Beschäftigt" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "Domäne/IP-Adresse/Dynamische Domäne" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "Dynamische Liste nicht erlaubter Nummern (Dynamische Blacklist)" - -msgid "Email" -msgstr "E-Mail" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "Eingehende Anrufe akzeptieren (registrieren via SIP)" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "Eingehende Anrufe akzeptieren (Status unten einstellen)" - -msgid "Enable Outgoing Calls" -msgstr "Ausgehende Anrufe aktivieren" - -msgid "Enabled" -msgstr "Aktiv" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" -"Geben Sie Telefonnummern ein, von denen Anrufe automatisch zurückgewiesen " -"werden sollen. Sie sollten die Ländervorwahl und alle führenden Nullen " -"weglassen, aber experimentieren Sie ruhig, damit Sie auch wirklich alle " -"Nummern blockieren, die blockiert werden sollen." - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" -"Geben SIe diese IP (oder IP:Port) in der Server-/Registrar-Einstellung der " -"SIP-Geräte an, die Sie NUR local und niemals von einem entfernten Ort " -"einsetzen werden." - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" -"Geben SIe diese IP (oder IP:Port) in der Server-/Registrar-Einstellung der " -"SIP-Geräte an, die Sie von einem entfernten Ort einsetzen werden (sie " -"funktionieren auch lokal)." - -msgid "External SIP Port" -msgstr "Externer SIP Port" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" -"Hier können Sie für jeden Dienstanbieter, der für eingehende Anrufe " -"eingerichtet ist, festlegen, welche Nutzer ein Klingelzeichen bei " -"eingehenden Anrufen erhalten. Ist die Liste leer, klingelt es bei allen " -"Nutzern, die eingehende Anrufe empfangen dürfen. Ungültige Benutzernamen " -"werden ohne Fehlermeldung zurückgewiesen. Außerdem überschreibt der Eintrag " -"eines Benutzernamens an dieser Stelle die evtl. vorhandene Einstellung für " -"diesen Benutzer, keine eingehenden Anrufe zu erhalten. Auf diese Weise kann " -"eingestellt werden, dass die Nutzer nur bei bestimmten Dienstanbietern ein " -"Klingelzeichen erhalten. Einträge in dieser Liste können entweder durch " -"Leerzeichen getrennt oder als ein Eintrag pro Zeile (Eingabetaste nach jedem " -"Eintrag) eingegeben werden." - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" -"Hier können Sie für jeden Benutzer, der für abgehende Anrufe eingerichtet " -"ist, festlegen, welche Dienstanbieter verwendet werden dürfen. In der " -"Voreinstellung dürfen alle Benutzer auch alle Dienstanbieter verwenden. Um " -"in der Liste unten aufzutauchen, sollte dem Benutzer auf der Seite " -"\"Benutzerkonten\" erlaubt werden, abgehende Anrufe machen zu dürfen. Geben " -"Sie VoIP-Dienstanbieter im Format Benutzername@Servername an, wie bereits " -"oben unter \"Abgehende Anrufe\". Am einfachsten kopieren Sie die " -"Dienstanbieter von dort und fügen sie hier wieder ein. Ungültige Einträge, " -"einschließlich nicht für abgehende Anrufe zugelassene Dienstanbieter, werden " -"ohne Fehlermeldung zurückgewiesen. Einträge in dieser Liste können entweder " -"durch Leerzeichen getrennt und/oder als ein Eintrag pro Zeile (Eingabetaste " -"nach jedem Eintrag) eingegeben werden." - -msgid "Full Name" -msgstr "Vollständiger Name" - -msgid "General Settings" -msgstr "Allgemeine Einstellungen" - -msgid "Google Accounts" -msgstr "Google-Konten" - -msgid "Google Talk Status" -msgstr "Status für Google Talk" - -msgid "Google Talk Status Message" -msgstr "Statusbenachrichtigung für Google Talk" - -msgid "Google Voice/Talk Accounts" -msgstr "Google Voice/Talk-Konten" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" -"Hier müssen Sie wenigstens ein SIP-Konto angeben, welches Sie zur Anmeldung " -"an diesen Dienst nutzen. Verwenden Sie dieses Konto entweder in einem " -"Adapter für analoges Telefonieren (ATA) oder einer SIP-Software wie " -"CSipSimple, Linphone, oder Sipdroid auf Ihrem Smartphone, oder Ekiga, " -"Linphone, oder X-Lite auf Ihrem Computer. In der Voreinstellung klingeln " -"alle SIP-Konten gleichzeitig, wenn ein Anruf auf eines Ihrer VoIP-Konten " -"oder Ihre GV-Nummern gemacht wird." - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" -"Wenn EInstellen des Servers/Registrars auf %s oder %s bei Ihnen nicht " -"funktioniert, versuchen Sie die Einstellung %s oder %s und geben Sie die " -"Portnummer in ein separates Feld für Server/Registrat-Portnummer ein. " -"Achtung: Einige Geräte haben eine verwirrende Einstellung, die den Port " -"setzt, von dem die SIP-Anfragen auf dem Gerät selbst herkommen (der Bindungs-" -"Port). Der Port auf dieser Seite meint NICHT diesen Bindungs-Port, sondern " -"den Port, an dem der Dienst lauscht." - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" -"Wenn Sie stotternden oder stark verzögerten Ton während großer Downloads " -"haben, sollten Sie QoS einschalten. QoS priorisiert Verkehr von und zu Ihrem " -"Netzwerk für bestimmte Ports und IP-Adressen mit dem Ergebnis einer besseren " -"Tonübertragung in unserem Fall. Wenn unten eingeschaltet, wird eine QoS-" -"Regel automatisch vom PBX eingerichtet, aber Sie müssen die QoS-" -"Konfigurationsseite (Netzwerk->QoS) aufrufen, um andere kritische QoS-" -"Einstellungen wie Upload-und Download-Geschwindigkeit vorzunehmen." - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" -"Wenn Sie mehr als ein Konto für abgehende Anrufe haben, sollten Sie eine " -"Liste von Telefonnummern/Vorwahlen in den folgenden Feldern für jeden " -"aufgeführten Dienstanbieter eintragen. Ungültige Vorwahlen werden ohne " -"Fehlermeldung entfernt, nur 0-9, X, Z, N, #, *, und + sind gültige Zeichen. " -"Der Buchstabe X entspricht 0-9, Z entrpricht 1-9, N entspricht 2-9. Zum " -"Beispiel können Sie 49 eingeben, um Anrufe nach Deutschland über einen " -"Dienstanbieter zu tätigen. Für Anrufe nach Nordamerika geben Sie 1NXXNXXXXXX " -"an. Unterstützt ein Dienstanbieter Ortsgespräche, wie im Gebiet 646 von New " -"York, geben Sie 646NXXXXXX für diesen Anbieter ein. Ein Konto sollte eine " -"leere Liste behalten, damit Sie darüber standardmäßig Anrufe tätigen können, " -"wenn keine der Vorwahlen für die anderen Anbieter übereinstimmt. Das System " -"ersetzt eine leere Liste automatisch mit dem Eintrag, dass dieser Anbieter " -"alle Vorwahlen unterstützt, die von den anderen Anbietern nicht unterstützt " -"werden. Seien Sie so spezifisch wie möglich (1NXXNXXXXXX ist besser als 1). " -"Bitte beachten Sie, dass alle internationalen Vorwahl-Codes (wie 00, 011, " -"010, 0011) verworfen werden. Einträge können durch Leezeichen getrennt und/" -"oder einzeln pro Zeile (Abschließen mit Eingabe-Taste) eingegeben werden." - -msgid "Incoming Calls" -msgstr "Eingehende Anrufe" - -msgid "Insert QoS Rules" -msgstr "QoS-Regeln einfügen" - -msgid "Makes Outgoing Calls" -msgstr "Macht ausgehende Anrufe" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" -"ACHTUNG: Es sind keine Konten für Google oder einen SIP-Dienstanbieter " -"eingerichtet." - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" -"ACHTUNG: Es sind keine Konten für Google oder einen SIP-Dienstanbieter für " -"eingehende Anrufe eingerichtet." - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" -"ACHTUNG: Es sind keine Konten für Google oder einen SIP-Dienstanbieter für " -"abgehende Anrufe eingerichtet." - -msgid "NOTE: There are no local user accounts configured." -msgstr "ACHTUNG: Es sind keine lokalen Benutzerkonten eingerichtet." - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" -"ACHTUNG: Es sind keine lokalen Benutzerkonten für abgehende Anrufe " -"eingerichtet." - -msgid "No" -msgstr "Nein" - -msgid "Number of Seconds to Ring" -msgstr "Dauer des Klingelns in Sekunden" - -msgid "Outbound Proxy" -msgstr "Proxy für ausgehende Verbindungen" - -msgid "Outgoing Calls" -msgstr "Abgehende Anrufe" - -msgid "PBX Main Page" -msgstr "PBX-Hauptseite" - -msgid "PBX Service Status" -msgstr "PBX-Dienststatus" - -msgid "PIN" -msgstr "PIN" - -msgid "Password" -msgstr "Passwort" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "Port-Einstellung für SIP-Geräte" - -msgid "Providers Used for Outgoing Calls" -msgstr "Provider für abgehende Anrufe" - -msgid "QoS Settings" -msgstr "QoS Einstellungen" - -msgid "RTP Port Range End" -msgstr "Ende des RTP-Port-Bereichs" - -msgid "RTP Port Range Start" -msgstr "Anfang des RTP-Port-Bereichs" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" -"RTP-Verkehr überträgt die aktuellen Sprachpakete. Dies ist der Anfang des " -"Port-Bereichs, der für die Einrichtung der RTP-Verbindung verwendet wird. " -"Normalerweise kann hier die Voreinstellung belassen werden." - -msgid "Receives Incoming Calls" -msgstr "Empfängt eingehende Anrufe" - -msgid "Remote Usage" -msgstr "Benutzung aus der Ferne" - -msgid "Rings users enabled for incoming calls" -msgstr "Für eingehende Anrufe freigeschaltete Nutzer erhalten Klingelzeichen" - -msgid "SIP Accounts" -msgstr "SIP-Konten" - -msgid "SIP Device/Softphone Accounts" -msgstr "SIP-Geräte-/Softphone-Konten" - -msgid "SIP Provider Accounts" -msgstr "SIP-Dienstanbieter-Konten" - -msgid "SIP Realm (needed by some providers)" -msgstr "SIP-Bereich (von manchen Dienstanbietern benötigt)" - -msgid "SIP Server/Registrar" -msgstr "SIP-Server/Registrar" - -msgid "SIP Server/Registrar Port" -msgstr "SIP-Server/Registrar Port" - -msgid "Server Setting" -msgstr "Servereinstellung" - -msgid "Server Setting for Local SIP Devices" -msgstr "Servereinstellung für lokale SIP-Geräte" - -msgid "Server Setting for Remote SIP Devices" -msgstr "Servereinstellung für entfernte SIP-Geräte" - -msgid "Service Status" -msgstr "Dienst-Status" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" -"Stellen Sie ein (in Sekunden), wie lange es bei den Benutzern klingeln soll, " -"bevor aufgelegt oder zur Voicemail (falls installiert und aktiv) " -"übergegangen wird. " - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "Mit Leerzeichen unterteilte Liste gesperrter Nummern" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" -"Geben Sie die Nummern hier einzeln an. Drücken Sie Eingabe, um weitere " -"Nummern hinzuzufügen." - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" -"Die oben angegebene(n) Nummer(n) können für ausgehende Anrufe mit den " -"Dienstanbietern dieses Nutzers verwendet werden. Ungültige Benutzernamen, " -"einschließlich Nutzer, die nicht für ausgehende Anrufe freigeschaltet sind, " -"werden ohne Fehlermeldung verworfen. Bitte überprüfen Sie deshalb, ob der " -"Eintrag akzeptiert wurde." - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" -"Diese Konfigurationsseite erlaubt Ihnen die Einrichtung eines " -"Telefonsystemdienstes (PBX), der Anrufe über mehrere Google- und SIP-Konten " -"(wie Sipgate, SipSorcery und Betamax) erlaubt. Sie können diese Konten für " -"viele SIP-Geräte verwenden. Beachten Sie, dass Google-, SIP- und lokale " -"Benutzer-Konten in den Abschnitten \"Google-Konten\", \"SIP-Konten\" und " -"\"Benutzerkonten\" eingerichtet werden. Sie müssen mindestens ein " -"Benutzerkonto für diesen PBX vorsehen und dann ein SIP-Gerät oder Softphone " -"für die Benutzung dieses Kontos einrichten, damit Sie Anrufe mit Ihren " -"Google-/SIP-Konten tätigen oder empfangen können. Wenn Sie mehr als ein " -"Google- / SIP-Konto eingerichtet haben, sollten Sie auf der Seite " -"\"Anrufweiterleitung\" einrichten, wie diese Anrufe behandelt werden. Wenn " -"Sie Ihr PBX von irgendwo auf der Welt nutzen wollen, schauen Sie auf den " -"Abschnitt \"Benutzung aus der Ferne\" auf der Seite \"Erweiterte " -"Einstellungen\". " - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" -"Dies ist der Name, den der VoIP-Server verwenden wird, um sich selbst bei " -"der Registrierung beim VoIP-Dienstanbieter zu identifizieren. Einige " -"Anbieter verlangen, dass dies ein spezieller Begriff ist, der einem Hardware-" -"SIP-Gerät entspricht." - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" -"Hier geben Sie an, welche Google-/SIP-Konten für welche Ländervorwahlen " -"benutzt werden sollen, welche Nutzer welche Konten verwenden dürfen, wie " -"Anrufe weitergeleitet werden, welche Nummern mit Password in diesen PBX " -"kommen, und welche Nummern ausgeschlossen werden." - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" -"Hier stellen Sie Ihre Google (Talk und Voice) Konten ein, um sie für " -"abgehende und ankommende Anrufe nutzen zu können (Voice Chat und Telefon-" -"Anrufe). Bitte tätigen Sie wenigstens einen Sprach-Anruf mit dem Google-Talk-" -"Plugin, das über das GMail-Interface zu installieren ist, und melden Sie " -"sich dann überall aus Ihrem Konto ab. Klicken Sie auf \"Hinzufügen\" um so " -"viele Konten hinzuzufügen, wie Sie wollen." - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" -"Hier stellen Sie Ihre SIP (VoIP) Konten, wie Sipgate, SipSorcery, die " -"populären Betamax-Anbieter, und alle anderen Anbieter mit SIP-Einstellungen " -"ein, um sie für abgehende und ankommende Anrufe nutzen zu können (SIP uri " -"und Telefon-Anrufe). Klicken Sie auf \"Hinzufügen\" um so viele Konten " -"hinzuzufügen, wie Sie wollen." - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" -"Diese Option sollte auf \"Ja\" gesetzt werden, wenn Sie eine DID (reale " -"Telefonnummer) haben, die mit diesem SIP-Konto verknüpft ist, oder wenn Sie " -"SIP-Anrufe über diesen Anbieter empfangen wollen." - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" -"Dieser Abschnitt enthält Einstellungen, die unter normalen Umständen nicht " -"geändert werden müssen. Zusätzlich konnen Sie hier Ihr System für die " -"Verwendung mit entfernten SIP-Geräten einrichten und Probleme bei der " -"Tonqualität beheben, indem Sie die Festlegung von QoS-Regeln aktivieren." - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" -"Verwenden Sie eine vier- bis fünfstellige Nummer als Benutzernamen, wenn Sie " -"normale Telefone mit ATA an dieses System anschließen (damit diese Namen " -"über deren Zifferntastatur eingegeben werden können)." - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" -"Dieses Konto für abgehende Anrufe verwenden, wie im Abschnitt " -"\"Anrufweiterleitung\" eingestellt." - -msgid "Use this account to make outgoing calls." -msgstr "Dieses Konto für abgehende Anrufe verwenden." - -msgid "User Accounts" -msgstr "Benutzerkonten" - -msgid "User Agent String" -msgstr "Benutzeridentifikation (User Agent)" - -msgid "User Name" -msgstr "Benutzername" - -msgid "Uses providers enabled for outgoing calls" -msgstr "Verwendet für abgehende Anrufe eingerichtete Dienstanbieter" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" -"Wenn jemand einen Voice-Chat mit Ihrem GTalk-Konto oder die GVoice-Nummer " -"(falls Sie Google Voice haben) anruft, wird der Anruf an jeden Benutzer " -"weiter geleitet, der Online ist (mit SIP-Gerät oder Softphone) und den Anruf " -"empfangen darf. Wenn Sie Google Voice haben, müssen Sie in Ihre GVoice-" -"Einstellungen gehen und Anrufe zu Google Chat weiter leiten, damit Sie " -"Anrufe auf Ihre GVoice-Nummer empfangen können. Bei Problemen mit dem " -"Empfang von Anrufen über GVoice, experimentieren Sie mit der Option " -"\"Anrufprüfung\" in den GVoice-Einstellungen. Stellen Sie schließlich " -"sicher, dass kein anderer Client mit diesem Konto Online ist (z.B. Browser " -"in GMail, Google Talk App mobil oder auf PC), denn das könnte Einfluss haben." - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" -"Wenn Ihr Passwort gespeichert wird, verschwindet es aus diesem Feld und wird " -"zu Ihrem Schutz nicht angezeigt. Ein vorher gespeichertes Passwort wird nur " -"geändert, wenn Sie ein geändertes Passwort eingeben." - -msgid "Yes" -msgstr "Ja" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" -"Sie können hier einen Klarnamen angeben, der als Name des Anrufers erscheint." - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" -"Sie können Ihre SIP-Geräte/Softphones mit diesem System auch von einem " -"entfernten Ort aus benutzen, so lange Ihnen Ihr Internet-Dienstanbieter eine " -"öffentliche IP-Adresse zuweist. Sie können andere lokale Benutzer kostenlos " -"anrufen (z.B. andere Analog-Telefon-Adapter (ATA)) und Ihre VoIP-Anbieter " -"für Anrufe verwenden, als ob Sie am lokalen PBX angeschlossen wären. Nach " -"der Einrichtung dieses Tabs gehen Sie zu den Benutzereinstellungen zurück " -"und schauen Sie nach den neuen Einstellungen für Server und Port, die Sie an " -"den entfernten SIP-Geräten vornehmen müssen. Bitte beachten Sie, dass Sie " -"NAT/Portweiterleitung auf dem Router/Gateway einrichten müssen, falls dieser " -"PBX nicht auf Ihrem Router/Gateway läuft. Bitte leiten Sie die unten " -"angegebenen Ports (SIP-Port und RTP-Bereich) auf die IP-Adresse dieses PBX " -"weiter." - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" -"Ihre PIN verschwindet beim Speichern aus diesem Feld und wird zu Ihrem " -"Schutz nicht angezeigt. Eine vorher gespeicherte PIN wird nur geändert, wenn " -"Sie eine geänderte PIN eingeben. Sie können die PIN leer lassen, aber denken " -"Sie an die Konsequenzen für die Sicherheit." - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" -"Ihr Passwort verschwindet beim Speichern und wird zu Ihrem Schutz nicht " -"angezeigt. Es wird nur geändert, wenn Sie ein anderes Passwort eingeben." - -#~ msgid "" -#~ "Designate numbers that are allowed to call through this system and which " -#~ "user's privileges it will have." -#~ msgstr "" -#~ "Nummern auswählen, die durch dieses System anrufen können, und deren " -#~ "Benutzerrechte einstellen" - -#~ msgid "" -#~ "Pick a random port number between 6500 and 9500 for the service to listen " -#~ "on. Do not pick the standard 5060, because it is often subject to brute-" -#~ "force attacks. When finished, (1) click \"Save and Apply\", and (2) click " -#~ "the \"Restart VoIP Service\" button above. Finally, (3) look in the \"SIP " -#~ "Device/Softphone Accounts\" section for updated Server and Port settings " -#~ "for your SIP Devices/Softphones." -#~ msgstr "" -#~ "Wählen Sie eine zufällige Portnummer zwischen 6500 und 9000 für den Dienst " -#~ "aus. Nehmen Sie nicht die standardmäßige 5060, weil sie oft attackiert wird. " -#~ "Wenn fertig (1) klicken Sie auf \"Speichern und Anwenden\" und (2) auf \"VoIP-" -#~ "Dienst neu starten\" oben. Schließlich (3) sehen Sie im Abschnitt \"SIP-Geräte" -#~ "/Softphone-Konten\" nach aktualisierten Einstellungen für Ihre SIP-" -#~ "Geräte/Softphones." - -#~ msgid "" -#~ "You can enter your domain name, external IP address, or dynamic domain " -#~ "name here Please keep in mind that if your IP address is dynamic and it " -#~ "changes your configuration will become invalid. Hence, it's recommended " -#~ "to set up Dynamic DNS in this case." -#~ msgstr "" -#~ "Sie können Ihren Domänennamen, externe IP-Adresse, oder dynamischen " -#~ "Domänennamen hier angeben.Bitte beachten Sie, dass Ihre Konfiguration " -#~ "ungältig wird, wenn Sie eine dynamische IP-Adresse besitzen und sich diese " -#~ "ändert. Für diesen Fall wird deshalb die Einrichtung von dnamischem DNS " -#~ "empfohlen." - -#~ msgid "Account Status" -#~ msgstr "Konto-Status" - -#~ msgid "Account Status Message" -#~ msgstr "Konto-Status Meldung" - -#~ msgid "Domain Name/Dynamic Domain Name" -#~ msgstr "DNS Name (auch dynamisch möglich)" diff --git a/applications/luci-app-pbx/po/el/pbx.po b/applications/luci-app-pbx/po/el/pbx.po deleted file mode 100644 index 717e2563b4..0000000000 --- a/applications/luci-app-pbx/po/el/pbx.po +++ /dev/null @@ -1,493 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"PO-Revision-Date: 2012-03-31 15:41+0200\n" -"Last-Translator: Vasilis <acinonyx@openwrt.gr>\n" -"Language-Team: none\n" -"Language: el\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=2; plural=(n != 1);\n" -"X-Generator: Pootle 2.0.4\n" - -msgid "Advanced Settings" -msgstr "" - -msgid "Available" -msgstr "" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" - -msgid "Away" -msgstr "" - -msgid "Blacklisted Numbers" -msgstr "" - -msgid "Call Routing" -msgstr "" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "Μην Ενοχλείτε" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "" - -msgid "Email" -msgstr "" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "" - -msgid "Enable Outgoing Calls" -msgstr "" - -msgid "Enabled" -msgstr "Ενεργοποιημένο" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "Πλήρες Όνομα" - -msgid "General Settings" -msgstr "Γενικές Ρυθμίσεις" - -msgid "Google Accounts" -msgstr "Λογαριασμοί Google" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "Λογαριασμοί Google Voice/Talk" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "Εισερχόμενες Κλήσεις" - -msgid "Insert QoS Rules" -msgstr "" - -msgid "Makes Outgoing Calls" -msgstr "" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "Όχι" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "Εξερχόμενες Κλήσεις" - -msgid "PBX Main Page" -msgstr "" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "PIN" - -msgid "Password" -msgstr "Κωδικός πρόσβασης" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "Λογαριασμοί SIP" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -#~ msgid "Account Status" -#~ msgstr "Κατάσταση Λογαριασμού" - -#~ msgid "Account Status Message" -#~ msgstr "Μήνυμα Κατάστασης Λογαριασμού" diff --git a/applications/luci-app-pbx/po/en/pbx.po b/applications/luci-app-pbx/po/en/pbx.po deleted file mode 100644 index 8b995e1a39..0000000000 --- a/applications/luci-app-pbx/po/en/pbx.po +++ /dev/null @@ -1,502 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"Last-Translator: Automatically generated\n" -"Language-Team: none\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=2; plural=(n != 1);\n" - -msgid "Advanced Settings" -msgstr "Advanced Settings" - -msgid "Available" -msgstr "Available" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." - -msgid "Away" -msgstr "Away" - -msgid "Blacklisted Numbers" -msgstr "Blacklisted Numbers" - -msgid "Call Routing" -msgstr "Call Routing" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "Call-through Numbers" - -msgid "Copy-paste large lists of numbers here." -msgstr "Copy-paste large lists of numbers here." - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "Do Not Disturb" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "Dynamic List of Blacklisted Numbers" - -msgid "Email" -msgstr "Email" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "Enable Incoming Calls (Register via SIP)" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "" - -msgid "Enable Outgoing Calls" -msgstr "Enable Outgoing Calls" - -msgid "Enabled" -msgstr "Enabled" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "External SIP Port" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "Full Name" - -msgid "General Settings" -msgstr "General Settings" - -msgid "Google Accounts" -msgstr "Google Accounts" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "Google Voice/Talk Accounts" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "Incoming Calls" - -msgid "Insert QoS Rules" -msgstr "Insert QoS Rules" - -msgid "Makes Outgoing Calls" -msgstr "Makes Outgoing Calls" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "No" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "Outbound Proxy" - -msgid "Outgoing Calls" -msgstr "Outgoing Calls" - -msgid "PBX Main Page" -msgstr "PBX Main Page" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "PIN" - -msgid "Password" -msgstr "Password" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "Port Setting for SIP Devices" - -msgid "Providers Used for Outgoing Calls" -msgstr "Providers Used for Outgoing Calls" - -msgid "QoS Settings" -msgstr "QoS Settings" - -msgid "RTP Port Range End" -msgstr "RTP Port Range End" - -msgid "RTP Port Range Start" -msgstr "RTP Port Range Start" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "Receives Incoming Calls" - -msgid "Remote Usage" -msgstr "Remote Usage" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "SIP Accounts" - -msgid "SIP Device/Softphone Accounts" -msgstr "SIP Device/Softphone Accounts" - -msgid "SIP Provider Accounts" -msgstr "SIP Provider Accounts" - -msgid "SIP Realm (needed by some providers)" -msgstr "SIP Realm (needed by some providers)" - -msgid "SIP Server/Registrar" -msgstr "SIP Server/Registrar" - -msgid "SIP Server/Registrar Port" -msgstr "SIP Server/Registrar Port" - -msgid "Server Setting" -msgstr "Server Setting" - -msgid "Server Setting for Local SIP Devices" -msgstr "Server Setting for Local SIP Devices" - -msgid "Server Setting for Remote SIP Devices" -msgstr "Server Setting for Remote SIP Devices" - -msgid "Service Status" -msgstr "Service