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diff --git a/applications/luci-app-pbx/COPYING b/applications/luci-app-pbx/COPYING new file mode 100644 index 000000000..94a9ed024 --- /dev/null +++ b/applications/luci-app-pbx/COPYING @@ -0,0 +1,674 @@ + 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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If not, see <http://www.gnu.org/licenses/>. + +Also add information on how to contact you by electronic and paper mail. + + If the program does terminal interaction, make it output a short +notice like this when it starts in an interactive mode: + + <program> Copyright (C) <year> <name of author> + This program comes with ABSOLUTELY NO WARRANTY; for details type `show w'. + This is free software, and you are welcome to redistribute it + under certain conditions; type `show c' for details. + +The hypothetical commands `show w' and `show c' should show the appropriate +parts of the General Public License. Of course, your program's commands +might be different; for a GUI interface, you would use an "about box". + + You should also get your employer (if you work as a programmer) or school, +if any, to sign a "copyright disclaimer" for the program, if necessary. +For more information on this, and how to apply and follow the GNU GPL, see +<http://www.gnu.org/licenses/>. + + The GNU General Public License does not permit incorporating your program +into proprietary programs. If your program is a subroutine library, you +may consider it more useful to permit linking proprietary applications with +the library. If this is what you want to do, use the GNU Lesser General +Public License instead of this License. But first, please read +<http://www.gnu.org/philosophy/why-not-lgpl.html>. diff --git a/applications/luci-app-pbx/CREDITS-SOUNDS b/applications/luci-app-pbx/CREDITS-SOUNDS new file mode 100644 index 000000000..1fa64bc6c --- /dev/null +++ b/applications/luci-app-pbx/CREDITS-SOUNDS @@ -0,0 +1,7 @@ +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 new file mode 100644 index 000000000..fe9c8221a --- /dev/null +++ b/applications/luci-app-pbx/LICENSE-SOUNDS @@ -0,0 +1,312 @@ +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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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 new file mode 100644 index 000000000..1379dcf89 --- /dev/null +++ b/applications/luci-app-pbx/Makefile @@ -0,0 +1,19 @@ +# +# 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:= \ + +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 new file mode 100644 index 000000000..b77814b15 --- /dev/null +++ b/applications/luci-app-pbx/luasrc/controller/pbx.lua @@ -0,0 +1,29 @@ +--[[ + 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 new file mode 100644 index 000000000..5d4f135c5 --- /dev/null +++ b/applications/luci-app-pbx/luasrc/model/cbi/pbx-advanced.lua @@ -0,0 +1,293 @@ +--[[ + 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" + +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 new file mode 100644 index 000000000..ca373d63a --- /dev/null +++ b/applications/luci-app-pbx/luasrc/model/cbi/pbx-calls.lua @@ -0,0 +1,424 @@ +--[[ + 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 new file mode 100644 index 000000000..3c36a168d --- /dev/null +++ b/applications/luci-app-pbx/luasrc/model/cbi/pbx-google.lua @@ -0,0 +1,122 @@ +--[[ + 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 new file mode 100644 index 000000000..c7c8b4d8b --- /dev/null +++ b/applications/luci-app-pbx/luasrc/model/cbi/pbx-users.lua @@ -0,0 +1,133 @@ +--[[ + 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 new file mode 100644 index 000000000..ed1ed1edb --- /dev/null +++ b/applications/luci-app-pbx/luasrc/model/cbi/pbx-voip.lua @@ -0,0 +1,116 @@ +--[[ + 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" + +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" + +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" + +return m diff --git a/applications/luci-app-pbx/luasrc/model/cbi/pbx.lua b/applications/luci-app-pbx/luasrc/model/cbi/pbx.lua new file mode 100644 index 000000000..4c5fcbdec --- /dev/null +++ b/applications/luci-app-pbx/luasrc/model/cbi/pbx.lua @@ -0,0 +1,115 @@ +--[[ + 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 new file mode 100644 index 000000000..c8a0a9967 --- /dev/null +++ b/applications/luci-app-pbx/po/ca/pbx.po @@ -0,0 +1,509 @@ +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 new file mode 100644 index 000000000..8b69ef15d --- /dev/null +++ b/applications/luci-app-pbx/po/cs/pbx.po @@ -0,0 +1,487 @@ +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 new file mode 100644 index 000000000..3bc4bd428 --- /dev/null +++ b/applications/luci-app-pbx/po/de/pbx.po @@ -0,0 +1,699 @@ +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 new file mode 100644 index 000000000..717e2563b --- /dev/null +++ b/applications/luci-app-pbx/po/el/pbx.po @@ -0,0 +1,493 @@ +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 new file mode 100644 index 000000000..8b995e1a3 --- /dev/null +++ b/applications/luci-app-pbx/po/en/pbx.po @@ -0,0 +1,502 @@ +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 new file mode 100644 index 000000000..8071b61f0 --- /dev/null +++ b/applications/luci-app-pbx/po/es/pbx.po @@ -0,0 +1,677 @@ +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 new file mode 100644 index 000000000..971a69648 --- /dev/null +++ b/applications/luci-app-pbx/po/fr/pbx.po @@ -0,0 +1,484 @@ +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 new file mode 100644 index 000000000..2a458214d --- /dev/null +++ b/applications/luci-app-pbx/po/he/pbx.po @@ -0,0 +1,484 @@ +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 new file mode 100644 index 000000000..2a458214d --- /dev/null +++ b/applications/luci-app-pbx/po/hu/pbx.po @@ -0,0 +1,484 @@ +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 new file mode 100644 index 000000000..6da8e45d9 --- /dev/null +++ b/applications/luci-app-pbx/po/it/pbx.po @@ -0,0 +1,487 @@ +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 new file mode 100644 index 000000000..76199f419 --- /dev/null +++ b/applications/luci-app-pbx/po/ja/pbx.po @@ -0,0 +1,493 @@ +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 new file mode 100644 index 000000000..23403f290 --- /dev/null +++ b/applications/luci-app-pbx/po/ms/pbx.po @@ -0,0 +1,483 @@ +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 new file mode 100644 index 000000000..2a458214d --- /dev/null +++ b/applications/luci-app-pbx/po/no/pbx.po @@ -0,0 +1,484 @@ +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 new file mode 100644 index 000000000..4e80a4581 --- /dev/null +++ b/applications/luci-app-pbx/po/pl/pbx.po @@ -0,0 +1,508 @@ +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 new file mode 100644 index 000000000..fd93e4fff --- /dev/null +++ b/applications/luci-app-pbx/po/pt-br/pbx.po @@ -0,0 +1,744 @@ +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 new file mode 100644 index 000000000..75b6c8cd1 --- /dev/null +++ b/applications/luci-app-pbx/po/pt/pbx.po @@ -0,0 +1,487 @@ +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 new file mode 100644 index 000000000..49e8daccf --- /dev/null +++ b/applications/luci-app-pbx/po/ro/pbx.po @@ -0,0 +1,488 @@ +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 new file mode 100644 index 000000000..e85c947e1 --- /dev/null +++ b/applications/luci-app-pbx/po/ru/pbx.po @@ -0,0 +1,525 @@ +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 new file mode 100644 index 000000000..7b6d4a5c6 --- /dev/null +++ b/applications/luci-app-pbx/po/sk/pbx.po @@ -0,0 +1,484 @@ +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 new file mode 100644 index 000000000..400289b6b --- /dev/null +++ b/applications/luci-app-pbx/po/sv/pbx.po @@ -0,0 +1,506 @@ +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 new file mode 100644 index 000000000..86dd2eb72 --- /dev/null +++ b/applications/luci-app-pbx/po/templates/pbx.pot @@ -0,0 +1,477 @@ +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 new file mode 100644 index 000000000..59af3e878 --- /dev/null +++ b/applications/luci-app-pbx/po/tr/pbx.po @@ -0,0 +1,484 @@ +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 new file mode 100644 index 000000000..d65a78443 --- /dev/null +++ b/applications/luci-app-pbx/po/uk/pbx.po @@ -0,0 +1,501 @@ +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 new file mode 100644 index 000000000..59af3e878 --- /dev/null +++ b/applications/luci-app-pbx/po/vi/pbx.po @@ -0,0 +1,484 @@ +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 new file mode 100644 index 000000000..8ac03e1aa --- /dev/null +++ b/applications/luci-app-pbx/po/zh-cn/pbx.po @@ -0,0 +1,495 @@ +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 "" +"输入你想自动屏蔽的电话号码。你应该忽略国家代码和任何前导零,但请测试来确保你成" +"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 new file mode 100644 index 000000000..aa05be778 --- /dev/null +++ b/applications/luci-app-pbx/po/zh-tw/pbx.po @@ -0,0 +1,507 @@ +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 "" +"打入你允許自動通話的號碼. 你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)你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 "你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.