Status" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "Space-Separated List of Blacklisted Numbers" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "Specify numbers individually here. Press enter to add more numbers." - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." - -msgid "Use this account to make outgoing calls." -msgstr "Use this account to make outgoing calls." - -msgid "User Accounts" -msgstr "User Accounts" - -msgid "User Agent String" -msgstr "User Agent String" - -msgid "User Name" -msgstr "User Name" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "Yes" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "You can specify a real name to show up in the Caller ID here." - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -#~ msgid "Account Status" -#~ msgstr "Account Status" - -#~ msgid "Account Status Message" -#~ msgstr "Account Status Message" - -#~ msgid "Domain Name/Dynamic Domain Name" -#~ msgstr "Domain Name/Dynamic Domain Name" - -#~ msgid "Enable Incoming Calls (See Status, Message below)" -#~ msgstr "Enable Incoming Calls (See Status, Message below)" - -#~ msgid "Service Control and Connection Status" -#~ msgstr "Service Control and Connection Status" diff --git a/applications/luci-app-pbx/po/es/pbx.po b/applications/luci-app-pbx/po/es/pbx.po deleted file mode 100644 index 8071b61f08..0000000000 --- a/applications/luci-app-pbx/po/es/pbx.po +++ /dev/null @@ -1,677 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"PO-Revision-Date: 2014-06-15 13:15+0200\n" -"Last-Translator: José Vicente <josevteg@gmail.com>\n" -"Language-Team: none\n" -"Language: es\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=2; plural=(n != 1);\n" -"X-Generator: Pootle 2.0.6\n" - -msgid "Advanced Settings" -msgstr "Configuración avanzada" - -msgid "Available" -msgstr "Disponible" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "Usar sólo caracteres alfanuméricos, espacio, coma y punto." - -msgid "Away" -msgstr "No disponible" - -msgid "Blacklisted Numbers" -msgstr "Lista negra" - -msgid "Call Routing" -msgstr "Enrutado de llamadas" - -msgid "Call-back Numbers" -msgstr "Números de call-back" - -msgid "Call-back Provider" -msgstr "Proveedor de call-back" - -msgid "Call-through Numbers" -msgstr "Números call-through" - -msgid "Copy-paste large lists of numbers here." -msgstr "Pegue aquí grandes listas de números." - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" -"Listar los números a los que se permitirá llamar desde este sistema y qué " -"privilegios de usuario tendrán." - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" -"Listar los números a los que el sistema colgará y volverá a llamar, qué " -"proveedor se usará para llamarles y qué privilegios de usuario se les dará." - -msgid "Dials numbers unmatched elsewhere" -msgstr "Marca el resto de números en cualquier lugar" - -msgid "Do Not Disturb" -msgstr "No molestar" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "Dominio/Dirección IP/Dominio dinámico" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "Lista dinámica de números en lista negra" - -msgid "Email" -msgstr "e-mail" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "Permitir llamadas entrantes (registrar vía SIP)" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "Permitir llamadas entrantes (ver estado abajo)" - -msgid "Enable Outgoing Calls" -msgstr "Permitir llamadas salientes" - -msgid "Enabled" -msgstr "Activado" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" -"Proveedor VoIP para callbacks en formato nombredeusuario@algun.nombre.host, " -"tal y como se detalla arriba en \"Llamadas salientes\". Puede copiar y pegar " -"los proveedores desde ahí. Las entradas no válidas, incluyendo a proveedores " -"no habilitados para llamadas saliente, serán rechazadas sin mostrar aviso." - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" -"Números de teléfono de los que se reclina la llamada automáticamente. Es " -"posible que tenga que omitir el código de país y ceros precedentes, pero " -"experimente para asegurarse que bloquea los números correctamente." - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" -"Ponga esta IP (o IP:puerto) en el parámetro Servidor/Registrador de los " -"dispositivos SIP que usará SOLO localmente y nunca desde una posición remota." - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" -"Ponga este nombre de máquina en el parámetro Servidor/Registrador de los " -"dispositivos SIP que usará desde posiciones remotas (también vale " -"localmente)." - -msgid "External SIP Port" -msgstr "Puerto externo SIP" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" -"Para cada proveedor al que se habilita a hacer llamadas entrantes puede " -"restringir a qué usuarios llamar. Si se deja vacío el sistema indicará que " -"llamará a todos los usuarios que puedan recibir llamadas entrantes. Los " -"nombres de usuario no válidos se rechazarán sin aviso. Estos nombres de " -"usuario hacen ignorar la configuración de usuario de no recibir llamadas. De " -"esta manera puede hacer que a ciertos usuarios sólo les llamen ciertos " -"proveedores. Puede separar los nombres con espacios o poniéndolos en líneas " -"diferentes." - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" -"Para cada usuario habilitado a hacer llamadas salientes puede restringir qué " -"proveedores usar. Por defecto todos los usuarios pueden usar a todos los " -"proveedores. Para mostrarse en la lista el usuario debe poder hacer llamadas " -"salientes (ver página \"Cuentas de usuario\"). Ponga los proveedores en " -"formato username@some.host.name igual que se listan en \"Llamadas salientes" -"\" arriba. Los nombres no válidos se rechazarán sin aviso.Puede separar los " -"nombres con espacios o poniéndolos en líneas diferentes." - -msgid "Full Name" -msgstr "Nombre completo" - -msgid "General Settings" -msgstr "Configuración general" - -msgid "Google Accounts" -msgstr "Cuentas en google" - -msgid "Google Talk Status" -msgstr "Estado de Google Talk" - -msgid "Google Talk Status Message" -msgstr "Mensaje de estado de Google Talk" - -msgid "Google Voice/Talk Accounts" -msgstr "Cuentas Google Voice/Talk" - -msgid "Hang-up Delay" -msgstr "Retraso para descolgar" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" -"Configure una cuenta SIP que usará para conectar con este servicio. Úsela " -"tanpo en un adaptador de telefonía analógico (ATA) o en un programa SIP como " -"CSipSimple, Linphone, o Sipdroid para smartphones, o Ekiga, Linphone, o X-" -"Lite para ordenadores. Por defecto, todas las cuentas SIP sonarán a la vez " -"si se hace una llamada desde una de las cuentas de su proveedor de VoIP o " -"números GV." - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" -"Cuánto esperar antes de descolgar. Si el proveedor que usas para marcar " -"automáticamente desvía a un correo de voz puedes ajustar este valor con un " -"retraso que permitirá descolgar antes de que se desvíe la llamada y se " -"facture." - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" -"Si la configuración Servidor/Registrador en %s o %s no le funciona, prueba a " -"poner %s o %s e introduzca este número de puerto en un campo separado que " -"especifique el número de puerto del Servidor/Registrador. Algunos " -"dispositivos tienen una configuración extraña que muestra este puerto desde " -"el que el SIP origina peticiones en el mismo dispositivo SIP (el puerto " -"asociado). El puerto que está configurando aquí NO es este puerto asociado " -"sino el puerto en el que el servicio escucha." - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" -"Si nota saltos o retrasos en el audio mientras realiza descargas puede " -"querer activar QoS. QoS prioriza el tráfico a y desde su red para ciertos " -"puertos y direcciones IP mejorando la latencia y el rendimiento del sonido " -"en dicho caso. Al activarlo el PBX creará una regla QoS para este servicio, " -"pero deberá rellenar en la página de configuración de QoS (Red/QoS) otros " -"parámetros necesarios como la velocidad de subida y la de bajada." - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" -"Si tiene más de una cuenta para hacer llamadas salientes, debe introducir " -"una lista de números de teléfono y/o prefijos para cada proveedor. Los " -"prefijos no válidos se rechazarán sin aviso y solo son caracteres válidos " -"0-9, X, Z, N, #, *, y +. La letra X equivale a 0-9, Z a 1-9 y N a 2-9. Por " -"ejemplo para hacer llamadas a Alemania con su proveedor debe introducir 49. " -"Para hacer llamadas a Estados Unidos 1NXXNXXXXXX. Si uno de sus proveedores " -"puede hacer llamadas locales a un código de área como el 646 de Nueva York " -"debe introducir 646NXXXXXX para ese proveedor. Debería dehar una cuenta con " -"una lista vacía para que haga las llamadas por defecto en caso de que ningún " -"prefijo encaje. El sistema reemplazará automáticamente la lista vacía con el " -"mensaje de que el proveedor marca todos los números que no estén en los " -"prefijos de otros proveedores. Sea todo lo específico que pueda (ej. " -"1NXXNXXXXXX es mejor que 1). Todos los códigos internaciones de marcado se " -"descartan (ej. 00, 011, 010, 0011). Las entradas pueden ser una lista " -"separada por espacios y/o cambios de línea." - -msgid "Incoming Calls" -msgstr "Llamadas entrantes" - -msgid "Insert QoS Rules" -msgstr "Reglas QoS" - -msgid "Makes Outgoing Calls" -msgstr "Realizar llamadas salientes" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "NOTA: Sin cuentas configuradas de Google o porveedor SIP." - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" -"NOTA: Sin cuentas configuradas de Google o porveedor SIP para llamadas " -"entrantes." - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" -"NOTA: Sin cuentas configuradas de Google o porveedor SIP para llamadas " -"salientes." - -msgid "NOTE: There are no local user accounts configured." -msgstr "NOTA: Sin cuentas locales configuradas." - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "NOTA: Sin cuentas locales habilitadas para llamadas saientes." - -msgid "No" -msgstr "No" - -msgid "Number of Seconds to Ring" -msgstr "Número de segundos a sonar" - -msgid "Outbound Proxy" -msgstr "Proxy saliente" - -msgid "Outgoing Calls" -msgstr "Llamadas salientes" - -msgid "PBX Main Page" -msgstr "Página principal de PBX" - -msgid "PBX Service Status" -msgstr "Estado del servicio PBX" - -msgid "PIN" -msgstr "PIN" - -msgid "Password" -msgstr "Contraseña" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" -"Escoge un número de puerto aleatorio entre 6500 y 9500 para el servicio. No " -"elijas el estándar 5060 ya que es objeto, a menudo, de ataques por fuerza " -"bruta. Cuando hayas terminado pulsa en \"Salvar y aplicar\" y busca en la " -"sección \"Cuentas SIP del dispositivo/softphone\" el puerto actual para tus " -"dispositivos/softphones SIP." - -msgid "Port Setting for SIP Devices" -msgstr "Configuración de puerto para dispositivos SIP" - -msgid "Providers Used for Outgoing Calls" -msgstr "Proveedores usados para llamadas salientess" - -msgid "QoS Settings" -msgstr "Configuración de QoS" - -msgid "RTP Port Range End" -msgstr "Fin del rango de puertos RTP" - -msgid "RTP Port Range Start" -msgstr "Inicio del rango de puertos RTP" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" -"El tráfico RTP es el que lleva los paquetes de voz. Este es el inicio del " -"rango de puertos que se usará para comunicaciones RTP. Suele ser correcto " -"dejar el valor por defecto." - -msgid "Receives Incoming Calls" -msgstr "Recibe llamadas entrantes" - -msgid "Remote Usage" -msgstr "Uso remoto" - -msgid "Rings users enabled for incoming calls" -msgstr "Llama a usuarios habilitados a recibir llamadas" - -msgid "SIP Accounts" -msgstr "Cuentas SIP" - -msgid "SIP Device/Softphone Accounts" -msgstr "Dispositivo SIP/Cuentas Softphone" - -msgid "SIP Provider Accounts" -msgstr "Cuentas del proveedor SIP" - -msgid "SIP Realm (needed by some providers)" -msgstr "Ámbito SIP (necesario para algunos proveedores)" - -msgid "SIP Server/Registrar" -msgstr "Servidor/Registrador del SIP" - -msgid "SIP Server/Registrar Port" -msgstr "Puerto del Servidor/Registrador del SIP" - -msgid "Server Setting" -msgstr "Configuración del servidor" - -msgid "Server Setting for Local SIP Devices" -msgstr "Dispositivos SIP locales" - -msgid "Server Setting for Remote SIP Devices" -msgstr "Dispositivos SIP remotos" - -msgid "Service Status" -msgstr "Estado del servicio" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" -"Segundos que se llamará a los usuarios antes de colgar o pasar a correo voz " -"(si el correo voz está instalado y habilitado)." - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "Lista negra (separar números con espacios)" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "Números individuales. Pulse enter para añadir más." - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" -"Especifica números individualmente. Pulsa enter para añadir más. Tendrás que " -"experimentar con qué códigos de país y área necesitas añadir al número." - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" -"Estos números podrán llamar con los proveedores de este usuario. Los nombres " -"de usuario no válidos se descartan sin aviso. Por favor, verifique que los " -"números se aceptan." - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" -"Aquí puede configurar un servicio de sistema telefónico (PBX) que le " -"permitirá hacer llamadas por múltiples cuentas Google y SIP (como Sipgate, " -"SipSorcery, and Betamax) y compartirlas entre muchos dispositivos SIP. Tenga " -"en cuenta que las cuentas Google, SIP y locales deben configurarse en " -"subsecciones diferentes. Debe añadir al menos una cuenta de usuarioa este " -"PBX y configurar un dispositivo SIP o softphone para usarla para recibir las " -"llamadas de sus cuentas Google/SIP. Configurar múltiples usuarios le " -"permitirá hacer llamadas gratuitas entre los usuarios y compartir las " -"cuentas Google/SIP configuradas. Si tiene más de una cuenta Google/SIP " -"configurada tendrá que configurar cómo se enrutan en la página \"Enrutado de " -"llamadas\". Si está interesado en usar su PBX desde cualquier sitio del " -"mundo puede visitar la sección \"Uso remoto\" en la página \"Configuración " -"avanzada\"." - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" -"Nombre del servidor VoIP que usará para identificarse cuando se registre en " -"proveedores de VoIP (SIP). Algunos requieres que sea una cadena específica a " -"una dispositivo hardware." - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" -"Indique las cuentas Google/SIP que usará para llamar a qué códigos de país/" -"zona, qué usuarios pueden usuarios pueden usar qué cuentas SIP/Google y cómo " -"se enrutan las llamadas entrantes, qué números pueden entrar en esta PBX con " -"una contraseña y qué números están en lista negra." - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" -"Configure sus cuentas Google (Talk y Voz) para empezar a usarlas para hacer " -"y recibir llamadas (chat de voz y teléfono real). Haga al menos una llamada " -"de voz con el plugin de Google Talk (instalable desde GMail) y desconéctese " -"de la cuenta en cualquier otro sitio." - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" -"Configure sus cuentas SIP (VoIP) como Sipgate, SipSorcery, los popular " -"proveedores Betamax y cualquier otro proveedor para empezar a usarlos para " -"hacer y recibir llamadas (uri SIP y teléfono real)." - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" -"Debería ser \"Sí\" si tiene un DID (teléfono real) asociado a esta cuenta " -"SIP o quiere recibir llamads uri SIP de este proveedor." - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" -"Algunos de estos parámetros no suele ser necesario cambiarlos. Además puede " -"configurar su sistema para usar con dispositivos SIP remotos y resolver " -"problemas de calidad de llamada habilitando algunas reglas QoS." - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" -"Use nombre de usuario númericos (cuatro o cinco dígitos) si conecta a " -"teléfonos normales con ATAs a este sistema (para que puedan marcar números " -"de usuario)." - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" -"Cuenta para llamadas salientes como se configura en la sección \"Enrutado de " -"llamadas\"." - -msgid "Use this account to make outgoing calls." -msgstr "Cuenta para llamadas salientes." - -msgid "User Accounts" -msgstr "Cuentas de usuario" - -msgid "User Agent String" -msgstr "Cadena \"User Agent\"" - -msgid "User Name" -msgstr "Nombre de usuario" - -msgid "Uses providers enabled for outgoing calls" -msgstr "Usar proveedores habilitados para llamadas salientes" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" -"Cuando alguien inicia un chat de voz con su cuenta de GTalk o llame al " -"número de GVoice (si tiene Google Voice) la llamada se transferirá a " -"cualquier usuario que esté conectado (registrado usando un dispositivo SIP o " -"softphone) y se le permitirá recibir la llamada. Si tiene Google Voice debe " -"ir a la configuración de GVoice y traspasar las llamadas a Google chat para " -"recibir las hechas a si número de GVoice. Si tiene problemas recibiendo " -"llamadas de GVoice pruebe con la opción \"Call Screening\" en la " -"configuración de GVoice. Asegúrese de que ningún otro cliente esté conectado " -"con esta cuenta (navegador en gmail, o una aplicación para móvil o " -"escritorio) ya que podría interferir." - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" -"Cuando se salve su contraseña desaparece de este campo y no se muestra para " -"su seguridad. La contraseña sólo se podrá cambiar si introduce un valor " -"diferente al salvado." - -msgid "Yes" -msgstr "Sí" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" -"Puedes introducir el nombre de dominio, dirección IP external o nombre " -"dinámino aquí. Lo mejor es introducir una dirección IP estática. Si la " -"dirección es dinámica la configuración sería inválida cuando cambiase. En " -"estos casos es recomendable configurar Dynamic DNS e introducir tu nombre de " -"host Dynamic DNS. Puedes instalar y configurar Dynamic DNS con el paquete " -"luci-app-ddns." - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "Nombre real a mostrar en el \"Caller ID\"." - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" -"Puede usar sus dispositivos SIP/softphones con este sistema desde una " -"ubicación remota mientras su proveedor de internet le dé una dirección IP " -"pública. Podrá llamar a usuarios locales gratis (ej. otros adaptadores de " -"teléfonos analógicos) y podrá usar sus proveedores de VoIP para hacer " -"llamadas como si estuviese en su PBX local. Tras configurar esta pestaña " -"vuelva a la configuración de usuarios y veo el nuevo servidor y puerto que " -"debe configurar en sus dispositivos SIP remotos. Tenga en cuenta que si este " -"PBX no funciona en su router/pasarela, tendrá que configurar el traspaso de " -"puertos (NAT) en su router/pasarela. Traspase los puertos indicados (Puerto " -"SIP y rango RTP) hacia la dirección IP del dispositivo en que corre esta PBX." - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" -"Su PIN desaparecerá cuando se salve para su protección. Se cambiará solo " -"cuando introduzca un valor diferente al salvado. No se puede dejar el PIN " -"vacío." - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" -"Su contraseña desaparecerá cuando se salve para su protección. Sólo se puede " -"cambiar si entra un valor diferente al salvado." - -#~ msgid "" -#~ "Designate numbers that are allowed to call through this system and which " -#~ "user's privileges it will have." -#~ msgstr "" -#~ "Números a los que se permite llamar por este sistema y privilegios de " -#~ "usuario." - -#~ msgid "" -#~ "Pick a random port number between 6500 and 9500 for the service to listen " -#~ "on. Do not pick the standard 5060, because it is often subject to brute-" -#~ "force attacks. When finished, (1) click \"Save and Apply\", and (2) click " -#~ "the \"Restart VoIP Service\" button above. Finally, (3) look in the \"SIP " -#~ "Device/Softphone Accounts\" section for updated Server and Port settings " -#~ "for your SIP Devices/Softphones." -#~ msgstr "" -#~ "Puerto aleatorio entre 6500 y 9500 en el que escuche el servicio. No elija " -#~ "el estándar 5060 porque es susceptible de ataques por fuerza bruta. Cuando " -#~ "termine (1) pulsa \"Salvar y aplicar\" y (2) pulse \"Rearrancar servicio VoIP\". " -#~ "Finalmente (3) busque en la sección \"Dispositivo SIP/Cuentas softphone\" la " -#~ "configuración del puerto." - -#~ msgid "" -#~ "You can enter your domain name, external IP address, or dynamic domain " -#~ "name here Please keep in mind that if your IP address is dynamic and it " -#~ "changes your configuration will become invalid. Hence, it's recommended " -#~ "to set up Dynamic DNS in this case." -#~ msgstr "" -#~ "Nombre de dominio, dirección IP externa o nombre de dominio dinámico. Si su " -#~ "dirección IP es dinámica y cambia su configuración podría resultar no " -#~ "válida. Se recomienda el uso de DNS dinámico en estos casos." diff --git a/applications/luci-app-pbx/po/fr/pbx.po b/applications/luci-app-pbx/po/fr/pbx.po deleted file mode 100644 index 971a696488..0000000000 --- a/applications/luci-app-pbx/po/fr/pbx.po +++ /dev/null @@ -1,484 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"Last-Translator: Automatically generated\n" -"Language-Team: none\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=2; plural=(n > 1);\n" - -msgid "Advanced Settings" -msgstr "" - -msgid "Available" -msgstr "" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" - -msgid "Away" -msgstr "" - -msgid "Blacklisted Numbers" -msgstr "" - -msgid "Call Routing" -msgstr "" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "" - -msgid "Email" -msgstr "" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "" - -msgid "Enable Outgoing Calls" -msgstr "" - -msgid "Enabled" -msgstr "" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "" - -msgid "General Settings" -msgstr "" - -msgid "Google Accounts" -msgstr "" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "" - -msgid "Insert QoS Rules" -msgstr "" - -msgid "Makes Outgoing Calls" -msgstr "" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "" - -msgid "PBX Main Page" -msgstr "" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "" - -msgid "Password" -msgstr "" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" diff --git a/applications/luci-app-pbx/po/he/pbx.po b/applications/luci-app-pbx/po/he/pbx.po deleted file mode 100644 index 2a458214df..0000000000 --- a/applications/luci-app-pbx/po/he/pbx.po +++ /dev/null @@ -1,484 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"Last-Translator: Automatically generated\n" -"Language-Team: none\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=2; plural=(n != 1);\n" - -msgid "Advanced Settings" -msgstr "" - -msgid "Available" -msgstr "" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" - -msgid "Away" -msgstr "" - -msgid "Blacklisted Numbers" -msgstr "" - -msgid "Call Routing" -msgstr "" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "" - -msgid "Email" -msgstr "" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "" - -msgid "Enable Outgoing Calls" -msgstr "" - -msgid "Enabled" -msgstr "" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "" - -msgid "General Settings" -msgstr "" - -msgid "Google Accounts" -msgstr "" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "" - -msgid "Insert QoS Rules" -msgstr "" - -msgid "Makes Outgoing Calls" -msgstr "" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "" - -msgid "PBX Main Page" -msgstr "" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "" - -msgid "Password" -msgstr "" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" diff --git a/applications/luci-app-pbx/po/hu/pbx.po b/applications/luci-app-pbx/po/hu/pbx.po deleted file mode 100644 index 2a458214df..0000000000 --- a/applications/luci-app-pbx/po/hu/pbx.po +++ /dev/null @@ -1,484 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"Last-Translator: Automatically generated\n" -"Language-Team: none\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=2; plural=(n != 1);\n" - -msgid "Advanced Settings" -msgstr "" - -msgid "Available" -msgstr "" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" - -msgid "Away" -msgstr "" - -msgid "Blacklisted Numbers" -msgstr "" - -msgid "Call Routing" -msgstr "" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "" - -msgid "Email" -msgstr "" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "" - -msgid "Enable Outgoing Calls" -msgstr "" - -msgid "Enabled" -msgstr "" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "" - -msgid "General Settings" -msgstr "" - -msgid "Google Accounts" -msgstr "" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "" - -msgid "Insert QoS Rules" -msgstr "" - -msgid "Makes Outgoing Calls" -msgstr "" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "" - -msgid "PBX Main Page" -msgstr "" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "" - -msgid "Password" -msgstr "" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" diff --git a/applications/luci-app-pbx/po/it/pbx.po b/applications/luci-app-pbx/po/it/pbx.po deleted file mode 100644 index 6da8e45d96..0000000000 --- a/applications/luci-app-pbx/po/it/pbx.po +++ /dev/null @@ -1,487 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"PO-Revision-Date: 2012-12-15 19:31+0200\n" -"Last-Translator: claudyus <claudyus84@gmail.com>\n" -"Language-Team: none\n" -"Language: it\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=2; plural=(n != 1);\n" -"X-Generator: Pootle 2.0.6\n" - -msgid "Advanced Settings" -msgstr "Opzioni avanzate" - -msgid "Available" -msgstr "" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" - -msgid "Away" -msgstr "" - -msgid "Blacklisted Numbers" -msgstr "" - -msgid "Call Routing" -msgstr "" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "" - -msgid "Email" -msgstr "" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "" - -msgid "Enable Outgoing Calls" -msgstr "" - -msgid "Enabled" -msgstr "" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "" - -msgid "General Settings" -msgstr "" - -msgid "Google Accounts" -msgstr "" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "" - -msgid "Insert QoS Rules" -msgstr "" - -msgid "Makes Outgoing Calls" -msgstr "" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "" - -msgid "PBX Main Page" -msgstr "" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "" - -msgid "Password" -msgstr "" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" diff --git a/applications/luci-app-pbx/po/ja/pbx.po b/applications/luci-app-pbx/po/ja/pbx.po deleted file mode 100644 index 76199f4191..0000000000 --- a/applications/luci-app-pbx/po/ja/pbx.po +++ /dev/null @@ -1,493 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"PO-Revision-Date: 2012-04-21 07:57+0200\n" -"Last-Translator: Kentaro <kentaro.matsuyama@gmail.com>\n" -"Language-Team: none\n" -"Language: ja\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=1; plural=0;\n" -"X-Generator: Pootle 2.0.4\n" - -msgid "Advanced Settings" -msgstr "詳細設定" - -msgid "Available" -msgstr "" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" - -msgid "Away" -msgstr "" - -msgid "Blacklisted Numbers" -msgstr "" - -msgid "Call Routing" -msgstr "" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "" - -msgid "Email" -msgstr "Eメール" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "" - -msgid "Enable Outgoing Calls" -msgstr "" - -msgid "Enabled" -msgstr "" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "外部SIPポート" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "" - -msgid "General Settings" -msgstr "基本設定" - -msgid "Google Accounts" -msgstr "Google アカウント" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "Google Voice/Talk アカウント" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "" - -msgid "Insert QoS Rules" -msgstr "QoS ルール設定を有効にする" - -msgid "Makes Outgoing Calls" -msgstr "発信を許可する" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "いいえ" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "" - -msgid "PBX Main Page" -msgstr "PBX メインページ" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "PIN" - -msgid "Password" -msgstr "パスワード" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "QoS 設定" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "受信を許可する" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "SIP アカウント" - -msgid "SIP Device/Softphone Accounts" -msgstr "SIP デバイス/ソフトフォン アカウント" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "サーバー設定" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "" - -msgid "User Agent String" -msgstr "ユーザーエージェント名" - -msgid "User Name" -msgstr "ユーザー名" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "はい" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -#~ msgid "Account Status" -#~ msgstr "アカウントのステータス" - -#~ msgid "Account Status Message" -#~ msgstr "アカウントステータス・メッセージ" diff --git a/applications/luci-app-pbx/po/ms/pbx.po b/applications/luci-app-pbx/po/ms/pbx.po deleted file mode 100644 index 23403f290f..0000000000 --- a/applications/luci-app-pbx/po/ms/pbx.po +++ /dev/null @@ -1,483 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"Last-Translator: Automatically generated\n" -"Language-Team: none\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" - -msgid "Advanced Settings" -msgstr "" - -msgid "Available" -msgstr "" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" - -msgid "Away" -msgstr "" - -msgid "Blacklisted Numbers" -msgstr "" - -msgid "Call Routing" -msgstr "" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "" - -msgid "Email" -msgstr "" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "" - -msgid "Enable Outgoing Calls" -msgstr "" - -msgid "Enabled" -msgstr "" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "" - -msgid "General Settings" -msgstr "" - -msgid "Google Accounts" -msgstr "" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "" - -msgid "Insert QoS Rules" -msgstr "" - -msgid "Makes Outgoing Calls" -msgstr "" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "" - -msgid "PBX Main Page" -msgstr "" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "" - -msgid "Password" -msgstr "" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" diff --git a/applications/luci-app-pbx/po/no/pbx.po b/applications/luci-app-pbx/po/no/pbx.po deleted file mode 100644 index 2a458214df..0000000000 --- a/applications/luci-app-pbx/po/no/pbx.po +++ /dev/null @@ -1,484 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"Last-Translator: Automatically generated\n" -"Language-Team: none\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=2; plural=(n != 1);\n" - -msgid "Advanced Settings" -msgstr "" - -msgid "Available" -msgstr "" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" - -msgid "Away" -msgstr "" - -msgid "Blacklisted Numbers" -msgstr "" - -msgid "Call Routing" -msgstr "" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "" - -msgid "Email" -msgstr "" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "" - -msgid "Enable Outgoing Calls" -msgstr "" - -msgid "Enabled" -msgstr "" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "" - -msgid "General Settings" -msgstr "" - -msgid "Google Accounts" -msgstr "" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "" - -msgid "Insert QoS Rules" -msgstr "" - -msgid "Makes Outgoing Calls" -msgstr "" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "" - -msgid "PBX Main Page" -msgstr "" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "" - -msgid "Password" -msgstr "" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" diff --git a/applications/luci-app-pbx/po/pl/pbx.po b/applications/luci-app-pbx/po/pl/pbx.po deleted file mode 100644 index 4e80a45815..0000000000 --- a/applications/luci-app-pbx/po/pl/pbx.po +++ /dev/null @@ -1,508 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"PO-Revision-Date: 2014-05-05 04:37+0200\n" -"Last-Translator: piosl <sleczek.piotr@gmail.com>\n" -"Language-Team: none\n" -"Language: pl\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=3; plural=(n==1 ? 0 : n%10>=2 && n%10<=4 && (n%100<10 " -"|| n%100>=20) ? 1 : 2);\n" -"X-Generator: Pootle 2.0.6\n" - -msgid "Advanced Settings" -msgstr "Ustawienia zaawansowane" - -msgid "Available" -msgstr "Dostępny" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "Unikaj znaków innych niż alfanumeryczne, spacja, przecinek i kropka." - -msgid "Away" -msgstr "Oddalony" - -msgid "Blacklisted Numbers" -msgstr "Numery na czarnej liście" - -msgid "Call Routing" -msgstr "Przekierowanie połączeń" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -# Chodzi tu o numery, przez które dzwoni się, aby obniżyć koszta połączeń zagranicznych. Jeśli ktoś ma pomysł na lepsze tłumaczenie, proszę zmienić. W sieci nie znalazłem. -msgid "Call-through Numbers" -msgstr "Numery pośredniczące" - -msgid "Copy-paste large lists of numbers here." -msgstr "Wklej tu wielkie listy numerów." - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "Nie przeszkadzać" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "Domena/adres IP/dynamiczna domena" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "Dynamiczna czarna lista numerów" - -msgid "Email" -msgstr "E-mail" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "Włącz połączenia przychodzące (rejestruj przez SIP)" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "Włącz połączenia przychodzące (zobacz status poniżej)" - -msgid "Enable Outgoing Calls" -msgstr "Włącz połączenia wychodzące" - -msgid "Enabled" -msgstr "Włączone" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" -"Podaj numery telefonów, które powinny być automatycznie odrzucane. " -"Prawdopodobnie powinieneś pominąć numer kierunkowy kraju i zera z przodu, " -"ale samemu to przetestuj, aby upewnić się, że blokowanie działa prawidłowo " -"dla Twojego położenia." - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" -"Podaj to IP (lub parę IP:port) w ustawieniach serwera/rejestratora urządzeń " -"SIP których będziesz używać WYŁĄCZNIE lokalnie i nigdy z zewnątrz." - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" -"Podaj tę nazwę hosta (lub parę nazwa hosta:port) w ustawieniach serwera/" -"rejestratora urządzeń SIP których będziesz używać z zewnątrz (będą też " -"działać lokalnie)." - -msgid "External SIP Port" -msgstr "Zewnętrzny port SIP" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" -"Dla każdego użytkownika z prawem wykonywania połączeń wychodzących możesz " -"ograniczyć których operatorów mogą używać do tych połączeń. Domyślnie każdy " -"użytkownik może używać dowolnego operatora. Użytkownik musi mieć prawo " -"wykonywania połączeń wychodzących ustawione na stronie \"Konta użytkowników" -"\", aby pojawić się na poniższej liście. Podaj operatorów VoIP w formacie " -"nazwa.użytkownika@jakaś.nazwa.hosta, tak jak są wypisani w \"Połączeniach " -"wychodzących\" powyżej. Łatwiej jest skopiować powyższych operatorów. " -"Nieprawidłowe wpisy, włącznie z operatorami bez prawa do połączeń " -"wychodzących, będą odrzucani bez komunikatów. Wpisy mogą być rozdzielone " -"spacjami albo podane po jednym w wierszu." - -msgid "Full Name" -msgstr "Pełne imię i nazwisko" - -msgid "General Settings" -msgstr "Ustawienia ogólne" - -msgid "Google Accounts" -msgstr "Konta Google" - -msgid "Google Talk Status" -msgstr "Status Google Talk" - -msgid "Google Talk Status Message" -msgstr "Opis Google Talk" - -msgid "Google Voice/Talk Accounts" -msgstr "Konta Google Voice/Talk" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "Połączenia przychodzące" - -msgid "Insert QoS Rules" -msgstr "" - -msgid "Makes Outgoing Calls" -msgstr "" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "" - -msgid "PBX Main Page" -msgstr "" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "" - -msgid "Password" -msgstr "" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" diff --git a/applications/luci-app-pbx/po/pt-br/pbx.po b/applications/luci-app-pbx/po/pt-br/pbx.po deleted file mode 100644 index fd93e4fffb..0000000000 --- a/applications/luci-app-pbx/po/pt-br/pbx.po +++ /dev/null @@ -1,744 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"PO-Revision-Date: 2014-08-04 09:00+0200\n" -"Last-Translator: Luiz Angelo <luizluca@gmail.com>\n" -"Language-Team: none\n" -"Language: pt_BR\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=2; plural=(n > 1);\n" -"X-Generator: Pootle 2.0.6\n" - -msgid "Advanced Settings" -msgstr "Configurações Avançadas" - -msgid "Available" -msgstr "Disponível" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" -"Evite usar qualquer carácter que não seja um alfanumérico, espaço, vírgula " -"ou ponto." - -msgid "Away" -msgstr "Ausente" - -msgid "Blacklisted Numbers" -msgstr "Números na Lista Negra" - -msgid "Call Routing" -msgstr "Roteamento de Chamada" - -# 20140630: edersg: tradução -msgid "Call-back Numbers" -msgstr "Voltar a discar os números" - -# 20140630: edersg: tradução -msgid "Call-back Provider" -msgstr "Voltar a chamar o provedor" - -msgid "Call-through Numbers" -msgstr "Números de Ligação Direta" - -msgid "Copy-paste large lists of numbers here." -msgstr "Copie e cole aqui listas de números extensas." - -# 20140630: edersg: tradução -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" -"Designar os números que estão autorizados a chamar por este sistema e quais " -"privilégios do usuário eles terão." - -# 20140630: edersg: tradução -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" -"Designar números para os quais o sistema irá desligar e ligar de volta, qual " -"provedor será utilizado para chamá-los, e quais privilégios do usuário " -"serão concedidos a eles." - -msgid "Dials numbers unmatched elsewhere" -msgstr "Disca números que não casam em qualquer lugar." - -msgid "Do Not Disturb" -msgstr "Não Perturbe" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "Domínio/Endereço IP/Domínio Dinâmico" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "Lista Dinâmica dos Números da Lista Negra" - -msgid "Email" -msgstr "Email" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "Habilitar Chamadas Recebidas (Registrar pelo SIP)" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "Habilitar Chamadas Recebidas (defina o Estado abaixo)" - -msgid "Enable Outgoing Calls" -msgstr "Habilitar Chamadas para Fora" - -msgid "Enabled" -msgstr "Habilitado" - -# 20140630: edersg: tradução -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" -"Digite um provedor VoIP para utilizar para voltar a chamada no formato " -"username@some.host.name conforme listado acima em \"Chamadas Originadas\". É " -"mais fácil copiar e colar os provedores. As entradas inválidas, incluindo " -"provedores não habilitados para chamadas de saída, serão rejeitados em " -"silêncio." - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" -"Entre com os números de telefone que você deseja rejeitar automaticamente. " -"Você pode omitir o código do país e qualquer zeros no início, mas, por " -"favor, teste para ter certeza que você está bloqueando da área desejada com " -"sucesso." - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" -"Entre este endereço IP (ou IP:porta) na configuração de servidor/registrador " -"dos seus dispositivos SIP que você irá usar SOMENTE localmente e nunca de um " -"local remoto." - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" -"Entre com o nome do equipamento (ou equipamento:porta) na configuração de " -"servidor/Registrar do seus dispositivos SIP que você irá usar de um local " -"remoto (eles também funcionarão localmente)." - -msgid "External SIP Port" -msgstr "Porta SIP Externa" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" -"Para cada provedor habilitado para receber chamadas, aqui você pode " -"restringir quais usuários tocarão quando receber chamadas. Se a lista " -"estiver vazia, o sistema indicará que todos os usuários com recepção de " -"chamadas habilitada tocarão. Nome de usuários inválidos serão rejeitados " -"silenciosamente. Além disto, entrar com um nome de usuário aqui sobrescreve " -"a configuração do usuário para não receber chamadas. Desta forma, você pode " -"fazer com que alguns usuários toquem somente para alguns provedores " -"específicos. As entradas podem ser inseridas usando uma lista separada por " -"espaço ou um por nova linha." - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" -"Para cada usuário habilitado para realizar chamadas externas, você pode " -"restringir quais provedores o usuário poderá usar. Por padrão, todos os " -"usuários podem usar todos os provedores. Para aparecer na lista abaixo, o " -"usuário deve estar habilitado para realizar chamadas externas na página de " -"\"Contas de Usuários\". Entre com os provedores de VoIP no formato " -"usuário@algum.nome.de.equipamento, como listado em \"Chamadas Efetuadas\" " -"abaixo. É mais fácil copiar e colar os provedores da lista abaixo. Entradas " -"inválidas, includindo provedores não habilitados para chamadas externas, " -"serão rejeitadas silenciosamente. As entradas podem ser inseridas usando uma " -"lista separada por espaço ou um por nova linha." - -msgid "Full Name" -msgstr "Nome Completo" - -msgid "General Settings" -msgstr "Configurações Gerais" - -msgid "Google Accounts" -msgstr "Contas do Google" - -msgid "Google Talk Status" -msgstr "Estado do Google Talk" - -msgid "Google Talk Status Message" -msgstr "Mensagem de Estado do Google Talk" - -msgid "Google Voice/Talk Accounts" -msgstr "Contas do Google Voice/Talk" - -# 20140630: edersg: tradução -msgid "Hang-up Delay" -msgstr "Atraso de hang-up" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" -"Aqui você deve configurar pelo menos uma conta SIP, que você irá usar para " -"se cadastrar neste serviço. Use essa conta, seja em um adaptador de " -"telefonia analógica (ATA), ou em um softphone SIP como Linphone, CSipSimple, " -"ou Sipdroid em seu smartphone, ou o Ekiga, Linphone, ou X-Lite no seu " -"computador. Por padrão, ao receber uma chamada em uma das suas contas nos " -"provedores VoIP ou em números GV, todas as contas SIP tocarão " -"simultaneamente." - -# 20140630: edersg: tradução -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" -"Quanto tempo esperar antes de desligar. Se o provedor que você utiliza para " -"discar automaticamente encaminha para a caixa postal de voz, você pode " -"definir este valor para um atraso que irá permitir que você desligue sua " -"chamada antes de ser encaminhada e cobrado financeiramente por isso." - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" -"Se definir o servidor/registrador como %s ou %s não funcionar para você, " -"tente defini-lo como %s ou %s e entre com este número de porta em um campo " -"separado que especifica o número da porta do servidor/registrador. Fique " -"ciente que alguns dispositivos têm uma configuração confusa que define a " -"porta de origem das solicitações SIP no dispositivo SIP em si (a porta local " -"no dispositivo). A porta especificada nesta página não é essa porta de " -"ligação, mas a porta na qual o serviço escutará serviço." - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" -"Se você sentir falhas ou alta latência enquanto baixa conteúdos pesados, " -"você pode querer habilitar o <abbr title=\"Quality of Service, Qualidade de " -"serviço\">QoS</abbr>. O <abbr title=\"Quality of Service, Qualidade de " -"serviço\">QoS</abbr> prioriza o tráfego de e para a sua rede para endereços " -"IP e portas específicas, resultando em melhor latência e redimento de som. " -"Se ativado, será configurada automaticamente pelo PABX uma regra de <abbr " -"title=\"Quality of Service, Qualidade de serviço\">QoS</abbr> para este " -"serviço, mas você deve visitar a página de configuração de <abbr title=" -"\"Quality of Service, Qualidade de serviço\">QoS</abbr> (Rede -> QoS) para " -"configurar outras configurações críticas de QoS como as velocidades da sua " -"conexão." - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" -"Se você tiver mais de uma conta que pode fazer chamadas externas, você deve " -"informar uma lista de números de telefone e/ou prefixos nos seguintes campos " -"para cada provedor listados. Prefixos inválidos são removidos " -"silênciosamente, e some os caracteres 0-9, X, Z, N, # *,, e + são válidos. A " -"letra X corresponde a 0-9, Z corresponde a 1-9, e N corresponde a 2-9. Por " -"exemplo, para fazer chamadas para a Alemanha através de um provedor, você " -"pode digitar 49. Para fazer chamadas para a América do Norte, você pode " -"entrar 1NXXNXXXXXX. Se um de seus provedores pode fazer chamadas locais para " -"um código de área como Nova York (646), você pode entrar com 646NXXXXXX para " -"esse provedor. Você deve deixar uma conta com uma lista vazia para fazer " -"chamadas com ele por padrão para o caso do prefixo não casar com nenhum " -"outro fornecedor. O sistema irá substituir automaticamente uma lista vazia " -"com uma mensagem que os este provedor será utilizado caso nenhuma das regras " -"dos demais provedores casem. Seja tão específico quanto possível (isto é " -"1NXXNXXXXXX é melhor do que 1). Por favor, note que todos os códigos de " -"discagem internacionais são descartados (por exemplo 00, 011, 010, 0011). As " -"entradas podem ser feitas em uma lista separada por espaços ou por nova " -"linha." - -msgid "Incoming Calls" -msgstr "Chamadas Recebidas" - -msgid "Insert QoS Rules" -msgstr "Inserir Regras QoS" - -msgid "Makes Outgoing Calls" -msgstr "Realiza Chamadas para Fora" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "NOTA: Não existe uma conta Google ou provedor SIP configurado." - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" -"NOTA: Não existe uma conta Google ou provedor SIP habilitado para receber " -"chamadas." - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" -"NOTA: Não existe uma conta Google ou provedor SIP habilitado para efetuar " -"chamadas externas." - -msgid "NOTE: There are no local user accounts configured." -msgstr "NOTA: Não existe uma conta local configurada." - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" -"NOTA: Não existe uma conta local configurada para efetuar chamadas externas." - -msgid "No" -msgstr "Não" - -msgid "Number of Seconds to Ring" -msgstr "Número de Segundos para Tocar" - -msgid "Outbound Proxy" -msgstr "Proxy Externo" - -msgid "Outgoing Calls" -msgstr "Chamadas Efetuadas" - -msgid "PBX Main Page" -msgstr "Página Principal do PBX" - -msgid "PBX Service Status" -msgstr "Estado do Serviço PBX" - -msgid "PIN" -msgstr "PIN" - -msgid "Password" -msgstr "Senha" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" -"Escolha uma porta aleatória entre 6500 e 9500 onde o serviço irá escutar. " -"Não escolha a porta padrão 5060 pois ela é frequentemente alvo de ataques de " -"força bruta. Quanto terminar, (1) clique em \"Salvar e Aplicar\", e (2) olhe " -"na seção \"Dispositivo SIP/Contas do Softphone\" para as configurações " -"atualizadas do servidor e porta para o seu Dispositivo SIP/Softphone." - -msgid "Port Setting for SIP Devices" -msgstr "Configuração da Porta para Dispositivos SIP" - -msgid "Providers Used for Outgoing Calls" -msgstr "Provedores Usados para as Chamadas para Fora" - -msgid "QoS Settings" -msgstr "Configurações de QoS" - -msgid "RTP Port Range End" -msgstr "Final da Faixa de Portas RTP" - -msgid "RTP Port Range Start" -msgstr "Inicio da Faixa de Portas RTP" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" -"O tráfego RTP transporta de fato os pacotes de voz. Este é o início do " -"intervalo de portas que será usado para a estabelecer uma comunicação RTP. " -"Geralmente não é um problema deixar esta configuração com o valor padrão." - -msgid "Receives Incoming Calls" -msgstr "Recebe Chamadas para Dentro" - -msgid "Remote Usage" -msgstr "Uso Remoto" - -msgid "Rings users enabled for incoming calls" -msgstr "Toca usuários habilitados para receber chamadas" - -msgid "SIP Accounts" -msgstr "Contas SIP" - -msgid "SIP Device/Softphone Accounts" -msgstr "Contas de Dispositivos SIP/Telefones em Software" - -msgid "SIP Provider Accounts" -msgstr "Contas dos Provedores SIP" - -msgid "SIP Realm (needed by some providers)" -msgstr "Domínio SIP (necessário para alguns provedores)" - -msgid "SIP Server/Registrar" -msgstr "Servidor SIP/Registrador" - -msgid "SIP Server/Registrar Port" -msgstr "Porta do Servidor SIP/Registrador" - -msgid "Server Setting" -msgstr "Configuração do Servidor" - -msgid "Server Setting for Local SIP Devices" -msgstr "Configuração do Servidor para Dispositivos SIP Locais" - -msgid "Server Setting for Remote SIP Devices" -msgstr "Configuração do Servidor para Dispositivos SIP Remotos" - -msgid "Service Status" -msgstr "Estado do Serviço" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" -"Define o número de segundos para tocar o telefone ao receber chamadas antes " -"de desligar ou ir para a caixa postal, se o correio de voz estiver instalado " -"e habilitado." - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "Números na Lista Negra separados por Espaço" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" -"Especifique os números individualmente aqui. Pressione o Enter para " -"adicionar mais números." - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" -"Especifique