你將能夠免費呼叫其他本地端用戶(e.g 其它類比電話機(ATAs)和使用你的" +"VoIP供應商講電話就像你在使用本地的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" +"你的路由器/GW上執行,你將必須在你的路由器/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你的PIN碼將不會顯示. 除非你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你的密碼將不會顯示. 除非你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 new file mode 100644 index 000000000..ca7c1669d --- /dev/null +++ b/applications/luci-app-pbx/root/etc/config/pbx @@ -0,0 +1 @@ +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 new file mode 100644 index 000000000..39da6f880 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/config/pbx-advanced @@ -0,0 +1,5 @@ +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 new file mode 100644 index 000000000..822bd4a1b --- /dev/null +++ b/applications/luci-app-pbx/root/etc/config/pbx-calls @@ -0,0 +1,7 @@ +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 new file mode 100644 index 000000000..e69de29bb --- /dev/null +++ b/applications/luci-app-pbx/root/etc/config/pbx-google diff --git a/applications/luci-app-pbx/root/etc/config/pbx-users b/applications/luci-app-pbx/root/etc/config/pbx-users new file mode 100644 index 000000000..a4277b1bf --- /dev/null +++ b/applications/luci-app-pbx/root/etc/config/pbx-users @@ -0,0 +1 @@ +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 new file mode 100644 index 000000000..e69de29bb --- /dev/null +++ b/applications/luci-app-pbx/root/etc/config/pbx-voip diff --git a/applications/luci-app-pbx/root/etc/init.d/pbx-asterisk b/applications/luci-app-pbx/root/etc/init.d/pbx-asterisk new file mode 100755 index 000000000..e05ae11cd --- /dev/null +++ b/applications/luci-app-pbx/root/etc/init.d/pbx-asterisk @@ -0,0 +1,837 @@ +#!/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 ${MODULENAME}-advanced + pbx_assemble_and_copy_config + + # Link sound files + pbx_link_sounds + + # Enforce ownership of specified files and directories. + pbx_fix_ownership +} diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE new file mode 100644 index 000000000..ac5439615 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE @@ -0,0 +1,17 @@ +[directories] +astetcdir => /etc/asterisk +astmoddir => /usr/lib/asterisk/modules +astvarlibdir => /usr/lib/asterisk +astdbdir => /usr/lib/asterisk +astkeydir => /usr/lib/asterisk +astdatadir => /usr/lib/asterisk +astagidir => /usr/lib/asterisk/agi-bin +astspooldir => /var/spool/asterisk +astrundir => /var/run/asterisk +astlogdir => /var/log/asterisk + +[options] +languageprefix = yes +dumpcore = no +runuser = nobody +rungroup = nogroup diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/delayedcallback b/applications/luci-app-pbx/root/etc/pbx-asterisk/delayedcallback new file mode 100755 index 000000000..903efe9ad --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/delayedcallback @@ -0,0 +1,18 @@ +#!/bin/sh + +# Check if there are more than one instance of this command +# with the same command line running at the same time for some +# reason, then quit. We are checking for the same +# commandline in order to permit two different callback +# attempts simultaneously. + +if ! grep -q "$@" /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: $@" >> /dev/shm/delayedcallback.err +fi diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE new file mode 100644 index 000000000..c8966edd8 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE @@ -0,0 +1,25 @@ +[general] +static = yes +writeprotect = yes +clearglobalvars = no + +[globals] +RINGTIME => |RINGTIME| + +[default] + +[context-user-hangup-call-context] +exten => s,1,Hangup() +exten => _X.,1,Hangup() + +[context-catch-all] +exten => _[!-~].,1,Dial(SIP/${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 new file mode 100644 index 000000000..54ee989b0 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist.conf.TEMPLATE @@ -0,0 +1 @@ +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 new file mode 100644 index 000000000..da964f238 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist_footer.conf.TEMPLATE @@ -0,0 +1,2 @@ +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 new file mode 100644 index 000000000..de0e98465 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_blacklist_header.conf.TEMPLATE @@ -0,0 +1,3 @@ + +[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 new file mode 100644 index 000000000..06b1a4b6b --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check.conf.TEMPLATE @@ -0,0 +1 @@ +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 new file mode 100644 index 000000000..282fe9e8f --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check_footer.conf.TEMPLATE @@ -0,0 +1,2 @@ +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 new file mode 100644 index 000000000..be289c4d3 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback-check_header.conf.TEMPLATE @@ -0,0 +1,3 @@ + +[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 new file mode 100644 index 000000000..43eec788f --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_gtalk.conf.TEMPLATE @@ -0,0 +1,4 @@ +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 new file mode 100644 index 000000000..0b8fb4c23 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_header.conf.TEMPLATE @@ -0,0 +1 @@ +[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 new file mode 100644 index 000000000..300e9fa0e --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_callback_sip.conf.TEMPLATE @@ -0,0 +1,4 @@ +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 new file mode 100644 index 000000000..35836e290 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_default.conf.TEMPLATE @@ -0,0 +1,11 @@ +[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 new file mode 100644 index 000000000..1910ff4d9 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_default_user.conf.TEMPLATE @@ -0,0 +1 @@ +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 new file mode 100644 index 000000000..ba2379b73 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check.conf.TEMPLATE @@ -0,0 +1 @@ +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 new file mode 100644 index 000000000..74056fa01 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check_footer.conf.TEMPLATE @@ -0,0 +1 @@ +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 new file mode 100644 index 000000000..e0d67b802 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-check_header.conf.TEMPLATE @@ -0,0 +1,2 @@ +[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 new file mode 100644 index 000000000..74e48de8c --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa-nopin.conf.TEMPLATE @@ -0,0 +1,5 @@ +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 new file mode 100644 index 000000000..3dd8fa35c --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa.conf.TEMPLATE @@ -0,0 +1,6 @@ +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 new file mode 100644 index 000000000..a74227114 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_disa_header.conf.TEMPLATE @@ -0,0 +1 @@ +[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 new file mode 100644 index 000000000..3f9cf4c7d --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk.conf.TEMPLATE @@ -0,0 +1,15 @@ +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|,${RINGTIME},D(:w11111111)) +same => n,Goto(context-voicemail,s,1) + diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk_header.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk_header.conf.TEMPLATE new file mode 100644 index 000000000..f6e44a5bf --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk_header.conf.TEMPLATE @@ -0,0 +1 @@ +[context-incoming-gtalk] diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE new file mode 100644 index 000000000..b2c3716bf --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE @@ -0,0 +1,12 @@ + +[context-incoming-|NAME|] +exten => s,1,NoOp(${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|,${RINGTIME},r) +exten => s,n,Goto(context-voicemail,s,1) diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_dial_local_user.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_dial_local_user.conf.TEMPLATE new file mode 100644 index 000000000..45e875884 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_dial_local_user.conf.TEMPLATE @@ -0,0 +1 @@ +exten => |DEFAULTUSER|,1,Dial(SIP/|DEFAULTUSER|,${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 new file mode 100644 index 000000000..259c2ceaa --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_gtalk.conf.TEMPLATE @@ -0,0 +1,9 @@ +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 new file mode 100644 index 000000000..1fa7713e2 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_gtalk.conf.TEMPLATE @@ -0,0 +1,2 @@ +exten => |PATTERN|,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN|SYMBOLSTOREMOVE|}@voice.google.com,60) + diff --git a/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_sip.conf.TEMPLATE b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_sip.conf.TEMPLATE new file mode 100644 index 000000000..178b6deaa --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_sip.conf.TEMPLATE @@ -0,0 +1 @@ +exten => |PATTERN|,1,Dial(SIP/${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 