aqui os números individualmente. Pressione o \"Enter\" para " -"adicionar mais números. Você terá que experimentar com qual código de país " -"ou de área você precisa adicionar aos números." - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" -"O número(s) acima especificados serão capazes de discar com os provedores " -"deste usuário. Nomes inválidos, incluindo usuários não habilitados para " -"chamadas externas, serão descartados silenciosamente. Por favor, verifique " -"se a entrada foi aceita." - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" -"Esta página de configuração permite configurar um sistema de serviço de " -"telefone (PABX), que permite fazer chamadas telefônicas através do Google " -"múltipla e SIP (como Sipgate, SipSorcery e Betamax) contas e compartilhá-los " -"entre diversos dispositivos SIP. Note-se que as contas do Google, contas " -"SIP, e contas de usuários locais são configurados em \"Contas do Google\", " -"\"Contas SIP\" e \"Contas de Usuário\" sub-seções. Você deve adicionar pelo " -"menos uma conta de usuário para este PABX e configurar um dispositivo SIP ou " -"softphone para usar a conta, a fim de fazer e receber chamadas com o " -"Google / SIP contas. Configurando vários usuários permitem que você faça " -"chamadas gratuitas entre todos os usuários, e partilhar o Google configurado " -"e contas SIP. Se você tem mais de um Google e contas SIP configurado, você " -"provavelmente deve configurar como as chamadas de e para eles são " -"encaminhados para a \"Call Routing\" página. Se você está interessado em " -"usar o seu próprio PABX de qualquer lugar do mundo, então, visitar o " -"\"Remote Uso\" na seção \"Advanced Settings\" página." - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" -"Este é o nome que o servidor VoIP será usado para identificar-se quando se " -"registrar para VoIP (SIP) fornecedores. Alguns provedores exigem isso para " -"uma seqüência específica de correspondência de um dispositivo de hardware " -"SIP." - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" -"Este é o local onde você indica quais contas Google/SIP serão usadas para " -"chamar quais códigos de área/país, que usuários poderão usar quais contas " -"Google/SIP, como as chamadas recebidas serão roteadas, que números podem ser " -"recebidos por este PBX com uma senha e qual números estão banidos." - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" -"Este é o local onde você configura suas contas Google (Talk e Voice) para " -"poder usá-las para realizar ou receber chamadas (conversa por voz e chamadas " -"para telefones reais). Por favor, realize ao menos uma chamada de voz usando " -"o plugin do Google Talk, instalável na interface do GMail. Após esta " -"chamada, saia da sua conta em todos os serviços. Clique em \"Adicionar\" " -"para adicionar quantas contas você desejar." - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" -"Este é o local onde você configura suas contas SIP (VoIP) como Sipgate, " -"SipSorcery, os populares provedores Betamax, e qualquer outro provedor com " -"suporte a SIP para permitir o uso destas contas para efetuar e receber " -"chamadas (URI de SIP e chamads para números reais). Clique em \"Adicionar\" " -"para adicionar quantas contas você desejar." - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" -"Esta opção deve estar definida como \"Sim\" se você tem um DDR (Discagem " -"Direta a Ramal) associado com esta conta SIP or quer receber chamadas URI de " -"SIP através deste provedor." - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" -"Esta seção contém configurações que não precisam ser modificadas em " -"condições normais. Aqui você pode configurar seu sistema para usar com " -"dispositivos SIP remotos e resolver problemas com a qualidade das chamadas " -"através da inserção de regras de <abbr title=\"Quality of Service, Qualidade " -"de serviço\">QoS</abbr>." - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" -"Use o nome de usuário numérico (4 a 5 dígitos) se você estiver conectando " -"telefones normais com ATAs para este sistema (para que eles possam discar os " -"nomes de seus usuários)." - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" -"Use esta conta para realizar chamadas externas como configurado na seção de " -"\"Roteamento de Chamada\"." - -msgid "Use this account to make outgoing calls." -msgstr "Use esta conta para realizar chamadas externas." - -msgid "User Accounts" -msgstr "Contas de Usuários" - -msgid "User Agent String" -msgstr "Texto para o Agente do Usuário" - -msgid "User Name" -msgstr "Nome do Usuário" - -msgid "Uses providers enabled for outgoing calls" -msgstr "Usa provedores habilitados para chamadas externas" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" -"Quando alguém iniciar uma conversa por voz com sua conta do GTalk ou chamar " -"seu número GVoice (se você tiver uma conta Google Voice), a chamada será " -"encaminhada para qualquer usuários que estão conectados (registados " -"utilizando um dispositivo SIP ou softphone) e autorizados a receber a " -"chamada. Se você tiver uma conta Google Voice, você deve ir para as " -"configurações da sua conta GVoice e encaminhar as chamadas para o Google " -"Chat, a fim de realmente receber chamadas feitas para o seu número GVoice. " -"Se você tiver problemas para receber chamadas oriundas do GVoice, " -"experimente a opção \"Call Screening/Monitoramento de Chamadas\" na " -"configurações da sua conta GVoice. Finalmente, certifique-se de nenhum outro " -"cliente está online com essa conta (navegador contado no GMail, aplicativo " -"Google Talk no Desktop ou Celular), pois isto pode interferir." - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" -"Quando a sua senha for salva, ela desaparece deste campo e não será exibida " -"para sua proteção. A senha será alterada somente quando você informar uma " -"nova senha diferente da que foi salva anteriormente." - -msgid "Yes" -msgstr "Sim" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" -"Você pode informar aqui o nome do domínio, endereço IP externo, ou um nome " -"de domínio dinâmico. O melhor é informar um endereço IP estático. Se o seu " -"endereço IP é dinâmico e ele muda, sua configuração se tornará inválida. " -"Desta forma, é recomendado configurar um serviço de domínios dinâmicos e " -"utilizar este nome aqui. Você pode configurar o serviço de domínios " -"dinâmicos com o pacote luci-app-ddns." - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" -"Você pode especificar um nome real para aparecer no identificador de " -"chamadas." - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" -"Você pode usar seus dispositivos SIP/softphones com este sistema a partir de " -"um local remoto, desde que o seu provedor de Internet lhe forneça um " -"endereço IP público. Você poderá ligar para outros usuários locais sem custo " -"(por exemplo, outros adaptadores de telefone analógico (ATAs)) e usar seus " -"provedores de VoIP para fazer chamadas como se fossem originadas do local do " -"seu PBX. Depois de configurar esta aba, volte para onde os usuários são " -"configurados e veja as novas configurações de servidor e porta com as quais " -"você precisa configurar os seus dispositivos SIP remotos. Por favor, note " -"que se este PABX não está rodando no seu roteador, você terá que configurar " -"o redirecionamento de portas (NAT) no seu roteador. Por favor, encaminhe as " -"portas abaixo (porta SIP e intervalo de porta RTP) para o endereço IP do " -"dispositivo que executa este PBX." - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" -"Seu PIN desaparece deste campo quando for salvo e não será exibido para sua " -"proteção. Ele será alterada somente quando você informar um PIN diferente do " -"que foi salvo anteriormente. É possível deixá-lo em branco mas fique atento " -"quanto as implicações na segurança." - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" -"Sua senha desaparece deste campo quando for salva e não será exibida para " -"sua proteção. A senha será alterada somente quando você informar uma nova " -"senha diferente da que foi salva anteriormente." - -#~ msgid "" -#~ "Designate numbers that are allowed to call through this system and which " -#~ "user's privileges it will have." -#~ msgstr "" -#~ "Números definidos que poderão realizar chamadas através deste sistema e " -#~ "quais privilégios o usuário terá." - -#~ msgid "" -#~ "Pick a random port number between 6500 and 9500 for the service to listen " -#~ "on. Do not pick the standard 5060, because it is often subject to brute-" -#~ "force attacks. When finished, (1) click \"Save and Apply\", and (2) click " -#~ "the \"Restart VoIP Service\" button above. Finally, (3) look in the \"SIP " -#~ "Device/Softphone Accounts\" section for updated Server and Port settings " -#~ "for your SIP Devices/Softphones." -#~ msgstr "" -#~ "Escolha um número de porta aleatória entre 6500 e 9500 para o serviço " -#~ "escutar. Não escolher o padrão 5060, porque é frequentemente alvo de ataques " -#~ "de força bruta. Quando terminar, (1) clique em \"Salvar e Aplicar\", e (2) " -#~ "clique no \"Reiniciar o serviço VoIP\" acima. Finalmente, (3) olhe na seção " -#~ "\"Contas de Dispositivos SIP/Telefones em Software\" para atualizar o endereço " -#~ "e porta do servidor para seu Dispositivos SIP/Telefones em Software." - -#~ msgid "" -#~ "You can enter your domain name, external IP address, or dynamic domain " -#~ "name here Please keep in mind that if your IP address is dynamic and it " -#~ "changes your configuration will become invalid. Hence, it's recommended " -#~ "to set up Dynamic DNS in this case." -#~ msgstr "" -#~ "Você pode digitar aqui o seu nome de domínio, endereço IP externo, ou nome " -#~ "de domínio dinâmico. Tenha em mente que se o seu endereço IP é dinâmico e " -#~ "ele mudar, a sua configuração se tornará inválida. Por isso, é recomendado " -#~ "configurar um DNS dinâmico neste caso." - -#~ msgid "Account Status" -#~ msgstr "Estado da Conta" - -#~ msgid "Account Status Message" -#~ msgstr "Mensagem do Estado da Conta" - -#~ msgid "Domain Name/Dynamic Domain Name" -#~ msgstr "Nome do Domínio/Nome do Domínio Dinâmico" - -#~ msgid "Enable Incoming Calls (See Status, Message below)" -#~ msgstr "Habilitar Chamadas Recebidas (Veja o Estado, Mensagem abaixo)" - -#~ msgid "Service Control and Connection Status" -#~ msgstr "Controle do Serviço e Estado da Conexão" diff --git a/applications/luci-app-pbx/po/pt/pbx.po b/applications/luci-app-pbx/po/pt/pbx.po deleted file mode 100644 index 75b6c8cd1a..0000000000 --- a/applications/luci-app-pbx/po/pt/pbx.po +++ /dev/null @@ -1,487 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"PO-Revision-Date: 2013-09-22 19:17+0200\n" -"Last-Translator: Low <pedroloureiro1@sapo.pt>\n" -"Language-Team: none\n" -"Language: pt\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=2; plural=(n != 1);\n" -"X-Generator: Pootle 2.0.6\n" - -msgid "Advanced Settings" -msgstr "" - -msgid "Available" -msgstr "Disponível" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" - -msgid "Away" -msgstr "" - -msgid "Blacklisted Numbers" -msgstr "" - -msgid "Call Routing" -msgstr "" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "" - -msgid "Email" -msgstr "" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "" - -msgid "Enable Outgoing Calls" -msgstr "" - -msgid "Enabled" -msgstr "Ativado" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "Nome Completo" - -msgid "General Settings" -msgstr "" - -msgid "Google Accounts" -msgstr "" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "" - -msgid "Insert QoS Rules" -msgstr "" - -msgid "Makes Outgoing Calls" -msgstr "" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "Não" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "" - -msgid "PBX Main Page" -msgstr "" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "" - -msgid "Password" -msgstr "" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "Sim" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" diff --git a/applications/luci-app-pbx/po/ro/pbx.po b/applications/luci-app-pbx/po/ro/pbx.po deleted file mode 100644 index 49e8daccf4..0000000000 --- a/applications/luci-app-pbx/po/ro/pbx.po +++ /dev/null @@ -1,488 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"PO-Revision-Date: 2014-06-28 18:50+0200\n" -"Last-Translator: xxvirusxx <condor20_05@yahoo.it>\n" -"Language-Team: none\n" -"Language: ro\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=3; plural=(n==1 ? 0 : (n==0 || (n%100 > 0 && n%100 < " -"20)) ? 1 : 2);;\n" -"X-Generator: Pootle 2.0.6\n" - -msgid "Advanced Settings" -msgstr "Setări avansate" - -msgid "Available" -msgstr "Disponibil" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" - -msgid "Away" -msgstr "" - -msgid "Blacklisted Numbers" -msgstr "" - -msgid "Call Routing" -msgstr "" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "Nu deranjaţi" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "Domeniu/Adresă IP/Domeniu dinamic" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "" - -msgid "Email" -msgstr "" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "" - -msgid "Enable Outgoing Calls" -msgstr "" - -msgid "Enabled" -msgstr "Activat" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "Nume complet" - -msgid "General Settings" -msgstr "Setări generale" - -msgid "Google Accounts" -msgstr "Conturi Google" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "" - -msgid "Insert QoS Rules" -msgstr "" - -msgid "Makes Outgoing Calls" -msgstr "" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "" - -msgid "PBX Main Page" -msgstr "" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "" - -msgid "Password" -msgstr "Parolă" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "Setări QoS" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" diff --git a/applications/luci-app-pbx/po/ru/pbx.po b/applications/luci-app-pbx/po/ru/pbx.po deleted file mode 100644 index e85c947e1a..0000000000 --- a/applications/luci-app-pbx/po/ru/pbx.po +++ /dev/null @@ -1,525 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"PO-Revision-Date: 2013-09-06 10:28+0200\n" -"Last-Translator: datasheet <michael.gritsaenko@gmail.com>\n" -"Language-Team: none\n" -"Language: ru\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=3; plural=(n%10==1 && n%100!=11 ? 0 : n%10>=2 && n%" -"10<=4 && (n%100<10 || n%100>=20) ? 1 : 2);\n" -"X-Generator: Pootle 2.0.6\n" - -msgid "Advanced Settings" -msgstr "Расширенные установки" - -msgid "Available" -msgstr "Доступен" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" -"Старайтесь не использовать ничего, кроме алфавитно-цифровых символов, " -"пробелов, запятых и точек." - -msgid "Away" -msgstr "Отошел" - -msgid "Blacklisted Numbers" -msgstr "Номера в \"черном\" списке" - -msgid "Call Routing" -msgstr "Маршрутизация вызовов" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "Номера сквозных вызовов" - -msgid "Copy-paste large lists of numbers here." -msgstr "Вставьте большие списки номеров здесь" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "Не беспокоить" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "Динамический список запрещенных номеров" - -msgid "Email" -msgstr "Эл. почта" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "Разрешить входящие вызовы (регистрация через SIP)" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "Разрешить входящие звонки (см. ниже Статус)" - -msgid "Enable Outgoing Calls" -msgstr "Разрешить исходящие вызовы" - -msgid "Enabled" -msgstr "Включено" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" -"Введите телефонные номера, звонки с которых вы хотите автоматически " -"отклонять. Вы, вероятно, не должны вводить код страны и ведущие нули, но, " -"чтобы удостовериться в этом, пожалуйста проверьте, что звонки из " -"нежелательной зоны успешно блокируются." - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" -"Введите этот IP (или IP порт) в установках Сервера/Регистратора SIP " -"устройств, который вы будете использовать ТОЛЬКО локально и никогда удаленно." - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" -"Введите это имя_хоста (или имя_хоста:порт) в установках Сервера/Регистратора " -"тех SIP-устройств, которые вы будете использовать удаленно (локально они " -"также будут работать)." - -msgid "External SIP Port" -msgstr "Внешний порт SIP" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "Полное имя" - -msgid "General Settings" -msgstr "Общие установки" - -msgid "Google Accounts" -msgstr "Учетные записи Google" - -msgid "Google Talk Status" -msgstr "Статус Google Talk" - -msgid "Google Talk Status Message" -msgstr "Сообщение статуса Google Talk" - -msgid "Google Voice/Talk Accounts" -msgstr "Учетные записи Google Voice/Talk" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "Входящие вызовы" - -msgid "Insert QoS Rules" -msgstr "Вставить правила QoS" - -msgid "Makes Outgoing Calls" -msgstr "Совершает исходящие вызовы" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "Нет" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "Outbound прокси сервер" - -msgid "Outgoing Calls" -msgstr "Исходящие вызовы" - -msgid "PBX Main Page" -msgstr "Главная страница АТС" - -#, fuzzy -msgid "PBX Service Status" -msgstr "Состояние службы АТС" - -msgid "PIN" -msgstr "PIN" - -msgid "Password" -msgstr "Пароль" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "Настройки порта устройств SIP" - -msgid "Providers Used for Outgoing Calls" -msgstr "Провайдеры исходящих вызовов" - -msgid "QoS Settings" -msgstr "Установки QoS" - -msgid "RTP Port Range End" -msgstr "Конец диапазона портов RTP" - -msgid "RTP Port Range Start" -msgstr "Начало диапазоно портов RTP" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "Принимает входящие вызовы" - -msgid "Remote Usage" -msgstr "Удаленное использование" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "Учетные записи SIP" - -msgid "SIP Device/Softphone Accounts" -msgstr "Учетные записи SIP устройства/программного телефона" - -msgid "SIP Provider Accounts" -msgstr "Учетные записи SIP провайдера" - -msgid "SIP Realm (needed by some providers)" -msgstr "SIP Realm (нужен для некоторых провайдеров)" - -msgid "SIP Server/Registrar" -msgstr "SIP Сервер/Регистратор" - -msgid "SIP Server/Registrar Port" -msgstr "Порт SIP Сервера/Регистратора" - -msgid "Server Setting" -msgstr "Настройки сервера" - -msgid "Server Setting for Local SIP Devices" -msgstr "Установки сервера для локальных SIP устройств" - -msgid "Server Setting for Remote SIP Devices" -msgstr "Настройки сервера для удаленных SIP устройств" - -msgid "Service Status" -msgstr "Состояние сервиса" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "Черный список номеров (пробел между номерами для разделения)" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" -"Укажите отдельные номера. Нажмите enter, чтобы добавить больше номеров." - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" -"Использовать эту учетную запись для исходящих вызовов в соответстии с " -"наcтройками секции \"Маршрутизация вызовов\"." - -msgid "Use this account to make outgoing calls." -msgstr "Использовать эту учетную запись для исходящих вызовов" - -msgid "User Accounts" -msgstr "Учетные записи пользователя" - -msgid "User Agent String" -msgstr "Строка агента пользователя" - -msgid "User Name" -msgstr "Имя пользователя" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "Да" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "Здесь Вы можете указать имя для отображения вместо ID звонящего." - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -#~ msgid "" -#~ "Designate numbers that are allowed to call through this system and which " -#~ "user's privileges it will have." -#~ msgstr "" -#~ "Указать телефонные номера, которым разрешено осуществлять звонки через эту " -#~ "систему, а также какими они будут обладать пользовательскими привилегиями." - -#~ msgid "Account Status" -#~ msgstr "Статус учетной записи" - -#~ msgid "Account Status Message" -#~ msgstr "Статус сообщение учетной записи" - -#~ msgid "Domain Name/Dynamic Domain Name" -#~ msgstr "Имя домена/Динамическое имя домена" - -#~ msgid "Enable Incoming Calls (See Status, Message below)" -#~ msgstr "Разрешить входящие вызовы (см. статус, сообщение ниже)" - -#~ msgid "Service Control and Connection Status" -#~ msgstr "Управление сервисом и статус соединения" diff --git a/applications/luci-app-pbx/po/sk/pbx.po b/applications/luci-app-pbx/po/sk/pbx.po deleted file mode 100644 index 7b6d4a5c64..0000000000 --- a/applications/luci-app-pbx/po/sk/pbx.po +++ /dev/null @@ -1,484 +0,0 @@ -msgid "" -msgstr "" -"Content-Type: text/plain; charset=UTF-8\n" -"Project-Id-Version: PACKAGE VERSION\n" -"Last-Translator: Automatically generated\n" -"Language-Team: none\n" -"MIME-Version: 1.0\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=3; plural=(n==1) ? 0 : (n>=2 && n<=4) ? 1 : 2;\n" - -msgid "Advanced Settings" -msgstr "" - -msgid "Available" -msgstr "" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" - -msgid "Away" -msgstr "" - -msgid "Blacklisted Numbers" -msgstr "" - -msgid "Call Routing" -msgstr "" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "" - -msgid "Email" -msgstr "" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "" - -msgid "Enable Outgoing Calls" -msgstr "" - -msgid "Enabled" -msgstr "" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "" - -msgid "General Settings" -msgstr "" - -msgid "Google Accounts" -msgstr "" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "" - -msgid "Insert QoS Rules" -msgstr "" - -msgid "Makes Outgoing Calls" -msgstr "" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "" - -msgid "PBX Main Page" -msgstr "" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "" - -msgid "Password" -msgstr "" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" diff --git a/applications/luci-app-pbx/po/sv/pbx.po b/applications/luci-app-pbx/po/sv/pbx.po deleted file mode 100644 index 400289b6bb..0000000000 --- a/applications/luci-app-pbx/po/sv/pbx.po +++ /dev/null @@ -1,506 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"PO-Revision-Date: 2014-04-28 06:11+0200\n" -"Last-Translator: Umeaboy <kristoffer.grundstrom1983@gmail.com>\n" -"Language-Team: none\n" -"Language: sv\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=2; plural=(n != 1);\n" -"X-Generator: Pootle 2.0.6\n" - -msgid "Advanced Settings" -msgstr "Avancerade inställningar" - -msgid "Available" -msgstr "Tillgänglig" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" -"Undvik att använda allt förutom alfa-numeriska karaktärer, mellanslag, komma-" -"tecken och punkt." - -msgid "Away" -msgstr "Borta" - -msgid "Blacklisted Numbers" -msgstr "Svartlistade nummer" - -msgid "Call Routing" -msgstr "" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "Kopiera och klistra in ett stort antal nummer här." - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "Ringer upp nummer som inte passar någon annanstans" - -msgid "Do Not Disturb" -msgstr "Stör ej" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "Domän/IP-adress/Dynamisk domän" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "Dynamisk lista över svartlistade nummer" - -msgid "Email" -msgstr "E-post" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "Aktivera inkommande samtal (Registrera via SIP)" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "Aktivera inkommande samtal (se status nedanför)" - -msgid "Enable Outgoing Calls" -msgstr "Aktivera utgående samtal" - -msgid "Enabled" -msgstr "Aktiverat" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" -"Ange telefonnummer som du vill neka samtal från automatiskt. Du borde " -"förmodligen utesluta landskoden och eventuella inledande nollor, men " -"experimentera gärna för att vara säker på att du lyckas blockera nummer från " -"ditt önskade område." - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" -"Ange den här IP:n (eller IP:port) i Server/Registrar-inställningarna för SIP-" -"enheter som du endast kommer att använda LOKALT och aldrig från en " -"fjärrstyrd anslutning." - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" -"Ange det här värdnamnet (eller värdnamn:port) under Server/Registrar " -"inställningen för SIP-enheten som du kommer att använda från en fjärrstyrd " -"plats (de kommer att fungera lokalt också)." - -msgid "External SIP Port" -msgstr "Extern SIP-port" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "Fullständigt namn" - -msgid "General Settings" -msgstr "Allmänna inställningar" - -msgid "Google Accounts" -msgstr "Google-konton" - -msgid "Google Talk Status" -msgstr "Status för Google Talk" - -msgid "Google Talk Status Message" -msgstr "Statusmeddelande för Google Talk" - -msgid "Google Voice/Talk Accounts" -msgstr "Google Voice/Talk-konton" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "Inkommande samtal" - -msgid "Insert QoS Rules" -msgstr "För in QoS-regler" - -msgid "Makes Outgoing Calls" -msgstr "Gör utgående samtal" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "NOTERA: Det finns inga lokala användarkonton konfigurerade." - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" -"NOTERA: Det finns inga lokala användar-konton aktiverade för utgående samtal." - -msgid "No" -msgstr "Nej" - -msgid "Number of Seconds to Ring" -msgstr "Antal sekunder att ringa" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "Utgående samtal" - -msgid "PBX Main Page" -msgstr "Huvudsida för PBX" - -msgid "PBX Service Status" -msgstr "Status för PBX-tjänsten" - -msgid "PIN" -msgstr "PIN-kod" - -msgid "Password" -msgstr "Lösenord" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "Port-inställning för SIP-enheter" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "QoS-inställningar" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "Tar emot inkommande samtal" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "Ringer användare som är aktiverade för inkommande samtal" - -msgid "SIP Accounts" -msgstr "SIP-konton" - -msgid "SIP Device/Softphone Accounts" -msgstr "SIP-enhet/Softphone-konton" - -msgid "SIP Provider Accounts" -msgstr "SIP-operatörskonton" - -msgid "SIP Realm (needed by some providers)" -msgstr "SIP-sfär (behövs av vissa operatörer)" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "Server-inställning" - -msgid "Server Setting for Local SIP Devices" -msgstr "Server-inställning för lokala SIP-enheter" - -msgid "Server Setting for Remote SIP Devices" -msgstr "Server-inställning för fjärrstyrda SIP-enheter" - -msgid "Service Status" -msgstr "Status för tjänst" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" -"Specificera nummer individuellt här. Tryck på enter-knappen för att lägga " -"till fler nummer." - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" -"Det här valet borde vara inställt på \"Ja\" om du har ett DID (riktigt " -"telefonnummer) associerat med det här SIP-kontot eller om du vill ta emot " -"SIP uri-samtal via den här operatören." - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "Använd det här kontot för att göra utgående samtal." - -msgid "User Accounts" -msgstr "Användar-konton" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "Användarnamn" - -msgid "Uses providers enabled for outgoing calls" -msgstr "Använder operatörer för utgående samtal" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "Ja" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" -"Du kan specifiera ett riktigt namn som visas i samband med nummret här." - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" diff --git a/applications/luci-app-pbx/po/templates/pbx.pot b/applications/luci-app-pbx/po/templates/pbx.pot deleted file mode 100644 index 86dd2eb72d..0000000000 --- a/applications/luci-app-pbx/po/templates/pbx.pot +++ /dev/null @@ -1,477 +0,0 @@ -msgid "" -msgstr "Content-Type: text/plain; charset=UTF-8" - -msgid "Advanced Settings" -msgstr "" - -msgid "Available" -msgstr "" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" - -msgid "Away" -msgstr "" - -msgid "Blacklisted Numbers" -msgstr "" - -msgid "Call Routing" -msgstr "" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "" - -msgid "Email" -msgstr "" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "" - -msgid "Enable Outgoing Calls" -msgstr "" - -msgid "Enabled" -msgstr "" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "" - -msgid "General Settings" -msgstr "" - -msgid "Google Accounts" -msgstr "" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "" - -msgid "Insert QoS Rules" -msgstr "" - -msgid "Makes Outgoing Calls" -msgstr "" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "" - -msgid "PBX Main Page" -msgstr "" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "" - -msgid "Password" -msgstr "" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" diff --git a/applications/luci-app-pbx/po/tr/pbx.po b/applications/luci-app-pbx/po/tr/pbx.po deleted file mode 100644 index 59af3e878d..0000000000 --- a/applications/luci-app-pbx/po/tr/pbx.po +++ /dev/null @@ -1,484 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"Last-Translator: Automatically generated\n" -"Language-Team: none\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=1; plural=0;\n" - -msgid "Advanced Settings" -msgstr "" - -msgid "Available" -msgstr "" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" - -msgid "Away" -msgstr "" - -msgid "Blacklisted Numbers" -msgstr "" - -msgid "Call Routing" -msgstr "" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "" - -msgid "Email" -msgstr "" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "" - -msgid "Enable Outgoing Calls" -msgstr "" - -msgid "Enabled" -msgstr "" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "" - -msgid "General Settings" -msgstr "" - -msgid "Google Accounts" -msgstr "" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "" - -msgid "Insert QoS Rules" -msgstr "" - -msgid "Makes Outgoing Calls" -msgstr "" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "" - -msgid "PBX Main Page" -msgstr "" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "" - -msgid "Password" -msgstr "" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" diff --git a/applications/luci-app-pbx/po/uk/pbx.po b/applications/luci-app-pbx/po/uk/pbx.po deleted file mode 100644 index d65a784435..0000000000 --- a/applications/luci-app-pbx/po/uk/pbx.po +++ /dev/null @@ -1,501 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"PO-Revision-Date: 2013-08-13 15:47+0200\n" -"Last-Translator: zubr_139 <zubr139@ukr.net>\n" -"Language-Team: none\n" -"Language: uk\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=3; plural=(n%10==1 && n%100!=11 ? 0 : n%10>=2 && n%" -"10<=4 && (n%100<10 || n%100>=20) ? 1 : 2);\n" -"X-Generator: Pootle 2.0.6\n" - -msgid "Advanced Settings" -msgstr "Розширені налаштування" - -msgid "Available" -msgstr "Доступний" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" -"Намагайтеся не використовувати нічого, крім алфавітно-цифрових символів, " -"пропусків, ком і крапок." - -msgid "Away" -msgstr "" - -msgid "Blacklisted Numbers" -msgstr "" - -msgid "Call Routing" -msgstr "Маршрутизація Викликів" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "Виклик через номери" - -msgid "Copy-paste large lists of numbers here." -msgstr "" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -#, fuzzy -msgid "Do Not Disturb" -msgstr "Не турбувати" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -#, fuzzy -msgid "Dynamic List of Blacklisted Numbers" -msgstr "Динамічний список небажаних дзвінків" - -msgid "Email" -msgstr "Електронна скринька" - -#, fuzzy -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "Активувати вхідні дзвінки (зареєструватися через SIP)" - -#, fuzzy -msgid "Enable Incoming Calls (set Status below)" -msgstr "Активувати вхідні дзвінки (Встановити низький статус)" - -msgid "Enable Outgoing Calls" -msgstr "Активувати вихідні виклики" - -msgid "Enabled" -msgstr "Активувати" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -#, fuzzy -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" -"Введіть цей IP (або IP:порт) Сервера/Реєстратор налаштування SIP пристрою ви " -"будете використовувати тільки локально й ніколи з віддаленого місця." - -#, fuzzy -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" -"Введіть це хост ім'я (або ім'я хоста:порт) сервер/Реєстратор налаштування " -"SIP пристрою ви будете використовувати з віддаленого місця розташування " -"(воно також буде працювати локально)." - -msgid "External SIP Port" -msgstr "Зовнішній порт SIP" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "Повне Ім'я" - -msgid "General Settings" -msgstr "Загальні Налаштування" - -msgid "Google Accounts" -msgstr "Облікові записи Google" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "" - -msgid "Insert QoS Rules" -msgstr "" - -msgid "Makes Outgoing Calls" -msgstr "" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "Ні" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "" - -msgid "PBX Main Page" -msgstr "" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "" - -msgid "Password" -msgstr "" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "Облікові записи користувачів" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "Ім'я користувача" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "Так" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" diff --git a/applications/luci-app-pbx/po/vi/pbx.po b/applications/luci-app-pbx/po/vi/pbx.po deleted file mode 100644 index 59af3e878d..0000000000 --- a/applications/luci-app-pbx/po/vi/pbx.po +++ /dev/null @@ -1,484 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"Last-Translator: Automatically generated\n" -"Language-Team: none\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=1; plural=0;\n" - -msgid "Advanced Settings" -msgstr "" - -msgid "Available" -msgstr "" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" - -msgid "Away" -msgstr "" - -msgid "Blacklisted Numbers" -msgstr "" - -msgid "Call Routing" -msgstr "" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "" - -msgid "Email" -msgstr "" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "" - -msgid "Enable Outgoing Calls" -msgstr "" - -msgid "Enabled" -msgstr "" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "" - -msgid "General Settings" -msgstr "" - -msgid "Google Accounts" -msgstr "" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "" - -msgid "Insert QoS Rules" -msgstr "" - -msgid "Makes Outgoing Calls" -msgstr "" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "" - -msgid "PBX Main Page" -msgstr "" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "" - -msgid "Password" -msgstr "" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" diff --git a/applications/luci-app-pbx/po/zh-cn/pbx.po b/applications/luci-app-pbx/po/zh-cn/pbx.po deleted file mode 100644 index 45325b99c1..0000000000 --- a/applications/luci-app-pbx/po/zh-cn/pbx.po +++ /dev/null @@ -1,495 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"PO-Revision-Date: 2014-07-15 16:11+0200\n" -"Last-Translator: Tanyingyu <Tanyingyu@163.com>\n" -"Language-Team: none\n" -"Language: zh_CN\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=1; plural=0;\n" -"X-Generator: Pootle 2.0.6\n" - -msgid "Advanced Settings" -msgstr "高级设置" - -msgid "Available" -msgstr "可用" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "避免使用除字母，数字，空格，逗号和句号外的其他字符。" - -msgid "Away" -msgstr "外" - -msgid "Blacklisted Numbers" -msgstr "黑名单" - -msgid "Call Routing" -msgstr "呼叫路由" - -msgid "Call-back Numbers" -msgstr "回调数" - -msgid "Call-back Provider" -msgstr "回呼提供者" - -msgid "Call-through Numbers" -msgstr "通过数字呼叫" - -msgid "Copy-paste large lists of numbers here." -msgstr "复制粘贴数字大名单。" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "其他地方无法匹配拨号号码" - -msgid "Do Not Disturb" -msgstr "请勿打扰" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "域名/ IP地址/动态域名" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "动态黑名单号码列表" - -msgid "Email" -msgstr "电子邮件" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "允许电话呼入（SIP注册者）" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "允许电话呼入（下面设置状态）" - -msgid "Enable Outgoing Calls" -msgstr "允许电话外呼" - -msgid "Enabled" -msgstr "允许" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" -"输入您想自动屏蔽的电话号码。您应该忽略国家代码和任何前导零,但请测试来确保您成" -"功屏蔽了想要屏蔽的号码。" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" -"在SIP设备注册服务器中输入IP(或IP:端口)，仅在本地使用，不可以在远程使用。" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "外部SIP端口" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "全名" - -msgid "General Settings" -msgstr "通用设置" - -msgid "Google Accounts" -msgstr "google账号" - -msgid "Google Talk Status" -msgstr "google Talk状态" - -msgid "Google Talk Status Message" -msgstr "google Talk状态消息" - -msgid "Google Voice/Talk Accounts" -msgstr "Google Voice/Talk账号" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "呼入电话" - -msgid "Insert QoS Rules" -msgstr "插入QoS规则" - -msgid "Makes Outgoing Calls" -msgstr "安排外呼列表" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "注意:没有google或SIP提供者账户配置。" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "注意:没有google或SIP提供者账户允许呼入电话。" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "注意:没有google或SIP提供者账户允许外呼电话。" - -msgid "NOTE: There are no local user accounts configured." -msgstr "注意:没有本地用户设置。" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "注意:没有本地用户允许外呼电话。" - -msgid "No" -msgstr "不" - -msgid "Number of Seconds to Ring" -msgstr "多少秒振铃" - -msgid "Outbound Proxy" -msgstr "外呼代理" - -msgid "Outgoing Calls" -msgstr "外呼电话" - -msgid "PBX Main Page" -msgstr "PBX主页" - -msgid "PBX Service Status" -msgstr "PBX服务状态" - -msgid "PIN" -msgstr "PIN" - -msgid "Password" -msgstr "密码" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "SIP设备端口设置" - -msgid "Providers Used for Outgoing Calls" -msgstr "用于外呼电话的提供者" - -msgid "QoS Settings" -msgstr "QoS设置" - -msgid "RTP Port Range End" -msgstr "RTP结束端口" - -msgid "RTP Port Range Start" -msgstr "RTP起始端口" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "收到呼入电话" - -msgid "Remote Usage" -msgstr "远程使用" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "SIP账号" - -msgid "SIP Device/Softphone Accounts" -msgstr "SIP 设备/软电话账号" - -msgid "SIP Provider Accounts" -msgstr "SIP提供者账户" - -msgid "SIP Realm (needed by some providers)" -msgstr "SIP Realm(一些供应商需要）" - -msgid "SIP Server/Registrar" -msgstr "SIP注册服务器" - -msgid "SIP Server/Registrar Port" -msgstr "SIP注册服务器端口" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -#~ msgid "" -#~ "Designate numbers that are allowed to call through this system and which " -#~ "user's privileges it will have." -#~ msgstr "设定号码作为用户拥有使用交换机呼叫的权限。" diff --git a/applications/luci-app-pbx/po/zh-tw/pbx.po b/applications/luci-app-pbx/po/zh-tw/pbx.po deleted file mode 100644 index 603b9df585..0000000000 --- a/applications/luci-app-pbx/po/zh-tw/pbx.po +++ /dev/null @@ -1,507 +0,0 @@ -msgid "" -msgstr "" -"Project-Id-Version: PACKAGE VERSION\n" -"PO-Revision-Date: 2014-05-16 13:59+0200\n" -"Last-Translator: omnistack <omnistack@gmail.com>\n" -"Language-Team: none\n" -"Language: zh_TW\n" -"MIME-Version: 1.0\n" -"Content-Type: text/plain; charset=UTF-8\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=1; plural=0;\n" -"X-Generator: Pootle 2.0.6\n" - -msgid "Advanced Settings" -msgstr "進階設定" - -msgid "Available" -msgstr "可運用" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "除了字母數字字符，空格，逗號和句號其它一概不用." - -msgid "Away" -msgstr "離線" - -msgid "Blacklisted Numbers" -msgstr "列入黑名單號碼" - -msgid "Call Routing" -msgstr "路由呼叫" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "通話接通號碼" - -msgid "Copy-paste large lists of numbers here." -msgstr "號碼大型清單複製貼上此地" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "撥號它處號碼不符" - -msgid "Do Not Disturb" -msgstr "勿擾中" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "網域/IP位址/動態網域" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "黑名單動態列表" - -msgid "Email" -msgstr "郵件信箱" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "啟用來話呼叫(透過SIP註冊)" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "啟用來話呼叫(在下面設定狀態)" - -msgid "Enable Outgoing Calls" -msgstr "啟用外撥" - -msgid "Enabled" -msgstr "已啟用" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" -"打入您允許自動通話的號碼. 您或許可以忽略國碼和0字開頭, 但僅供實驗以確保期望區" -"的號碼被阻斷成功." - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" -"要設定SIP設備在Server/Registrar內打入IP(或IP:埠號)您僅能本地端使用絕不要打入" -"遠端位置" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" -"要設定SIP設備從遠端使用(在本地端也一樣能執行),在Server/Registrar內打入主機名" -"稱(或主機名稱:埠號)" - -msgid "External SIP Port" -msgstr "外部SIP埠號" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "全名" - -msgid "General Settings" -msgstr "一般設定" - -msgid "Google Accounts" -msgstr "Google帳戶" - -msgid "Google Talk Status" -msgstr "Google Talk狀態" - -msgid "Google Talk Status Message" -msgstr "Google Talk訊息狀態" - -msgid "Google Voice/Talk Accounts" -msgstr "Google 語音/簡訊帳戶" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "來電呼叫" - -msgid "Insert QoS Rules" -msgstr "插入QoS規則" - -msgid "Makes Outgoing Calls" -msgstr "開啟外撥" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "注意:尚缺Google或者SIP提供者帳戶被設置" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "注意:尚缺Google或者SIP供應商帳戶被啟用才能接收來電呼叫" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "注意:尚缺Google或者SIP供應商帳戶被啟用才能外撥." - -msgid "NOTE: There are no local user accounts configured." -msgstr "注意:尚未設置本地端帳戶" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "注意:啟用本地端帳戶才能外撥" - -msgid "No" -msgstr "不" - -msgid "Number of Seconds to Ring" -msgstr "響鈴秒數" - -msgid "Outbound Proxy" -msgstr "外連代理" - -msgid "Outgoing Calls" -msgstr "去電外撥" - -msgid "PBX Main Page" -msgstr "PBX總機主頁" - -msgid "PBX Service Status" -msgstr "PBX服務狀態" - -msgid "PIN" -msgstr "PIN碼" - -msgid "Password" -msgstr "密碼" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "SIP設備的埠號設置" - -msgid "Providers Used for Outgoing Calls" -msgstr "已採用的外撥供應商" - -msgid "QoS Settings" -msgstr "QoS語音品質設置" - -msgid "RTP Port Range End" -msgstr "RTP協定埠域結束" - -msgid "RTP Port Range Start" -msgstr "RTP協定埠域啟始" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "接受來電呼叫" - -msgid "Remote Usage" -msgstr "遠端啟用" - -msgid "Rings users enabled for incoming calls" -msgstr "來電呼叫時震鈴通知使用者" - -msgid "SIP Accounts" -msgstr "SIP帳戶" - -msgid "SIP Device/Softphone Accounts" -msgstr "SIP設備/軟體式手機帳戶" - -msgid "SIP Provider Accounts" -msgstr "SIP供應商帳戶" - -msgid "SIP Realm (needed by some providers)" -msgstr "SIP領域(某些供應商需用到)" - -msgid "SIP Server/Registrar" -msgstr "SIP伺服器/登記處" - -msgid "SIP Server/Registrar Port" -msgstr "SIP伺服器/登記埠" - -msgid "Server Setting" -msgstr "伺服器設置" - -msgid "Server Setting for Local SIP Devices" -msgstr "本地SIP設備的伺服器設置" - -msgid "Server Setting for Remote SIP Devices" -msgstr "遠端SIP設備的伺服器設置" - -msgid "Service Status" -msgstr "服務狀態" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "以空格分隔的黑名單號碼列表" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "在此指定獨立號碼. 按enter 可新增更多號碼" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "使用這個帳號外撥." - -msgid "User Accounts" -msgstr "使用者帳號" - -msgid "User Agent String" -msgstr "用戶代理字串" - -msgid "User Name" -msgstr "用戶名稱" - -msgid "Uses providers enabled for outgoing calls" -msgstr "採用供應商啟用以便外撥" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "是" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "您可以在此指定一個真實名稱以便顯示在來電ID" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" -"不管多遠,只要能取得ISP提供的公眾合法IP,依然可以使用這個系統的SIP設備/PC電話表" -"現一樣的好.您將能夠免費呼叫其他本地端用戶(e.g 其它類比電話機(ATAs)和使用您的" -"VoIP供應商講電話就像您在使用本地的PBX總機電話一樣.在設定這個標籤後, 需回到用" -"戶設定並且查看新的伺服器和埠設定以便能設定遠端SIP設備.請注意假如PBX總機若不在" -"您的路由器/GW上執行,您將必須在您的路由器/GW上設定埠轉發(NAT).在PBX主機上請轉" -"發下列(SIP埠+RTP所採用的範圍埠)埠號址定到跑PBX服務設備上的IP位址." - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" -"當存檔時為保護起見您的PIN碼將不會顯示. 除非您打入不同於原始存檔的值它才會變" -"更. 也可以把PIN碼保留空白, 但要提防會有安全的隱憂." - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" -"當存檔時為保護起見您的密碼將不會顯示. 除非您打入不同於原始存檔的值它才會變更." - -#~ msgid "" -#~ "Designate numbers that are allowed to call through this system and which " -#~ "user's privileges it will have." -#~ msgstr "依據系統和戶用的權限允許通話的指定號碼" diff --git a/applications/luci-app-pbx/root/etc/config/pbx b/applications/luci-app-pbx/root/etc/config/pbx deleted file mode 100644 index ca7c1669d0..0000000000 --- a/applications/luci-app-pbx/root/etc/config/pbx +++ /dev/null @@ -1 +0,0 @@ -config 'main' 'connection_status' diff --git a/applications/luci-app-pbx/root/etc/config/pbx-advanced b/applications/luci-app-pbx/root/etc/config/pbx-advanced deleted file mode 100644 index 39da6f880c..0000000000 --- a/applications/luci-app-pbx/root/etc/config/pbx-advanced +++ /dev/null @@ -1,5 +0,0 @@ -config 'settings' 'advanced' - option 'useragent' 'PBX' - option 'ringtime' '30' - option 'rtpstart' '19850' - option 'rtpend' '19900' diff --git a/applications/luci-app-pbx/root/etc/config/pbx-calls b/applications/luci-app-pbx/root/etc/config/pbx-calls deleted file mode 100644 index 822bd4a1be..0000000000 --- a/applications/luci-app-pbx/root/etc/config/pbx-calls +++ /dev/null @@ -1,7 +0,0 @@ -config 'call_routing' 'outgoing_calls' - -config 'call_routing' 'incoming_calls' - -config 'call_routing' 'providers_user_can_use' - -config 'call_routing' 'blacklisting' diff --git a/applications/luci-app-pbx/root/etc/config/pbx-google b/applications/luci-app-pbx/root/etc/config/pbx-google deleted file mode 100644 index e69de29bb2..0000000000 --- a/applications/luci-app-pbx/root/etc/config/pbx-google +++ /dev/null diff --git a/applications/luci-app-pbx/root/etc/config/pbx-users b/applications/luci-app-pbx/root/etc/config/pbx-users deleted file mode 100644 index a4277b1bfe..0000000000 --- a/applications/luci-app-pbx/root/etc/config/pbx-users +++ /dev/null @@ -1 +0,0 @@ -config 'user' 'server' diff --git a/applications/luci-app-pbx/root/etc/config/pbx-voip b/applications/luci-app-pbx/root/etc/config/pbx-voip deleted file mode 100644 index e69de29bb2..0000000000 --- a/applications/luci-app-pbx/root/etc/config/pbx-voip +++ /dev/null diff --git a/applications/luci-app-pbx/root/etc/init.d/pbx-asterisk b/applications/luci-app-pbx/root/etc/init.d/pbx-asterisk deleted file mode 100755 index e05ae11cd6..0000000000 --- a/applications/luci-app-pbx/root/etc/init.d/pbx-asterisk +++ /dev/null @@ -1,837 +0,0 @@ -#!/bin/sh /etc/rc.common -# -# Copyright 2011 Iordan Iordanov <iiordanov (AT) gmail.com> -# -# This file is part of luci-pbx. -# -# luci-pbx is free software: you can redistribute it and/or modify -# it under the terms of the GNU General Public License as published by -# the Free Software Foundation, either version 3 of the License, or -# (at your option) any later version. -# -# luci-pbx is distributed in the hope that it will be useful, -# but WITHOUT ANY WARRANTY; without even the implied warranty of -# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -# GNU General Public License for more details. -# -# You should have received a copy of the GNU General Public License -# along with luci-pbx. If not, see <http://www.gnu.org/licenses/>. - -. /lib/functions.sh - -START=60 - -# Some global variables -MODULENAME=pbx -USERAGENT="PBX" -HANGUPCNTXT=hangup-call-context -GTALKUNVL=unavailable - -ASTUSER=nobody -ASTGROUP=nogroup -ASTDIRSRECURSIVE="/var/run/asterisk /var/log/asterisk /var/spool/asterisk" -ASTDIRS="/usr/lib/asterisk" -ASTSOUNDSDIR="/usr/lib/asterisk/sounds" - -TEMPLATEDIR=/etc/${MODULENAME}-asterisk -PBXSOUNDSDIR=$TEMPLATEDIR/sounds -VMTEMPLATEDIR=/etc/${MODULENAME}-voicemail -VMSOUNDSDIR=$VMTEMPLATEDIR/sounds -ASTERISKDIR=/etc/asterisk -WORKDIR=/tmp/$MODULENAME.$$ -MD5SUMSFILE=/tmp/$MODULENAME-sums.