new file mode 100644 index 000000000..9b1d9addc --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_outgoing_sip.conf.TEMPLATE @@ -0,0 +1,8 @@ +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 new file mode 100644 index 000000000..a2ba28c05 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_user_context_footer.conf.TEMPLATE @@ -0,0 +1,2 @@ +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 new file mode 100644 index 000000000..5931eaf28 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_user_context_header.conf.TEMPLATE @@ -0,0 +1,3 @@ + +[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 new file mode 100644 index 000000000..be23c294d --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_voicemail_disabled.conf.TEMPLATE @@ -0,0 +1,4 @@ +[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 new file mode 100644 index 000000000..4edd9cb42 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/extensions_voicemail_enabled.conf.TEMPLATE @@ -0,0 +1,27 @@ +[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 new file mode 100644 index 000000000..4f07a7166 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/gtalk.conf.TEMPLATE @@ -0,0 +1,10 @@ +[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 new file mode 100644 index 000000000..d7088db7c --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/indications.conf.TEMPLATE @@ -0,0 +1,733 @@ +; 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 new file mode 100644 index 000000000..cf71e1f8f --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/jabber.conf.TEMPLATE @@ -0,0 +1,4 @@ +[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 new file mode 100644 index 000000000..3ee2463ed --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/jabber_users.conf.TEMPLATE @@ -0,0 +1,8 @@ +[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 new file mode 100644 index 000000000..e57325013 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/logger.conf.TEMPLATE @@ -0,0 +1,7 @@ +[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 new file mode 100644 index 000000000..2ac2f0033 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/manager.conf.TEMPLATE @@ -0,0 +1,7 @@ +[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 new file mode 100644 index 000000000..93c74336d --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/modules.conf.TEMPLATE @@ -0,0 +1,34 @@ +[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 new file mode 100644 index 000000000..10d577d3a --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/rtp.conf.TEMPLATE @@ -0,0 +1,6 @@ +[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 new file mode 100644 index 000000000..8f3b112ff --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/sip.conf.TEMPLATE @@ -0,0 +1,39 @@ +[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 new file mode 100644 index 000000000..30abaadd5 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_peer.TEMPLATE @@ -0,0 +1,13 @@ + +[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 new file mode 100644 index 000000000..e139d43f0 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_registration.TEMPLATE @@ -0,0 +1,2 @@ +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 new file mode 100644 index 000000000..61a8b0b86 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/sip_user.TEMPLATE @@ -0,0 +1,11 @@ + +[|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 differnew file mode 100644 index 000000000..83fe27ecf --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-alreadyon.gsm 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 differnew file mode 100644 index 000000000..27d934beb --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-incorrect.gsm 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 differnew file mode 100644 index 000000000..f95637bb3 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/agent-pass.gsm 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 differnew file mode 100644 index 000000000..12fec25d5 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/auth-incorrect.gsm 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 differnew file mode 100644 index 000000000..93f936d1a --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/auth-thankyou.gsm 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 differnew file mode 100644 index 000000000..d38eda9cc --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/beep.gsm 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 differnew file mode 100644 index 000000000..735b281c8 --- /dev/null +++ b/applications/luci-app-pbx/root/etc/pbx-asterisk/sounds/vm-goodbye.gsm |