$$ - -TMPL_ASTERISK=$TEMPLATEDIR/asterisk.conf.TEMPLATE -TMPL_GTALK=$TEMPLATEDIR/gtalk.conf.TEMPLATE -TMPL_INDICATIONS=$TEMPLATEDIR/indications.conf.TEMPLATE -TMPL_LOGGER=$TEMPLATEDIR/logger.conf.TEMPLATE -TMPL_MANAGER=$TEMPLATEDIR/manager.conf.TEMPLATE -TMPL_MODULES=$TEMPLATEDIR/modules.conf.TEMPLATE -TMPL_RTP=$TEMPLATEDIR/rtp.conf.TEMPLATE - -TMPL_EXTCTHRUCHECKHDR=$TEMPLATEDIR/extensions_disa-check_header.conf.TEMPLATE -TMPL_EXTCTHRUCHECK=$TEMPLATEDIR/extensions_disa-check.conf.TEMPLATE -TMPL_EXTCTHRUCHECKFTR=$TEMPLATEDIR/extensions_disa-check_footer.conf.TEMPLATE -TMPL_EXTCTHRUHDR=$TEMPLATEDIR/extensions_disa_header.conf.TEMPLATE -TMPL_EXTCTHRU=$TEMPLATEDIR/extensions_disa.conf.TEMPLATE -TMPL_EXTCTHRUNOPIN=$TEMPLATEDIR/extensions_disa-nopin.conf.TEMPLATE - -TMPL_EXTCBACKCHECKHDR=$TEMPLATEDIR/extensions_callback-check_header.conf.TEMPLATE -TMPL_EXTCBACKCHECK=$TEMPLATEDIR/extensions_callback-check.conf.TEMPLATE -TMPL_EXTCBACKCHECKFTR=$TEMPLATEDIR/extensions_callback-check_footer.conf.TEMPLATE -TMPL_EXTCBACKHDR=$TEMPLATEDIR/extensions_callback_header.conf.TEMPLATE -TMPL_EXTCBACKSIP=$TEMPLATEDIR/extensions_callback_sip.conf.TEMPLATE -TMPL_EXTCBACKGTALK=$TEMPLATEDIR/extensions_callback_gtalk.conf.TEMPLATE - -TMPL_EXTENSIONS=$TEMPLATEDIR/extensions.conf.TEMPLATE - -TMPL_EXTVMDISABLED=$TEMPLATEDIR/extensions_voicemail_disabled.conf.TEMPLATE -TMPL_EXTVMENABLED=$TEMPLATEDIR/extensions_voicemail_enabled.conf.TEMPLATE - -TMPL_EXTBLKLIST=$TEMPLATEDIR/extensions_blacklist.conf.TEMPLATE -TMPL_EXTBLKLISTFTR=$TEMPLATEDIR/extensions_blacklist_footer.conf.TEMPLATE -TMPL_EXTBLKLISTHDR=$TEMPLATEDIR/extensions_blacklist_header.conf.TEMPLATE - -TMPL_EXTDEFAULT=$TEMPLATEDIR/extensions_default.conf.TEMPLATE -TMPL_EXTDEFAULTUSER=$TEMPLATEDIR/extensions_default_user.conf.TEMPLATE - -TMPL_EXTINCNTXTSIP=$TEMPLATEDIR/extensions_incoming_context_sip.conf.TEMPLATE -TMPL_EXTINCNTXTGTALKHDR=$TEMPLATEDIR/extensions_incoming_context_gtalk_header.conf.TEMPLATE -TMPL_EXTINCNTXTGTALK=$TEMPLATEDIR/extensions_incoming_context_gtalk.conf.TEMPLATE - -TMPL_EXTUSERCNTXT=$TEMPLATEDIR/extensions_user_context.conf.TEMPLATE -TMPL_EXTUSERCNTXTFTR=$TEMPLATEDIR/extensions_user_context_footer.conf.TEMPLATE -TMPL_EXTUSERCNTXTHDR=$TEMPLATEDIR/extensions_user_context_header.conf.TEMPLATE - -TMPL_EXTOUTHDR=$TEMPLATEDIR/extensions_default_outgoing_header.conf.TEMPLATE -TMPL_EXTOUTGTALK=$TEMPLATEDIR/extensions_outgoing_gtalk.conf.TEMPLATE -TMPL_EXTOUTLOCAL=$TEMPLATEDIR/extensions_outgoing_dial_local_user.conf.TEMPLATE -TMPL_EXTOUTSIP=$TEMPLATEDIR/extensions_outgoing_sip.conf.TEMPLATE - -TMPL_JABBER=$TEMPLATEDIR/jabber.conf.TEMPLATE -TMPL_JABBERUSER=$TEMPLATEDIR/jabber_users.conf.TEMPLATE -TMPL_SIP=$TEMPLATEDIR/sip.conf.TEMPLATE -TMPL_SIPPEER=$TEMPLATEDIR/sip_peer.TEMPLATE -TMPL_SIPREG=$TEMPLATEDIR/sip_registration.TEMPLATE -TMPL_SIPUSR=$TEMPLATEDIR/sip_user.TEMPLATE - -TMPL_MSMTPDEFAULT=$VMTEMPLATEDIR/pbx-msmtprc-defaults.TEMPLATE -TMPL_MSMTPACCOUNT=$VMTEMPLATEDIR/pbx-msmtprc-account.TEMPLATE -TMPL_MSMTPAUTH=$VMTEMPLATEDIR/pbx-msmtprc-account-auth.TEMPLATE -TMPL_MSMTPACCTDFLT=$VMTEMPLATEDIR/pbx-msmtprc-account-default.TEMPLATE - - -INCLUDED_FILES="$WORKDIR/extensions_blacklist.conf $WORKDIR/extensions_callthrough.conf\ - $WORKDIR/extensions_incoming.conf $WORKDIR/extensions_incoming_gtalk.conf\ - $WORKDIR/extensions_user.conf $WORKDIR/jabber_users.conf\ - $WORKDIR/sip_peers.conf $WORKDIR/sip_registrations.conf\ - $WORKDIR/sip_users.conf $WORKDIR/extensions_voicemail.conf\ - $WORKDIR/extensions_default.conf" - - -# In this string, we concatenate all local users enabled to receive calls -# readily formatted for the Dial command. -localusers_to_ring="" - -# In this string, we keep a list of all users that are enabled for outgoing -# calls. It is used at the end to create the user contexts. -localusers_can_dial="" - -# In this string, we put together a space-separated list of provider names -# (alphanumeric, with all non-alpha characters replaced with underscores), -# which will be used to dial out by default (whose outgoing contexts will -# be included in users' contexts by default. -outbound_providers="" -sip_outbound_providers="" -gtalk_outbound_providers="" - -# Function which escapes non-alpha-numeric characters in a string -escape_non_alpha() { - echo $@ | sed 's/\([^a-zA-Z0-9]\)/\\\1/g' -} - -# Function which replaces non-alpha-numeric characters with an underscore -sub_underscore_for_non_alpha() { - echo $@ | sed 's/[^a-zA-Z0-9]/_/g' -} - -# Copies the template files which we don't edit. -copy_unedited_templates_over() -{ - cp $TMPL_ASTERISK $WORKDIR/asterisk.conf - cp $TMPL_GTALK $WORKDIR/gtalk.conf - cp $TMPL_INDICATIONS $WORKDIR/indications.conf - cp $TMPL_LOGGER $WORKDIR/logger.conf - cp $TMPL_MANAGER $WORKDIR/manager.conf - cp $TMPL_MODULES $WORKDIR/modules.conf - # If this file isn't present at this stage, voicemail is disabled. - [ ! -f $WORKDIR/extensions_voicemail.conf ] && \ - cp $TMPL_EXTVMDISABLED $WORKDIR/extensions_voicemail.conf -} - -# Touches all the included files, to prevent asterisk from refusing to -# start if a config item is missing and an included config file isn't created. -create_included_files() -{ - touch $INCLUDED_FILES -} - -# Puts together all the extensions.conf related configuration. -pbx_create_extensions_config() -{ - local ringtime - config_get ringtime advanced ringtime - - sed "s/|RINGTIME|/$ringtime/" $TMPL_EXTENSIONS > $WORKDIR/extensions.conf - mv $WORKDIR/inext.TMP $WORKDIR/extensions_incoming.conf - cp $TMPL_EXTINCNTXTGTALKHDR $WORKDIR/extensions_incoming_gtalk.conf - cat $WORKDIR/outextgtalk.TMP >> $WORKDIR/extensions_incoming_gtalk.conf 2>/dev/null - rm -f $WORKDIR/outextgtalk.TMP - mv $WORKDIR/blacklist.TMP $WORKDIR/extensions_blacklist.conf - mv $WORKDIR/userext.TMP $WORKDIR/extensions_user.conf - - cp $TMPL_EXTCTHRUHDR $WORKDIR/extensions_callthrough.conf - cat $WORKDIR/callthrough.TMP >> $WORKDIR/extensions_callthrough.conf 2>/dev/null - rm -f $WORKDIR/callthrough.TMP - cat $TMPL_EXTCTHRUCHECKHDR >> $WORKDIR/extensions_callthrough.conf 2>/dev/null - cat $WORKDIR/callthroughcheck.TMP >> $WORKDIR/extensions_callthrough.conf 2>/dev/null - rm -f $WORKDIR/callthroughcheck.TMP - cat $TMPL_EXTCTHRUCHECKFTR >> $WORKDIR/extensions_callthrough.conf 2>/dev/null - - cp $TMPL_EXTCBACKHDR $WORKDIR/extensions_callback.conf - cat $WORKDIR/callback.TMP >> $WORKDIR/extensions_callback.conf 2>/dev/null - rm -f $WORKDIR/callback.TMP - cat $TMPL_EXTCBACKCHECKHDR >> $WORKDIR/extensions_callback.conf 2>/dev/null - cat $WORKDIR/callbackcheck.TMP >> $WORKDIR/extensions_callback.conf 2>/dev/null - rm -f $WORKDIR/callbackcheck.TMP - cat $TMPL_EXTCBACKCHECKFTR >> $WORKDIR/extensions_callback.conf 2>/dev/null - - rm -f $WORKDIR/outext-*.TMP - rm -f $WORKDIR/localext.TMP - sed "s/|LOCALUSERS|/$localusers_to_ring/g" $TMPL_EXTDEFAULT \ - > $WORKDIR/extensions_default.conf - cat $WORKDIR/inextuser.TMP >> $WORKDIR/extensions_default.conf - rm -f $WORKDIR/inextuser.TMP -} - -# Puts together all the sip.conf related configuration. -pbx_create_sip_config() -{ - mv $WORKDIR/sip_regs.TMP $WORKDIR/sip_registrations.conf - mv $WORKDIR/sip_peers.TMP $WORKDIR/sip_peers.conf - mv $WORKDIR/sip_users.TMP $WORKDIR/sip_users.conf -} - -# Creates the jabber.conf related config -pbx_create_jabber_config() -{ - cp $TMPL_JABBER $WORKDIR/jabber.conf - mv $WORKDIR/jabber.TMP $WORKDIR/jabber_users.conf -} - -# Gets rid of any config files from $ASTERISKDIR not found in $WORKDIR. -clean_up_asterisk_config_dir() -{ - for f in $ASTERISKDIR/* ; do - basef="`basename $f`" - if [ ! -e "$WORKDIR/$basef" ] ; then - rm -rf "$f" - fi - done -} - -# Compares md5sums of the config files in $WORKDIR to those -# in $ASTERISKDIR, and copies only changed files over to reduce -# wear on flash in embedded devices. -compare_configs_and_copy_changed() -{ - # First, compute md5sums of the config files in $WORKDIR. - cd $WORKDIR/ - md5sum * > $MD5SUMSFILE - - # Now, check the files in $ASTERISKDIR against the md5sums. - cd $ASTERISKDIR/ - changed_files="`md5sum -c $MD5SUMSFILE 2>/dev/null | fgrep ": FAILED" | awk -F: '{print $1}'`" - - rm -f $MD5SUMSFILE - - [ -z "$changed_files" ] && return - - # Now copy over the changed files. - for f in $changed_files ; do - cp "$WORKDIR/$f" "$ASTERISKDIR/$f" - done -} - -# Calls the functions that create the final config files -# Calls the function which compares which files have changed -# Puts the final touches on $ASTERISKDIR -# Gets rid of $WORKDIR -pbx_assemble_and_copy_config() -{ - mkdir -p $ASTERISKDIR - - copy_unedited_templates_over - create_included_files - pbx_create_extensions_config - pbx_create_sip_config - pbx_create_jabber_config - - touch $WORKDIR/features.conf - - # At this point, $WORKDIR should contain a complete, working config. - clean_up_asterisk_config_dir - - compare_configs_and_copy_changed - - [ ! -d $ASTERISKDIR/manager.d ] && mkdir -p $ASTERISKDIR/manager.d/ - - # Get rid of the working directory - rm -rf $WORKDIR/ -} - -# Creates configuration for a user and adds it to the temporary file that holds -# all users configured so far. -pbx_add_user() -{ - local fullname - local defaultuser - local rawdefaultuser - local secret - local ring - local can_call - - config_get fullname $1 fullname - fullname=`escape_non_alpha $fullname` - config_get rawdefaultuser $1 defaultuser - defaultuser=`escape_non_alpha $rawdefaultuser` - config_get secret $1 secret - secret=`escape_non_alpha $secret` - config_get ring $1 ring - config_get can_call $1 can_call - - [ -z "$defaultuser" -o -z "$secret" ] && return - [ -z "$fullname" ] && fullname="$defaultuser" - - sed "s/|DEFAULTUSER|/$defaultuser/g" $TMPL_SIPUSR > $WORKDIR/sip_user.tmp - - if [ "$can_call" = "yes" ] ; then - # Add user to list of all users that are allowed to make calls. - localusers_can_dial="$localusers_can_dial $rawdefaultuser" - sed -i "s/|CONTEXTNAME|/$defaultuser/g" $WORKDIR/sip_user.tmp - else - sed -i "s/|CONTEXTNAME|/$HANGUPCNTXT/g" $WORKDIR/sip_user.tmp - fi - - # Add this user's configuration to the temp file containing all user configs. - sed "s/|FULLNAME|/$fullname/" $WORKDIR/sip_user.tmp |\ - sed "s/|SECRET|/$secret/g" >> $WORKDIR/sip_users.TMP - - if [ "$ring" = "yes" ] ; then - if [ -z "$localusers_to_ring" ] ; then - localusers_to_ring="SIP\/$defaultuser" - else - localusers_to_ring="$localusers_to_ring\&SIP\/$defaultuser" - fi - fi - - # Add configuration which allows local users to call each other. - sed "s/|DEFAULTUSER|/$defaultuser/g" $TMPL_EXTOUTLOCAL >> $WORKDIR/localext.TMP - - # Add configuration which puts calls to users through the default - # context, so that blacklists and voicemail take effect for this user. - sed "s/|DEFAULTUSER|/$defaultuser/g" $TMPL_EXTDEFAULTUSER >> $WORKDIR/inextuser.TMP - - rm -f $WORKDIR/sip_user.tmp -} - -# Creates configuration for a Google account, and adds it to the temporary file that holds -# all accounts configured so far. -# Also creates the outgoing extensions which are used in users' outgoing contexts. -pbx_add_jabber() -{ - local username - local secret - local numprefix - local register - local make_outgoing_calls - local name - local users_to_ring - local status - local statusmessage - - config_get username $1 username - username=`escape_non_alpha $username` - config_get secret $1 secret - secret=`escape_non_alpha $secret` - #TODO: Is this really necessary here? Numprefix is retrieved below. - config_get numprefix $1 numprefix - config_get register $1 register - config_get make_outgoing_calls $1 make_outgoing_calls - config_get name $1 name - config_get status $1 status - status=`escape_non_alpha $status` - config_get statusmessage $1 statusmessage - statusmessage=`escape_non_alpha $statusmessage` - - [ -z "$username" -o -z "$secret" ] && return - - # Construct a jabber entry for this provider. - sed "s/|USERNAME|/$username/g" $TMPL_JABBERUSER |\ - sed "s/|NAME|/$name/g" > $WORKDIR/jabber.tmp - - if [ "$register" = yes ] ; then - # If this provider is enabled for incoming calls, we need to set the - # status of the user to something other than unavailable in order to receive calls. - sed -i "s/|STATUS|/$status/g" $WORKDIR/jabber.tmp - sed -i "s/|STATUSMESSAGE|/\"$statusmessage\"/g" $WORKDIR/jabber.tmp - - users_to_ring="`uci -q get ${MODULENAME}-calls.incoming_calls.$name`" - # If no users have been specified to ring, we ring all users enabled for incoming calls. - if [ -z "$users_to_ring" ] ; then - users_to_ring=$localusers_to_ring - else - # Else, we cook up a string formatted for the Dial command - # with the specified users (SIP/user1&SIP/user2&...). We do it - # with set, shift and a loop in order to be more tolerant of ugly whitespace - # messes entered by users. - set $users_to_ring - users_to_ring="SIP\/$1" && shift - for u in $@ ; do u=`escape_non_alpha $u` ; users_to_ring=$users_to_ring\\\&SIP\\\/$u ; done - fi - - # Now, we add this account to the gtalk incoming context. - sed "s/|USERNAME|/$username/g" $TMPL_EXTINCNTXTGTALK |\ - sed "s/|LOCALUSERS|/$users_to_ring/g" >> $WORKDIR/outextgtalk.TMP - else - sed -i "s/|STATUS|/$GTALKUNVL/g" $WORKDIR/jabber.tmp - sed -i "s/|STATUSMESSAGE|/\"\"/g" $WORKDIR/jabber.tmp - fi - - # Add this account's configuration to the temp file containing all account configs. - sed "s/|SECRET|/$secret/g" $WORKDIR/jabber.tmp >> $WORKDIR/jabber.TMP - - # If this provider is enabled for outgoing calls. - if [ "$make_outgoing_calls" = "yes" ] ; then - - numprefix="`uci -q get ${MODULENAME}-calls.outgoing_calls.$name`" - - # If no prefixes are specified, then we use "X" which matches any prefix. - [ -z "$numprefix" ] && numprefix="X" - - for p in $numprefix ; do - p=`escape_non_alpha $p` - sed "s/|NUMPREFIX|/$p/g" $TMPL_EXTOUTGTALK |\ - sed "s/|NAME|/$name/g" >> $WORKDIR/outext-$name.TMP - done - - # Add this provider to the list of enabled outbound providers. - if [ -z "$outbound_providers" ] ; then - outbound_providers="$name" - else - outbound_providers="$outbound_providers $name" - fi - - # Add this provider to the list of enabled gtalk outbound providers. - if [ -z "$gtalk_outbound_providers" ] ; then - gtalk_outbound_providers="$name" - else - gtalk_outbound_providers="$gtalk_outbound_providers $name" - fi - fi - - rm -f $WORKDIR/jabber.tmp -} - -# Creates configuration for a SIP provider account, and adds it to the temporary file that holds -# all accounts configured so far. -# Also creates the outgoing extensions which are used in users' outgoing contexts. -pbx_add_peer() -{ - local defaultuser - local secret - local host - local fromdomain - local register - local numprefix - local make_outgoing_calls - local name - local users_to_ring - local port - local outboundproxy - - config_get defaultuser $1 defaultuser - defaultuser=`escape_non_alpha $defaultuser` - config_get secret $1 secret - secret=`escape_non_alpha $secret` - config_get host $1 host - host=`escape_non_alpha $host` - config_get port $1 port - config_get outbountproxy $1 outboundproxy - outbountproxy=`escape_non_alpha $outbountproxy` - config_get fromdomain $1 fromdomain - fromdomain=`escape_non_alpha $fromdomain` - config_get register $1 register - config_get numprefix $1 numprefix - config_get make_outgoing_calls $1 make_outgoing_calls - config_get name $1 name - - [ -z "$defaultuser" -o -z "$secret" -o -z "$host" ] && return - [ -z "$fromdomain" ] && fromdomain=$host - [ -n "$port" ] && port="port=$port" - [ -n "$outboundproxy" ] && outboundproxy="outboundproxy=$outboundproxy" - - # Construct a sip peer entry for this provider. - sed "s/|DEFAULTUSER|/$defaultuser/" $TMPL_SIPPEER > $WORKDIR/sip_peer.tmp - sed -i "s/|NAME|/$name/" $WORKDIR/sip_peer.tmp - sed -i "s/|FROMUSER|/$defaultuser/" $WORKDIR/sip_peer.tmp - sed -i "s/|SECRET|/$secret/" $WORKDIR/sip_peer.tmp - sed -i "s/|HOST|/$host/" $WORKDIR/sip_peer.tmp - sed -i "s/|PORT|/$port/" $WORKDIR/sip_peer.tmp - sed -i "s/|OUTBOUNDPROXY|/$outboundproxy/" $WORKDIR/sip_peer.tmp - # Add this account's configuration to the temp file containing all account configs. - sed "s/|FROMDOMAIN|/$host/" $WORKDIR/sip_peer.tmp >> $WORKDIR/sip_peers.TMP - - # If this provider is enabled for incoming calls. - if [ "$register" = "yes" ] ; then - # Then we create a registration string for this provider. - sed "s/|DEFAULTUSER|/$defaultuser/g" $TMPL_SIPREG > $WORKDIR/sip_reg.tmp - sed -i "s/|SECRET|/$secret/g" $WORKDIR/sip_reg.tmp - sed "s/|NAME|/$name/g" $WORKDIR/sip_reg.tmp >> $WORKDIR/sip_regs.TMP - - users_to_ring="`uci -q get ${MODULENAME}-calls.incoming_calls.$name`" - # If no users have been specified to ring, we ring all users enabled for incoming calls. - if [ -z "$users_to_ring" ] ; then - users_to_ring=$localusers_to_ring - else - # Else, we cook up a string formatted for the Dial command - # with the specified users (SIP/user1&SIP/user2&...). We do it - # with set, shift and a loop in order to be more tolerant of ugly whitespace - # messes entered by users. - set $users_to_ring - users_to_ring="SIP\/$1" && shift - for u in $@ ; do users_to_ring=$users_to_ring\\\&SIP\\\/$u ; done - fi - - # And we create an incoming calls context for this provider. - sed "s/|NAME|/$name/g" $TMPL_EXTINCNTXTSIP |\ - sed "s/|LOCALUSERS|/$users_to_ring/g" >> $WORKDIR/inext.TMP - fi - - # If this provider is enabled for outgoing calls. - if [ "$make_outgoing_calls" = "yes" ] ; then - - numprefix="`uci -q get ${MODULENAME}-calls.outgoing_calls.$name`" - # If no prefixes are specified, then we use "X" which matches any prefix. - [ -z "$numprefix" ] && numprefix="X" - for p in $numprefix ; do - p=`escape_non_alpha $p` - sed "s/|NUMPREFIX|/$p/g" $TMPL_EXTOUTSIP |\ - sed "s/|NAME|/$name/g" >> $WORKDIR/outext-$name.TMP - done - - # Add this provider to the list of enabled outbound providers. - if [ -z "$outbound_providers" ] ; then - outbound_providers="$name" - else - outbound_providers="$outbound_providers $name" - fi - - # Add this provider to the list of enabled sip outbound providers. - if [ -z "$sip_outbound_providers" ] ; then - sip_outbound_providers="$name" - else - sip_outbound_providers="$sip_outbound_providers $name" - fi - fi - - rm -f $WORKDIR/sip_peer.tmp - rm -f $WORKDIR/sip_reg.tmp -} - -# For all local users enabled for outbound calls, creates a context -# containing the extensions for Google and SIP accounts this user is -# allowed to use. -pbx_create_user_contexts() -{ - local providers - - for u in $localusers_can_dial ; do - u=`escape_non_alpha $u` - sed "s/|DEFAULTUSER|/$u/g" $TMPL_EXTUSERCNTXTHDR >> $WORKDIR/userext.TMP - cat $WORKDIR/localext.TMP >> $WORKDIR/userext.TMP - providers="`uci -q get ${MODULENAME}-calls.providers_user_can_use.$u`" - [ -z "$providers" ] && providers="$outbound_providers" - - # For each provider, cat the contents of outext-$name.TMP into the user's outgoing calls extension - for p in $providers ; do - [ -f $WORKDIR/outext-$p.TMP ] && cat $WORKDIR/outext-$p.TMP >> $WORKDIR/userext.TMP - done - cat $TMPL_EXTUSERCNTXTFTR >> $WORKDIR/userext.TMP - done -} - -# Creates the blacklist context which hangs up on blacklisted numbers. -pbx_add_blacklist() -{ - local blacklist1 - local blacklist2 - - config_get blacklist1 blacklisting blacklist1 - config_get blacklist2 blacklisting blacklist2 - - # We create the blacklist context no matter whether the blacklist - # actually contains entries or not, since the PBX will send calls - # to the context for a check against the list anyway. - cp $TMPL_EXTBLKLISTHDR $WORKDIR/blacklist.TMP - for n in $blacklist1 $blacklist2 ; do - n=`escape_non_alpha $n` - sed "s/|BLACKLISTITEM|/$n/g" $TMPL_EXTBLKLIST >> $WORKDIR/blacklist.TMP - done - cat $TMPL_EXTBLKLISTFTR >> $WORKDIR/blacklist.TMP -} - -# Creates the callthrough context which allows specified numbers to get -# into the PBX and dial out as the configured user. -pbx_add_callthrough() -{ - local callthrough_number_list - local defaultuser - local pin - local enabled - local F - - config_get callthrough_number_list $1 callthrough_number_list - config_get defaultuser $1 defaultuser - defaultuser=`escape_non_alpha $defaultuser` - config_get pin $1 pin - pin=`escape_non_alpha $pin` - config_get enabled $1 enabled - - [ "$enabled" = "no" ] && return - [ "$defaultuser" = "" ] && return - - for callthrough_number in $callthrough_number_list ; do - sed "s/|NUMBER|/$callthrough_number/g" $TMPL_EXTCTHRUCHECK >> $WORKDIR/callthroughcheck.TMP - - if [ -n "$pin" ] ; then F=$TMPL_EXTCTHRU ; else F=$TMPL_EXTCTHRUNOPIN ; fi - sed "s/|NUMBER|/$callthrough_number/g" $F |\ - sed "s/|DEFAULTUSER|/$defaultuser/" |\ - sed "s/|PIN|/$pin/" >> $WORKDIR/callthrough.TMP - done -} - - -# Creates the callback context which allows specified numbers to get -# a callback into the PBX and dial out as the configured user. -pbx_add_callback() -{ - local callback_number_list - local defaultuser - local pin - local enabled - local callback_provider - local callback_hangup_delay - local FB - local FT - - config_get callback_number_list $1 callback_number_list - config_get defaultuser $1 defaultuser - defaultuser=`escape_non_alpha $defaultuser` - config_get pin $1 pin - pin=`escape_non_alpha $pin` - config_get enabled $1 enabled - config_get callback_provider $1 callback_provider - callback_provider=`sub_underscore_for_non_alpha $callback_provider` - config_get callback_hangup_delay $1 callback_hangup_delay - - [ "$enabled" = "no" ] && return - [ "$defaultuser" = "" ] && return - - # If the provider is a SIP provider, set the file to use to $TMPL_EXTCBACKSIP - # otherwise, set it to $TMPL_EXTCBACKGTALK - if echo $sip_outbound_providers | grep -q $callback_provider 2>/dev/null - then - FB=$TMPL_EXTCBACKSIP - else - FB=$TMPL_EXTCBACKGTALK - fi - - for callback_number in $callback_number_list ; do - sed "s/|NUMBER|/$callback_number/g" $TMPL_EXTCBACKCHECK >> $WORKDIR/callbackcheck.TMP - - sed "s/|NUMBER|/$callback_number/g" $FB |\ - sed "s/|CALLBACKPROVIDER|/$callback_provider/" |\ - sed "s/|CALLBACKHUPDELAY|/$callback_hangup_delay/" >> $WORKDIR/callback.TMP - - # Perhaps a bit confusingly, we create "callthrough" configuration for callback - # numbers, because we use the same DISA construct as for callthrough. - if [ -n "$pin" ] ; then FT=$TMPL_EXTCTHRU ; else FT=$TMPL_EXTCTHRUNOPIN ; fi - sed "s/|NUMBER|/$callback_number/g" $FT |\ - sed "s/|DEFAULTUSER|/$defaultuser/" |\ - sed "s/|PIN|/$pin/" >> $WORKDIR/callthrough.TMP - done -} - - -# Creates sip.conf from its template. -pbx_cook_sip_template() -{ - local useragent - local externhost - local bindport - - config_get useragent advanced useragent - useragent=`escape_non_alpha $useragent` - config_get externhost advanced externhost - config_get bindport advanced bindport - - [ -z "$useragent" ] && useragent="$USERAGENT" - - sed "s/|USERAGENT|/$useragent/g" $TMPL_SIP > $WORKDIR/sip.conf - - if [ -z "$externhost" ] ; then - sed -i "s/externhost=|EXTERNHOST|//g" $WORKDIR/sip.conf - else - sed -i "s/|EXTERNHOST|/$externhost/g" $WORKDIR/sip.conf - fi - - if [ -z "$bindport" ] ; then - sed -i "s/bindport=|BINDPORT|//g" $WORKDIR/sip.conf - else - sed -i "s/|BINDPORT|/$bindport/g" $WORKDIR/sip.conf - fi - - -} - -# Creates rtp.conf from its template. -pbx_cook_rtp_template() -{ - local rtpstart - local rtpend - - config_get rtpstart advanced rtpstart - config_get rtpend advanced rtpend - - sed "s/|RTPSTART|/$rtpstart/" $TMPL_RTP |\ - sed "s/|RTPEND|/$rtpend/" > $WORKDIR/rtp.conf -} - -# Links any sound files found in $PBXSOUNDSDIR and $VMSOUNDSDIR -# into $ASTSOUNDSDIR for use by Asterisk. Does not overwrite files. -pbx_link_sounds() -{ - mkdir -p $ASTSOUNDSDIR - - for dir in $PBXSOUNDSDIR $VMSOUNDSDIR ; do - if [ -d $dir ] ; then - for f in $dir/* ; do - ln -s $f $ASTSOUNDSDIR 2>/dev/null - done - fi - done -} - - -# Makes sure the ownership of specified directories is proper. -pbx_fix_ownership() -{ - chown $ASTUSER:$ASTGROUP $ASTDIRS - chown $ASTUSER:$ASTGROUP -R $ASTDIRSRECURSIVE -} - - -# Creates voicemail config if installed and enabled. -pbx_configure_voicemail() -{ - local enabled - local global_timeout - local global_email_addresses - - local smtp_tls - local smtp_server - local smtp_port - local smtp_auth - local smtp_user - local smtp_password - - config_get enabled global_voicemail enabled - - # First check if voicemail is enabled. - [ "$enabled" != "yes" ] && return - - config_get global_timeout global_voicemail global_timeout - #config_get global_email_addresses global_voicemail global_email_addresses - config_get smtp_auth voicemail_smtp smtp_auth - config_get smtp_tls voicemail_smtp smtp_tls - config_get smtp_server voicemail_smtp smtp_server - config_get smtp_port voicemail_smtp smtp_port - config_get smtp_user voicemail_smtp smtp_user - smtp_user=`escape_non_alpha $smtp_user` - config_get smtp_password voicemail_smtp smtp_password - smtp_password=`escape_non_alpha $smtp_password` - - sed "s/|AUTH|/$smtp_auth/" $TMPL_MSMTPDEFAULT |\ - sed "s/|TLS|/$smtp_tls/" > $WORKDIR/pbx-msmtprc - - sed "s/|HOST|/$smtp_server/" $TMPL_MSMTPACCOUNT |\ - sed "s/|PORT|/$smtp_port/" >> $WORKDIR/pbx-msmtprc - - if [ "$smtp_auth" = "on" ] ; then - sed "s/|USER|/$smtp_user/" $TMPL_MSMTPAUTH |\ - sed "s/|PASSWORD|/$smtp_password/" >> $WORKDIR/pbx-msmtprc - fi - - cat $TMPL_MSMTPACCTDFLT >> $WORKDIR/pbx-msmtprc - - [ ! -f /etc/pbx-msmtprc ] && cp $WORKDIR/pbx-msmtprc /etc/pbx-msmtprc - cmp -s $WORKDIR/pbx-msmtprc /etc/pbx-msmtprc 1>/dev/null \ - || mv $WORKDIR/pbx-msmtprc /etc/pbx-msmtprc - chmod 600 /etc/pbx-msmtprc - chown nobody /etc/pbx-msmtprc - - # Copy over the extensions file which has voicemail enabled. - cp $TMPL_EXTVMENABLED $WORKDIR/extensions_voicemail.conf - - # Create the voicemail directory in /tmp - mkdir -p /tmp/voicemail - chown nobody /tmp/voicemail - - # Create the recordings directory - mkdir -p /etc/pbx-voicemail/recordings - chown nobody /etc/pbx-voicemail/recordings - - # Working around a bug in OpenWRT 12.09-rc1 - # TODO: REMOVE AS SOON AS POSSIBLE - chmod ugo+w /tmp -} - - -start() { - mkdir -p $WORKDIR - - # Create the users. - config_load ${MODULENAME}-users - config_foreach pbx_add_user local_user - - # Create configuration for each google account. - config_unset - config_load ${MODULENAME}-google - config_foreach pbx_add_jabber gtalk_jabber - - # Create configuration for each voip provider. - config_unset - config_load ${MODULENAME}-voip - config_foreach pbx_add_peer voip_provider - - # Create the user contexts, callthroug/back, and phone blacklist. - config_unset - config_load ${MODULENAME}-calls - pbx_create_user_contexts - pbx_add_blacklist - config_foreach pbx_add_callthrough callthrough_numbers - config_foreach pbx_add_callback callback_numbers - - # Prepare sip.conf using settings from the "advanced" section. - config_unset - config_load ${MODULENAME}-advanced - pbx_cook_sip_template - pbx_cook_rtp_template - - # Prepare voicemail config. - config_unset - config_load ${MODULENAME}-voicemail - pbx_configure_voicemail - - # Assemble the configuration, and copy changed files over. - config_unset - config_load ${MODULENAME}-advanced - pbx_assemble_and_copy_config - - # Link sound files - pbx_link_sounds - - # Enforce ownership of specified files and directories. - pbx_fix_ownership -} diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE deleted file mode 100644 index ac5439615b..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE +++ /dev/null @@ -1,17 +0,0 @@ -[directories] -astetcdir => /etc/asterisk -astmoddir => /usr/lib/asterisk/modules -astvarlibdir => /usr/lib/asterisk -astdbdir => /usr/lib/asterisk -astkeydir => /usr/lib/asterisk -astdatadir => /usr/lib/asterisk -astagidir => /usr/lib/asterisk/agi-bin -astspooldir => /var/spool/asterisk -astrundir => /var/run/asterisk -astlogdir => /var/log/asterisk - -[options] -languageprefix = yes -dumpcore = no -runuser = nobody -rungroup = nogroup diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/delayedcallback b/applications/luci-app-pbx/root/etc/pbx-asterisk/delayedcallback deleted file mode 100755 index 903efe9ad9..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/delayedcallback +++ /dev/null @@ -1,18 +0,0 @@ -#!/bin/sh - -# Check if there are more than one instance of this command -# with the same command line running at the same time for some -# reason, then quit. We are checking for the same -# commandline in order to permit two different callback -# attempts simultaneously. - -if ! grep -q "$@" /dev/shm/delayedcallback.[0-9]* 2>/dev/null -then - echo "$@" > /dev/shm/delayedcallback.$$ - sleep 25 - asterisk -r -x "$1 $2 \"$3\" $4 $5 $6" - rm /dev/shm/delayedcallback.$$ -# echo "`date` $@": >> /dev/shm/delayedcallback.log -#else -# echo "`date` ERROR: There appears to be a callback attempt in progress to: $@" >> /dev/shm/delayedcallback.err -fi diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE deleted file mode 100644 index c8966edd87..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE +++ /dev/null @@ -1,25 +0,0 @@ -[general] -static = yes -writeprotect = yes -clearglobalvars = no - -[globals] -RINGTIME => |RINGTIME| - -[default] - -[context-user-hangup-call-context] -exten => s,1,Hangup() -exten => _X.,1,Hangup() - -[context-catch-all] -exten => _[!-~].,1,Dial(SIP/${EXTEN},60,r) - -#include extensions_default.conf -#include extensions_voicemail.conf -#include extensions_incoming.conf -#include extensions_incoming_gtalk.conf -#include extensions_blacklist.conf -#include extensions_callthrough.conf -#include extensions_callback.conf -#include extensions_user.conf diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist.conf.TEMPLATE deleted file mode 100644 index 54ee989b0f..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist.conf.TEMPLATE +++ /dev/null @@ -1 +0,0 @@ -exten => s,n,Gotoif($[ "${CALLERID(NUM)}" = "|BLACKLISTITEM|" ]?context-user-hangup,s,1) diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist_footer.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist_footer.conf.TEMPLATE deleted file mode 100644 index da964f2388..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist_footer.conf.TEMPLATE +++ /dev/null @@ -1,2 +0,0 @@ -exten => s,n,Goto(${SOURCECONTEXT},${SOURCEEXTEN},doneblacklist) - diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist_header.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist_header.conf.TEMPLATE deleted file mode 100644 index de0e984652..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist_header.conf.TEMPLATE +++ /dev/null @@ -1,3 +0,0 @@ - -[blacklist-call-context] -exten => s,1,Noop() diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check.conf.TEMPLATE deleted file mode 100644 index 06b1a4b6b9..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check.conf.TEMPLATE +++ /dev/null @@ -1 +0,0 @@ -exten => s,n,Gotoif($[ "${CALLERID(NUM)}" =~ ".*|NUMBER|" ]?context-user-callback,|NUMBER|,1) diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check_footer.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check_footer.conf.TEMPLATE deleted file mode 100644 index 282fe9e8ff..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check_footer.conf.TEMPLATE +++ /dev/null @@ -1,2 +0,0 @@ -exten => s,n,Goto(${SOURCECONTEXT},${SOURCEEXTEN},donecallback) - diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check_header.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check_header.conf.TEMPLATE deleted file mode 100644 index be289c4d33..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check_header.conf.TEMPLATE +++ /dev/null @@ -1,3 +0,0 @@ - -[callback-check-call-context] -exten => s,1,Noop() diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_gtalk.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_gtalk.conf.TEMPLATE deleted file mode 100644 index 43eec788f3..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_gtalk.conf.TEMPLATE +++ /dev/null @@ -1,4 +0,0 @@ -exten => |NUMBER|,1,System(/etc/pbx-asterisk/delayedcallback "channel originate Gtalk/gtalk-|CALLBACKPROVIDER|/|NUMBER|@voice.google.com extension |NUMBER|@disa-call-context" &) -exten => |NUMBER|,n,Wait(|CALLBACKHUPDELAY|) -exten => |NUMBER|,n,Hangup() - diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_header.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_header.conf.TEMPLATE deleted file mode 100644 index 0b8fb4c23f..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_header.conf.TEMPLATE +++ /dev/null @@ -1 +0,0 @@ -[context-user-callback] diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_sip.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_sip.conf.TEMPLATE deleted file mode 100644 index 300e9fa0e8..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_sip.conf.TEMPLATE +++ /dev/null @@ -1,4 +0,0 @@ -exten => |NUMBER|,1,System(/etc/pbx-asterisk/delayedcallback "channel originate SIP/|NUMBER|@peer-|CALLBACKPROVIDER| extension |NUMBER|@disa-call-context" &) -exten => |NUMBER|,n,Wait(|CALLBACKHUPDELAY|) -exten => |NUMBER|,n,Hangup() - diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_default.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_default.conf.TEMPLATE deleted file mode 100644 index 35836e290a..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_default.conf.TEMPLATE +++ /dev/null @@ -1,11 +0,0 @@ -[default-incoming-call-context] -exten => s,1,NoOp(${CALLERID}) -exten => s,n,Set(SOURCECONTEXT=default-incoming-call-context) -exten => s,n,Set(SOURCEEXTEN=s) -exten => s,n,Goto(blacklist-call-context,s,1) -exten => s,n(doneblacklist),NoOp() -exten => s,n,Goto(callback-check-call-context,s,1) -exten => s,n(donecallback),NoOp() -exten => s,n,Goto(disa-check-call-context,s,1) -exten => s,n(donedisacheck),Dial(|LOCALUSERS|,${RINGTIME},r) -exten => s,n,Goto(context-voicemail,s,1) diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_default_user.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_default_user.conf.TEMPLATE deleted file mode 100644 index 1910ff4d96..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_default_user.conf.TEMPLATE +++ /dev/null @@ -1 +0,0 @@ -exten => |DEFAULTUSER|,1,Goto(default-incoming-call-context,s,1) diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check.conf.TEMPLATE deleted file mode 100644 index ba2379b738..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check.conf.TEMPLATE +++ /dev/null @@ -1 +0,0 @@ -exten => s,n,Gotoif($[ "${CALLERID(NUM)}" =~ ".*|NUMBER|" ]?disa-call-context,|NUMBER|,1) diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check_footer.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check_footer.conf.TEMPLATE deleted file mode 100644 index 74056fa01d..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check_footer.conf.TEMPLATE +++ /dev/null @@ -1 +0,0 @@ -exten => s,n,Goto(${SOURCECONTEXT},${SOURCEEXTEN},donedisacheck) diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check_header.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check_header.conf.TEMPLATE deleted file mode 100644 index e0d67b8025..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check_header.conf.TEMPLATE +++ /dev/null @@ -1,2 +0,0 @@ -[disa-check-call-context] -exten => s,1,Noop() diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-nopin.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-nopin.conf.TEMPLATE deleted file mode 100644 index 74e48de8c1..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-nopin.conf.TEMPLATE +++ /dev/null @@ -1,5 +0,0 @@ -exten => |NUMBER|,1,Noop() -exten => |NUMBER|,n,Set(TIMEOUT(digit)=15) -exten => |NUMBER|,n,Set(TIMEOUT(response)=40) -exten => |NUMBER|,n,DISA(no-password,context-user-|DEFAULTUSER|) - diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa.conf.TEMPLATE deleted file mode 100644 index 3dd8fa35c9..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa.conf.TEMPLATE +++ /dev/null @@ -1,6 +0,0 @@ -exten => |NUMBER|,1,Noop() -exten => |NUMBER|,n,Set(TIMEOUT(digit)=7) -exten => |NUMBER|,n,Set(TIMEOUT(response)=21) -exten => |NUMBER|,n,Authenticate(|PIN|) -exten => |NUMBER|,n,DISA(no-password,context-user-|DEFAULTUSER|) - diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa_header.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa_header.conf.TEMPLATE deleted file mode 100644 index a742271146..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa_header.conf.TEMPLATE +++ /dev/null @@ -1 +0,0 @@ -[disa-call-context] diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk.conf.TEMPLATE deleted file mode 100644 index 3f9cf4c7d9..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk.conf.TEMPLATE +++ /dev/null @@ -1,15 +0,0 @@ -exten => |USERNAME|,1,NoOp(${CALLERID}) -same => n,Set(CALLERID(name)=${CUT(CALLERID(name),@,1)}) -same => n,GotoIf($["${CALLERID(name):0:2}" != "+1"]?notrim) -same => n,Set(CALLERID(name)=${CALLERID(name):2}) -same => n(notrim),Set(CALLERID(number)=${CALLERID(name)}) -same => n,Set(SOURCECONTEXT=context-incoming-gtalk) -same => n,Set(SOURCEEXTEN=|USERNAME|) -same => n,Goto(blacklist-call-context,s,1) -same => n(doneblacklist),NoOp() -same => n,Goto(callback-check-call-context,s,1) -same => n(donecallback),NoOp() -same => n,Goto(disa-check-call-context,s,1) -same => n(donedisacheck),Dial(|LOCALUSERS|,${RINGTIME},D(:w11111111)) -same => n,Goto(context-voicemail,s,1) - diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk_header.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk_header.conf.TEMPLATE deleted file mode 100644 index f6e44a5bf0..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk_header.conf.TEMPLATE +++ /dev/null @@ -1 +0,0 @@ -[context-incoming-gtalk] diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE deleted file mode 100644 index b2c3716bf4..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE +++ /dev/null @@ -1,12 +0,0 @@ - -[context-incoming-|NAME|] -exten => s,1,NoOp(${CALLERID}) -exten => s,n,Set(SOURCECONTEXT=context-incoming-|NAME|) -exten => s,n,Set(SOURCEEXTEN=s) -exten => s,n,Goto(blacklist-call-context,s,1) -exten => s,n(doneblacklist),NoOp() -exten => s,n,Goto(callback-check-call-context,s,1) -exten => s,n(donecallback),NoOp() -exten => s,n,Goto(disa-check-call-context,s,1) -exten => s,n(donedisacheck),Dial(|LOCALUSERS|,${RINGTIME},r) -exten => s,n,Goto(context-voicemail,s,1) diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_dial_local_user.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_dial_local_user.conf.TEMPLATE deleted file mode 100644 index 45e8758846..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_dial_local_user.conf.TEMPLATE +++ /dev/null @@ -1 +0,0 @@ -exten => |DEFAULTUSER|,1,Dial(SIP/|DEFAULTUSER|,${RINGTIME},r) diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_gtalk.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_gtalk.conf.TEMPLATE deleted file mode 100644 index 259c2ceaa1..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_gtalk.conf.TEMPLATE +++ /dev/null @@ -1,9 +0,0 @@ -exten => _|NUMPREFIX|,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN}@voice.google.com,60) -exten => _+|NUMPREFIX|,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:1}@voice.google.com,60) -exten => _|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN}@voice.google.com,60) -exten => _+|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:1}@voice.google.com,60) -exten => _00|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:2}@voice.google.com,60) -exten => _011|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:3}@voice.google.com,60) -exten => _010|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:3}@voice.google.com,60) -exten => _0011|NUMPREFIX|.,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN:4}@voice.google.com,60) - diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_gtalk.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_gtalk.conf.TEMPLATE deleted file mode 100644 index 1fa7713e23..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_gtalk.conf.TEMPLATE +++ /dev/null @@ -1,2 +0,0 @@ -exten => |PATTERN|,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN|SYMBOLSTOREMOVE|}@voice.google.com,60) - diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_sip.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_sip.conf.TEMPLATE deleted file mode 100644 index 178b6deaa6..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_sip.conf.TEMPLATE +++ /dev/null @@ -1 +0,0 @@ -exten => |PATTERN|,1,Dial(SIP/${EXTEN|SYMBOLSTOREMOVE|}@peer-|NAME|,60,r) diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_sip.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_sip.conf.TEMPLATE deleted file mode 100644 index 9b1d9addc9..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_sip.conf.TEMPLATE +++ /dev/null @@ -1,8 +0,0 @@ -exten => _|NUMPREFIX|,1,Dial(SIP/${EXTEN}@peer-|NAME|,60,r) -exten => _+|NUMPREFIX|,1,Dial(SIP/${EXTEN}@peer-|NAME|,60,r) -exten => _|NUMPREFIX|.,1,Dial(SIP/${EXTEN}@peer-|NAME|,60,r) -exten => _+|NUMPREFIX|.,1,Dial(SIP/${EXTEN:1}@peer-|NAME|,60,r) -exten => _00|NUMPREFIX|.,1,Dial(SIP/${EXTEN:2}@peer-|NAME|,60,r) -exten => _011|NUMPREFIX|.,1,Dial(SIP/${EXTEN:3}@peer-|NAME|,60,r) -exten => _010|NUMPREFIX|.,1,Dial(SIP/${EXTEN:3}@peer-|NAME|,60,r) -exten => _0011|NUMPREFIX|.,1,Dial(SIP/${EXTEN:4}@peer-|NAME|,60,r) diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_user_context_footer.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_user_context_footer.conf.TEMPLATE deleted file mode 100644 index a2ba28c055..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_user_context_footer.conf.TEMPLATE +++ /dev/null @@ -1,2 +0,0 @@ -include => context-voicemail-record-greeting -include => context-catch-all diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_user_context_header.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_user_context_header.conf.TEMPLATE deleted file mode 100644 index 5931eaf28b..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_user_context_header.conf.TEMPLATE +++ /dev/null @@ -1,3 +0,0 @@ - -[context-user-|DEFAULTUSER|] - diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_voicemail_disabled.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_voicemail_disabled.conf.TEMPLATE deleted file mode 100644 index be23c294df..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_voicemail_disabled.conf.TEMPLATE +++ /dev/null @@ -1,4 +0,0 @@ -[context-voicemail-record-greeting] - -[context-voicemail] -exten => s,1,Hangup() diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_voicemail_enabled.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_voicemail_enabled.conf.TEMPLATE deleted file mode 100644 index 4edd9cb426..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_voicemail_enabled.conf.TEMPLATE +++ /dev/null @@ -1,27 +0,0 @@ -[context-voicemail-record-greeting] -exten => *789,1,Wait(1) -exten => *789,n,Playback(/etc/pbx-voicemail/recordings/greeting) -exten => *789,n,Wait(1) -exten => *789,n,Playback(beep) -exten => *789,n,Playback(beep) -exten => *789,n,WaitExten(30) - -exten => t,1,Playback(vm-goodbye) -exten => t,n,Wait(2) -exten => t,n,Hangup() - -exten => *,1,Playback(beep) -exten => *,n,Playback(beep) -exten => *,n,Record(/tmp/voicemail/greeting:gsm,20,120,k) -exten => *,n,Wait(1) -exten => *,n,Playback(/tmp/voicemail/greeting) - -exten => h,1,System(/etc/pbx-voicemail/pbx-move-greeting &) - -[context-voicemail] -exten => s,1,Wait(2) -exten => s,2,Playback(/etc/pbx-voicemail/recordings/greeting) -exten => s,3,Wait(2) -exten => s,n,Record(/tmp/voicemail/voicemail%d:WAV,20,180,k) - -exten => h,1,System(/etc/pbx-voicemail/pbx-send-voicemail '${RECORDED_FILE}.WAV' '${CALLERID(all)}' &) diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/gtalk.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/gtalk.conf.TEMPLATE deleted file mode 100644 index 4f07a71660..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/gtalk.conf.TEMPLATE +++ /dev/null @@ -1,10 +0,0 @@ -[general] -context=context-incoming-gtalk -allowguest=yes -allowguests=yes -bindaddr=0.0.0.0 - -[guest] -disallow=all -allow=ulaw -context=context-incoming-gtalk diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/indications.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/indications.conf.TEMPLATE deleted file mode 100644 index d7088db7c4..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/indications.conf.TEMPLATE +++ /dev/null @@ -1,733 +0,0 @@ -; indications.conf -; Configuration file for location specific tone indications -; used by the pbx_indications module. -; -; NOTE: -; When adding countries to this file, please keep them in alphabetical -; order according to the 2-character country codes! -; -; The [general] category is for certain global variables. -; All other categories are interpreted as location specific indications -; -; -[general] -country=us ; default location - - -; [example] -; description = string -; The full name of your country, in English. -; alias = iso[,iso]* -; List of other countries 2-letter iso codes, which have the same -; tone indications. -; ringcadence = num[,num]* -; List of durations the physical bell rings. -; dial = tonelist -; Set of tones to be played when one picks up the hook. -; busy = tonelist -; Set of tones played when the receiving end is busy. -; congestion = tonelist -; Set of tones played when there is some congestion (on the network?) -; callwaiting = tonelist -; Set of tones played when there is a call waiting in the background. -; dialrecall = tonelist -; Not well defined; many phone systems play a recall dial tone after hook -; flash. -; record = tonelist -; Set of tones played when call recording is in progress. -; info = tonelist -; Set of tones played with special information messages (e.g., "number is -; out of service") -; 'name' = tonelist -; Every other variable will be available as a shortcut for the "PlayList" command -; but will not be used automatically by Asterisk. -; -; -; The tonelist itself is defined by a comma-separated sequence of elements. -; Each element consist of a frequency (f) with an optional duration (in ms) -; attached to it (f/duration). The frequency component may be a mixture of two -; frequencies (f1+f2) or a frequency modulated by another frequency (f1*f2). -; The implicit modulation depth is fixed at 90%, though. -; If the list element starts with a !, that element is NOT repeated, -; therefore, only if all elements start with !, the tonelist is time-limited, -; all others will repeat indefinitely. -; -; concisely: -; element = [!]freq[+|*freq2][/duration] -; tonelist = element[,element]* -; -; Please note that SPACES ARE NOT ALLOWED in tone lists! -; - -[at] -description = Austria -ringcadence = 1000,5000 -; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf -dial = 420 -busy = 420/400,0/400 -ring = 420/1000,0/5000 -congestion = 420/200,0/200 -callwaiting = 420/40,0/1960 -dialrecall = 420 -; RECORDTONE - not specified -record = 1400/80,0/14920 -info = 950/330,1450/330,1850/330,0/1000 -stutter = 380+420 - -[au] -description = Australia -; Reference http://www.acif.org.au/__data/page/3303/S002_2001.pdf -; Normal Ring -ringcadence = 400,200,400,2000 -; Distinctive Ring 1 - Forwarded Calls -; 400,400,200,200,400,1400 -; Distinctive Ring 2 - Selective Ring 2 + Operator + Recall -; 400,400,200,2000 -; Distinctive Ring 3 - Multiple Subscriber Number 1 -; 200,200,400,2200 -; Distinctive Ring 4 - Selective Ring 1 + Centrex -; 400,2600 -; Distinctive Ring 5 - Selective Ring 3 -; 400,400,200,400,200,1400 -; Distinctive Ring 6 - Multiple Subscriber Number 2 -; 200,400,200,200,400,1600 -; Distinctive Ring 7 - Multiple Subscriber Number 3 + Data Privacy -; 200,400,200,400,200,1600 -; Tones -dial = 413+438 -busy = 425/375,0/375 -ring = 413+438/400,0/200,413+438/400,0/2000 -; XXX Congestion: Should reduce by 10 db every other cadence XXX -congestion = 425/375,0/375,420/375,0/375 -callwaiting = 425/200,0/200,425/200,0/4400 -dialrecall = 413+438 -; Record tone used for Call Intrusion/Recording or Conference -record = !425/1000,!0/15000,425/360,0/15000 -info = 425/2500,0/500 -; Other Australian Tones -; The STD "pips" indicate the call is not an untimed local call -std = !525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100 -; Facility confirmation tone (eg. Call Forward Activated) -facility = 425 -; Message Waiting "stutter" dialtone -stutter = 413+438/100,0/40 -; Ringtone for calls to Telstra mobiles -ringmobile = 400+450/400,0/200,400+450/400,0/2000 - -[bg] -; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf -description = Bulgaria -ringcadence = 1000,4000 -; -dial = 425 -busy = 425/500,0/500 -ring = 425/1000,0/4000 -congestion = 425/250,0/250 -callwaiting = 425/150,0/150,425/150,0/4000 -dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 -record = 1400/425,0/15000 -info = 950/330,1400/330,1800/330,0/1000 -stutter = 425/1500,0/100 - -[br] -description = Brazil -ringcadence = 1000,4000 -dial = 425 -busy = 425/250,0/250 -ring = 425/1000,0/4000 -congestion = 425/250,0/250,425/750,0/250 -callwaiting = 425/50,0/1000 -; Dialrecall not used in Brazil standard (using UK standard) -dialrecall = 350+440 -; Record tone is not used in Brazil, use busy tone -record = 425/250,0/250 -; Info not used in Brazil standard (using UK standard) -info = 950/330,1400/330,1800/330 -stutter = 350+440 - -[be] -description = Belgium -; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf -ringcadence = 1000,3000 -dial = 425 -busy = 425/500,0/500 -ring = 425/1000,0/3000 -congestion = 425/167,0/167 -callwaiting = 1400/175,0/175,1400/175,0/3500 -; DIALRECALL - not specified -dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440" -; RECORDTONE - not specified -record = 1400/500,0/15000 -info = 900/330,1400/330,1800/330,0/1000 -stutter = 425/1000,0/250 - -[ch] -description = Switzerland -; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf -ringcadence = 1000,4000 -dial = 425 -busy = 425/500,0/500 -ring = 425/1000,0/4000 -congestion = 425/200,0/200 -callwaiting = 425/200,0/200,425/200,0/4000 -; DIALRECALL - not specified -dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 -; RECORDTONE - not specified -record = 1400/80,0/15000 -info = 950/330,1400/330,1800/330,0/1000 -stutter = 425+340/1100,0/1100 - -[cl] -description = Chile -; According to specs from Telefonica CTC Chile -ringcadence = 1000,3000 -dial = 400 -busy = 400/500,0/500 -ring = 400/1000,0/3000 -congestion = 400/200,0/200 -callwaiting = 400/250,0/8750 -dialrecall = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400 -record = 1400/500,0/15000 -info = 950/333,1400/333,1800/333,0/1000 -stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400 - -[cn] -description = China -; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf -ringcadence = 1000,4000 -dial = 450 -busy = 450/350,0/350 -ring = 450/1000,0/4000 -congestion = 450/700,0/700 -callwaiting = 450/400,0/4000 -dialrecall = 450 -record = 950/400,0/10000 -info = 450/100,0/100,450/100,0/100,450/100,0/100,450/400,0/400 -; STUTTER - not specified -stutter = 450+425 - -[cz] -description = Czech Republic -; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf -ringcadence = 1000,4000 -dial = 425/330,0/330,425/660,0/660 -busy = 425/330,0/330 -ring = 425/1000,0/4000 -congestion = 425/165,0/165 -callwaiting = 425/330,0/9000 -; DIALRECALL - not specified -dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425/330,0/330,425/660,0/660 -; RECORDTONE - not specified -record = 1400/500,0/14000 -info = 950/330,0/30,1400/330,0/30,1800/330,0/1000 -; STUTTER - not specified -stutter = 425/450,0/50 - -[de] -description = Germany -; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf -ringcadence = 1000,4000 -dial = 425 -busy = 425/480,0/480 -ring = 425/1000,0/4000 -congestion = 425/240,0/240 -callwaiting = !425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,0 -; DIALRECALL - not specified -dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 -; RECORDTONE - not specified -record = 1400/80,0/15000 -info = 950/330,1400/330,1800/330,0/1000 -stutter = 425+400 - -[dk] -description = Denmark -; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf -ringcadence = 1000,4000 -dial = 425 -busy = 425/500,0/500 -ring = 425/1000,0/4000 -congestion = 425/200,0/200 -callwaiting = !425/200,!0/600,!425/200,!0/3000,!425/200,!0/200,!425/200,0 -; DIALRECALL - not specified -dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 -; RECORDTONE - not specified -record = 1400/80,0/15000 -info = 950/330,1400/330,1800/330,0/1000 -; STUTTER - not specified -stutter = 425/450,0/50 - -[ee] -description = Estonia -; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf -ringcadence = 1000,4000 -dial = 425 -busy = 425/300,0/300 -ring = 425/1000,0/4000 -congestion = 425/200,0/200 -; CALLWAIT not in accordance to ITU -callwaiting = 950/650,0/325,950/325,0/30,1400/1300,0/2600 -; DIALRECALL - not specified -dialrecall = 425/650,0/25 -; RECORDTONE - not specified -record = 1400/500,0/15000 -; INFO not in accordance to ITU -info = 950/650,0/325,950/325,0/30,1400/1300,0/2600 -; STUTTER not specified -stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 - -[es] -description = Spain -ringcadence = 1500,3000 -dial = 425 -busy = 425/200,0/200 -ring = 425/1500,0/3000 -congestion = 425/200,0/200,425/200,0/200,425/200,0/600 -callwaiting = 425/175,0/175,425/175,0/3500 -dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425 -record = 1400/500,0/15000 -info = 950/330,0/1000 -dialout = 500 - - -[fi] -description = Finland -ringcadence = 1000,4000 -dial = 425 -busy = 425/300,0/300 -ring = 425/1000,0/4000 -congestion = 425/200,0/200 -callwaiting = 425/150,0/150,425/150,0/8000 -dialrecall = 425/650,0/25 -record = 1400/500,0/15000 -info = 950/650,0/325,950/325,0/30,1400/1300,0/2600 -stutter = 425/650,0/25 - -[fr] -description = France -; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf -ringcadence = 1500,3500 -; Dialtone can also be 440+330 -dial = 440 -busy = 440/500,0/500 -ring = 440/1500,0/3500 -; CONGESTION - not specified -congestion = 440/250,0/250 -callwait = 440/300,0/10000 -; DIALRECALL - not specified -dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 -; RECORDTONE - not specified -record = 1400/500,0/15000 -info = !950/330,!1400/330,!1800/330 -stutter = !440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,440 - -[gr] -description = Greece -ringcadence = 1000,4000 -dial = 425/200,0/300,425/700,0/800 -busy = 425/300,0/300 -ring = 425/1000,0/4000 -congestion = 425/200,0/200 -callwaiting = 425/150,0/150,425/150,0/8000 -dialrecall = 425/650,0/25 -record = 1400/400,0/15000 -info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0 -stutter = 425/650,0/25 - -[hu] -description = Hungary -; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf -ringcadence = 1250,3750 -dial = 425 -busy = 425/300,0/300 -ring = 425/1250,0/3750 -congestion = 425/300,0/300 -callwaiting = 425/40,0/1960 -dialrecall = 425+450 -; RECORDTONE - not specified -record = 1400/400,0/15000 -info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0 -stutter = 350+375+400 - -[il] -description = Israel -ringcadence = 1000,3000 -dial = 414 -busy = 414/500,0/500 -ring = 414/1000,0/3000 -congestion = 414/250,0/250 -callwaiting = 414/100,0/100,414/100,0/100,414/600,0/3000 -dialrecall = !414/100,!0/100,!414/100,!0/100,!414/100,!0/100,414 -record = 1400/500,0/15000 -info = 1000/330,1400/330,1800/330,0/1000 -stutter = !414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,414 - - -[in] -description = India -ringcadence = 400,200,400,2000 -dial = 400*25 -busy = 400/750,0/750 -ring = 400*25/400,0/200,400*25/400,0/2000 -congestion = 400/250,0/250 -callwaiting = 400/200,0/100,400/200,0/7500 -dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 -record = 1400/500,0/15000 -info = !950/330,!1400/330,!1800/330,0/1000 -stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 - -[it] -description = Italy -; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf -ringcadence = 1000,4000 -dial = 425/200,0/200,425/600,0/1000 -busy = 425/500,0/500 -ring = 425/1000,0/4000 -congestion = 425/200,0/200 -callwaiting = 425/400,0/100,425/250,0/100,425/150,0/14000 -dialrecall = 470/400,425/400 -record = 1400/400,0/15000 -info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0 -stutter = 470/400,425/400 - -[lt] -description = Lithuania -ringcadence = 1000,4000 -dial = 425 -busy = 425/350,0/350 -ring = 425/1000,0/4000 -congestion = 425/200,0/200 -callwaiting = 425/150,0/150,425/150,0/4000 -; DIALRECALL - not specified -dialrecall = 425/500,0/50 -; RECORDTONE - not specified -record = 1400/500,0/15000 -info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0 -; STUTTER - not specified -stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 - -[jp] -description = Japan -ringcadence = 1000,2000 -dial = 400 -busy = 400/500,0/500 -ring = 400+15/1000,0/2000 -congestion = 400/500,0/500 -callwaiting = 400+16/500,0/8000 -dialrecall = !400/200,!0/200,!400/200,!0/200,!400/200,!0/200,400 -record = 1400/500,0/15000 -info = !950/330,!1400/330,!1800/330,0 -stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400 - -[mx] -description = Mexico -ringcadence = 2000,4000 -dial = 425 -busy = 425/250,0/250 -ring = 425/1000,0/4000 -congestion = 425/250,0/250 -callwaiting = 425/200,0/600,425/200,0/10000 -dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 -record = 1400/500,0/15000 -info = 950/330,0/30,1400/330,0/30,1800/330,0/1000 -stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 - -[my] -description = Malaysia -ringcadence = 2000,4000 -dial = 425 -busy = 425/500,0/500 -ring = 425/400,0/200 -congestion = 425/500,0/500 - -[nl] -description = Netherlands -; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf -ringcadence = 1000,4000 -; Most of these 425's can also be 450's -dial = 425 -busy = 425/500,0/500 -ring = 425/1000,0/4000 -congestion = 425/250,0/250 -callwaiting = 425/500,0/9500 -; DIALRECALL - not specified -dialrecall = 425/500,0/50 -; RECORDTONE - not specified -record = 1400/500,0/15000 -info = 950/330,1400/330,1800/330,0/1000 -stutter = 425/500,0/50 - -[no] -description = Norway -ringcadence = 1000,4000 -dial = 425 -busy = 425/500,0/500 -ring = 425/1000,0/4000 -congestion = 425/200,0/200 -callwaiting = 425/200,0/600,425/200,0/10000 -dialrecall = 470/400,425/400 -record = 1400/400,0/15000 -info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0 -stutter = 470/400,425/400 - -[nz] -description = New Zealand -;NOTE - the ITU has different tonesets for NZ, but according to some residents there, -; this is, indeed, the correct way to do it. -ringcadence = 400,200,400,2000 -dial = 400 -busy = 400/250,0/250 -ring = 400+450/400,0/200,400+450/400,0/2000 -congestion = 400/375,0/375 -callwaiting = !400/200,!0/3000,!400/200,!0/3000,!400/200,!0/3000,!400/200 -dialrecall = !400/100!0/100,!400/100,!0/100,!400/100,!0/100,400 -record = 1400/425,0/15000 -info = 400/750,0/100,400/750,0/100,400/750,0/100,400/750,0/400 -stutter = !400/100!0/100,!400/100,!0/100,!400/100,!0/100,!400/100!0/100,!400/100,!0/100,!400/100,!0/100,400 -unobtainable = 400/75,0/100,400/75,0/100,400/75,0/100,400/75,0/400 - -[ph] - -; reference http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf - -description = Philippines -ringcadence = 1000,4000 -dial = 425 -busy = 480+620/500,0/500 -ring = 425+480/1000,0/4000 -congestion = 480+620/250,0/250 -callwaiting = 440/300,0/10000 -; DIALRECALL - not specified -dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 -; RECORDTONE - not specified -record = 1400/500,0/15000 -; INFO - not specified -info = !950/330,!1400/330,!1800/330,0 -; STUTTER - not specified -stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 - - -[pl] -description = Poland -ringcadence = 1000,4000 -dial = 425 -busy = 425/500,0/500 -ring = 425/1000,0/4000 -congestion = 425/500,0/500 -callwaiting = 425/150,0/150,425/150,0/4000 -; DIALRECALL - not specified -dialrecall = 425/500,0/50 -; RECORDTONE - not specified -record = 1400/500,0/15000 -; 950/1400/1800 3x0.33 on 1.0 off repeated 3 times -info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000 -; STUTTER - not specified -stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 - -[pt] -description = Portugal -ringcadence = 1000,5000 -dial = 425 -busy = 425/500,0/500 -ring = 425/1000,0/5000 -congestion = 425/200,0/200 -callwaiting = 440/300,0/10000 -dialrecall = 425/1000,0/200 -record = 1400/500,0/15000 -info = 950/330,1400/330,1800/330,0/1000 -stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 - -[ru] -; References: -; http://www.minsvyaz.ru/site.shtml?id=1806 -; http://www.aboutphone.info/lib/gost/45-223-2001.html -description = Russian Federation / ex Soviet Union -ringcadence = 1000,4000 -dial = 425 -busy = 425/350,0/350 -ring = 425/1000,0/4000 -congestion = 425/175,0/175 -callwaiting = 425/200,0/5000 -record = 1400/400,0/15000 -info = 950/330,1400/330,1800/330,0/1000 -dialrecall = 425/400,0/40 -stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 - -[se] -description = Sweden -ringcadence = 1000,5000 -dial = 425 -busy = 425/250,0/250 -ring = 425/1000,0/5000 -congestion = 425/250,0/750 -callwaiting = 425/200,0/500,425/200,0/9100 -dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 -record = 1400/500,0/15000 -info = !950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,0 -stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 -; stutter = 425/320,0/20 ; Real swedish standard, not used for now - -[sg] -description = Singapore -; Singapore -; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf -; Frequency specs are: 425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz -ringcadence = 400,200,400,2000 -dial = 425 -ring = 425*24/400,0/200,425*24/400,0/2000 ; modulation should be 100%, not 90% -busy = 425/750,0/750 -congestion = 425/250,0/250 -callwaiting = 425*24/300,0/200,425*24/300,0/3200 -stutter = !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425 -info = 950/330,1400/330,1800/330,0/1000 ; not currently in use acc. to reference -dialrecall = 425*24/500,0/500,425/500,0/2500 ; unspecified in IDA reference, use repeating Holding Tone A,B -record = 1400/500,0/15000 ; unspecified in IDA reference, use 0.5s tone every 15s -; additionally defined in reference -nutone = 425/2500,0/500 -intrusion = 425/250,0/2000 -warning = 425/624,0/4376 ; end of period tone, warning -acceptance = 425/125,0/125 -holdinga = !425*24/500,!0/500 ; followed by holdingb -holdingb = !425/500,!0/2500 - -[th] -description = Thailand -ringcadence = 1000,4000 -; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf -dial = 400*50 -busy = 400/500,0/500 -ring = 420/1000,0/5000 -congestion = 400/300,0/300 -callwaiting = 1000/400,10000/400,1000/400 -; DIALRECALL - not specified - use special dial tone instead. -dialrecall = 400*50/400,0/100,400*50/400,0/100 -; RECORDTONE - not specified -record = 1400/500,0/15000 -; INFO - specified as an announcement - use special information tones instead -info = 950/330,1400/330,1800/330 -; STUTTER - not specified -stutter = !400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,400 - -[uk] -description = United Kingdom -ringcadence = 400,200,400,2000 -; These are the official tones taken from BT SIN350. The actual tones -; used by BT include some volume differences so sound slightly different -; from Asterisk-generated ones. -dial = 350+440 -; Special dial is the intermittent dial tone heard when, for example, -; you have a divert active on the line -specialdial = 350+440/750,440/750 -; Busy is also called "Engaged" -busy = 400/375,0/375 -; "Congestion" is the Beep-bip engaged tone -congestion = 400/400,0/350,400/225,0/525 -; "Special Congestion" is not used by BT very often if at all -specialcongestion = 400/200,1004/300 -unobtainable = 400 -ring = 400+450/400,0/200,400+450/400,0/2000 -callwaiting = 400/100,0/4000 -; BT seem to use "Special Call Waiting" rather than just "Call Waiting" tones -specialcallwaiting = 400/250,0/250,400/250,0/250,400/250,0/5000 -; "Pips" used by BT on payphones. (Sounds wrong, but this is what BT claim it -; is and I've not used a payphone for years) -creditexpired = 400/125,0/125 -; These two are used to confirm/reject service requests on exchanges that -; don't do voice announcements. -confirm = 1400 -switching = 400/200,0/400,400/2000,0/400 -; This is the three rising tones Doo-dah-dee "Special Information Tone", -; usually followed by the BT woman saying an appropriate message. -info = 950/330,0/15,1400/330,0/15,1800/330,0/1000 -; Not listed in SIN350 -record = 1400/500,0/60000 -stutter = 350+440/750,440/750 - -[us] -description = United States / North America -ringcadence = 2000,4000 -dial = 350+440 -busy = 480+620/500,0/500 -ring = 440+480/2000,0/4000 -congestion = 480+620/250,0/250 -callwaiting = 440/300,0/10000 -dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 -record = 1400/500,0/15000 -info = !950/330,!1400/330,!1800/330,0 -stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 - -[us-old] -description = United States Circa 1950/ North America -ringcadence = 2000,4000 -dial = 600*120 -busy = 500*100/500,0/500 -ring = 420*40/2000,0/4000 -congestion = 500*100/250,0/250 -callwaiting = 440/300,0/10000 -dialrecall = !600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,600*120 -record = 1400/500,0/15000 -info = !950/330,!1400/330,!1800/330,0 -stutter = !600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,600*120 - -[tw] -description = Taiwan -; http://nemesis.lonestar.org/reference/telecom/signaling/dialtone.html -; http://nemesis.lonestar.org/reference/telecom/signaling/busy.html -; http://www.iproducts.com.tw/ee/kylink/06ky-1000a.htm -; http://www.pbx-manufacturer.com/ky120dx.htm -; http://www.nettwerked.net/tones.txt -; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/taiw_sup/taiw2.htm -; -; busy tone 480+620Hz 0.5 sec. on ,0.5 sec. off -; reorder tone 480+620Hz 0.25 sec. on,0.25 sec. off -; ringing tone 440+480Hz 1 sec. on ,2 sec. off -; -ringcadence = 1000,4000 -dial = 350+440 -busy = 480+620/500,0/500 -ring = 440+480/1000,0/2000 -congestion = 480+620/250,0/250 -callwaiting = 350+440/250,0/250,350+440/250,0/3250 -dialrecall = 300/1500,0/500 -record = 1400/500,0/15000 -info = !950/330,!1400/330,!1800/330,0 -stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 - -[ve] -; Tone definition source for ve found on -; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf -description = Venezuela / South America -ringcadence = 1000,4000 -dial = 425 -busy = 425/500,0/500 -ring = 425/1000,0/4000 -congestion = 425/250,0/250 -callwaiting = 400+450/300,0/6000 -dialrecall = 425 -record = 1400/500,0/15000 -info = !950/330,!1440/330,!1800/330,0/1000 - - -[za] -description = South Africa -; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/safr_sup/saf02.htm -; (definitions for other countries can also be found there) -; Note, though, that South Africa uses two switch types in their network -- -; Alcatel switches -- mainly in the Western Cape, and Siemens elsewhere. -; The former use 383+417 in dial, ringback etc. The latter use 400*33 -; I've provided both, uncomment the ones you prefer -ringcadence = 400,200,400,2000 -; dial/ring/callwaiting for the Siemens switches: -dial = 400*33 -ring = 400*33/400,0/200,400*33/400,0/2000 -callwaiting = 400*33/250,0/250,400*33/250,0/250,400*33/250,0/250,400*33/250,0/250 -; dial/ring/callwaiting for the Alcatel switches: -; dial = 383+417 -; ring = 383+417/400,0/200,383+417/400,0/2000 -; callwaiting = 383+417/250,0/250,383+417/250,0/250,383+417/250,0/250,383+417/250,0/250 -congestion = 400/250,0/250 -busy = 400/500,0/500 -dialrecall = 350+440 -; XXX Not sure about the RECORDTONE -record = 1400/500,0/10000 -info = 950/330,1400/330,1800/330,0/330 -stutter = !400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,400*33 diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/jabber.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/jabber.conf.TEMPLATE deleted file mode 100644 index cf71e1f8f4..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/jabber.conf.TEMPLATE +++ /dev/null @@ -1,4 +0,0 @@ -[general] -autoregister=yes - -#include jabber_users.conf diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/jabber_users.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/jabber_users.conf.TEMPLATE deleted file mode 100644 index 3ee2463ed2..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/jabber_users.conf.TEMPLATE +++ /dev/null @@ -1,8 +0,0 @@ -[gtalk-|NAME|] -type=client -serverhost=talk.google.com -username=|USERNAME|/Talk -secret=|SECRET| -timeout=150 -status=|STATUS| -statusmessage=|STATUSMESSAGE| diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/logger.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/logger.conf.TEMPLATE deleted file mode 100644 index e57325013a..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/logger.conf.TEMPLATE +++ /dev/null @@ -1,7 +0,0 @@ -[general] -queue_log = no -event_log = no - -[logfiles] -console => notice,warning,error -messages => error diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/manager.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/manager.conf.TEMPLATE deleted file mode 100644 index 2ac2f0033f..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/manager.conf.TEMPLATE +++ /dev/null @@ -1,7 +0,0 @@ -[general] -enabled = no - -port = 5038 -bindaddr = 0.0.0.0 - - diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/modules.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/modules.conf.TEMPLATE deleted file mode 100644 index 93c74336d1..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/modules.conf.TEMPLATE +++ /dev/null @@ -1,34 +0,0 @@ -[modules] -autoload=no -load => res_jabber.so ; Used for Gtalk -load => res_clioriginate.so ; originate calls from commandline -load => res_rtp_asterisk.so ; rtp "engine" is now a loadable module in asterisk 1.8 -load => pbx_config.so ; Text Extension Configuration Requires N/A -load => func_callerid.so ; Gets or sets Caller*ID data on the channel. - Requires ? -load => func_channel.so -load => func_logic.so ; Logic functions (if, etc.) -load => func_strings.so ; string manipulation functions -load => cdr_manager.so ; Asterisk Call Manager CDR Backend - Requires N/A -load => chan_local.so ; Show status of local channels- Requires N/A -load => chan_gtalk.so ; Use gtalk -load => chan_sip.so ; Session Initiation Protocol (SIP) - Requires res_features.so -load => codec_alaw.so ; A-law Coder/Decoder - Requires N/A -load => codec_a_mu.so ; A-law and Mulaw direct Coder/Decoder - Requires N/A -load => codec_gsm.so ; GSM/PCM16 (signed linear) Codec Translat - Requires N/A -load => codec_ulaw.so ; Mu-law Coder/Decoder - Requires N/A -load => format_gsm.so ; Raw GSM data - Requires N/A -load => format_pcm.so ; Raw uLaw 8khz Audio support (PCM) - Requires N/A -load => format_wav_gsm.so -load => app_dial.so ; Dialing Application - Requires res_features.so, res_musiconhold.so -load => app_parkandannounce.so ; Call Parking and Announce Application - Requires res_features.so -load => app_playback.so ; Sound File Playback Application - Requires N/A -load => app_record.so ; Sound File Record Application - Requires N/A -load => app_system.so ; Execute a system command - Requires N/A -load => app_disa.so ; Direct Inward System Access -load => app_authenticate.so ; Authenticate via pin -load => app_senddtmf.so ; Ability to send DTMF tones on the line. -load => func_cut.so ; To manipulate strings -load => func_timeout.so ; Used for DISA timeouts - -[global] -chan_modem.so=no diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/rtp.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/rtp.conf.TEMPLATE deleted file mode 100644 index 10d577d3a2..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/rtp.conf.TEMPLATE +++ /dev/null @@ -1,6 +0,0 @@ -[general] -rtpstart=|RTPSTART| -rtpend=|RTPEND| -rtpchecksums=no -dtmftimeout=3000 -rtcpinterval = 2000 diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sip.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/sip.conf.TEMPLATE deleted file mode 100644 index 8f3b112ff6..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sip.conf.TEMPLATE +++ /dev/null @@ -1,39 +0,0 @@ -[general] -transport=udp -context=default-incoming-call-context -allowoverlap=yes -allowtransfer=yes -realm=asterisk -bindaddr=0.0.0.0 -srvlookup=yes -maxexpiry=600 -minexpiry=60 -defaultexpiry=300 -qualifyfreq=55 -disallow=all -allow=ulaw -allow=alaw -dtmfmode = inband -alwaysauthreject = yes -t1min=100 -timert1=500 -timerb=16000 -rtptimeout=600 -rtpkeepalive=30 -useragent=|USERAGENT| -localnet=192.168.0.0/16 -localnet=10.0.0.0/8 -localnet=172.16.0.0/12 -nat=yes -directmedia=no -sipdebug=no -bindport=|BINDPORT| -externhost=|EXTERNHOST| -externrefresh=60 - -#include sip_registrations.conf - -[authentication] - -#include sip_peers.conf -#include sip_users.conf diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_peer.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_peer.TEMPLATE deleted file mode 100644 index 30abaadd58..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_peer.TEMPLATE +++ /dev/null @@ -1,13 +0,0 @@ - -[peer-|NAME|] -type = peer -defaultuser = |DEFAULTUSER| -fromuser = |FROMUSER| -secret = |SECRET| -host = |HOST| -fromdomain = |FROMDOMAIN| -context = context-incoming-|NAME| -insecure = port,invite -qualify = 2000 -|PORT| -|OUTBOUNDPROXY| diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_registration.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_registration.TEMPLATE deleted file mode 100644 index e139d43f03..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_registration.TEMPLATE +++ /dev/null @@ -1,2 +0,0 @@ -register => |DEFAULTUSER|:|SECRET|@peer-|NAME| - diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_user.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_user.TEMPLATE deleted file mode 100644 index 61a8b0b86b..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_user.TEMPLATE +++ /dev/null @@ -1,11 +0,0 @@ - -[|DEFAULTUSER|] -fullname = |FULLNAME| -defaultuser = |DEFAULTUSER| -secret = |SECRET| -hassip = yes -hasvoicemail = no -host = dynamic -type = friend -context = context-user-|CONTEXTNAME| -qualify = no
\ No newline at end of file diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-alreadyon.gsm b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-alreadyon.gsm Binary files differdeleted file mode 100644 index 83fe27ecfa..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-alreadyon.gsm +++ /dev/null diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-incorrect.gsm b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-incorrect.gsm Binary files differdeleted file mode 100644 index 27d934beb0..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-incorrect.gsm +++ /dev/null diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-pass.gsm b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-pass.gsm Binary files differdeleted file mode 100644 index f95637bb32..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-pass.gsm +++ /dev/null diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/auth-incorrect.gsm b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/auth-incorrect.gsm Binary files differdeleted file mode 100644 index 12fec25d56..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/auth-incorrect.gsm +++ /dev/null diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/auth-thankyou.gsm b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/auth-thankyou.gsm Binary files differdeleted file mode 100644 index 93f936d1a0..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/auth-thankyou.gsm +++ /dev/null diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/beep.gsm b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/beep.gsm Binary files differdeleted file mode 100644 index d38eda9cc5..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/beep.gsm +++ /dev/null diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/vm-goodbye.gsm b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/vm-goodbye.gsm Binary files differdeleted file mode 100644 index 735b281c8e..0000000000 --- a/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/vm-goodbye.gsm +++ /dev/null |