This is the initial release of luci-pbx as submitted to the mailing list.

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+                    GNU GENERAL PUBLIC LICENSE
+                       Version 3, 29 June 2007
+
+ Copyright (C) 2007 Free Software Foundation, Inc. <http://fsf.org/>
+ Everyone is permitted to copy and distribute verbatim copies
+ of this license document, but changing it is not allowed.
+
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+PURPOSE.  THE ENTIRE RISK AS TO THE QUALITY AND PERFORMANCE OF THE PROGRAM
+IS WITH YOU.  SHOULD THE PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF
+ALL NECESSARY SERVICING, REPAIR OR CORRECTION.
+
+  16. Limitation of Liability.
+
+  IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING
+WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MODIFIES AND/OR CONVEYS
+THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES, INCLUDING ANY
+GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING OUT OF THE
+USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED TO LOSS OF
+DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY YOU OR THIRD
+PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER PROGRAMS),
+EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF
+SUCH DAMAGES.
+
+  17. Interpretation of Sections 15 and 16.
+
+  If the disclaimer of warranty and limitation of liability provided
+above cannot be given local legal effect according to their terms,
+reviewing courts shall apply local law that most closely approximates
+an absolute waiver of all civil liability in connection with the
+Program, unless a warranty or assumption of liability accompanies a
+copy of the Program in return for a fee.
+
+                     END OF TERMS AND CONDITIONS
+
+            How to Apply These Terms to Your New Programs
+
+  If you develop a new program, and you want it to be of the greatest
+possible use to the public, the best way to achieve this is to make it
+free software which everyone can redistribute and change under these terms.
+
+  To do so, attach the following notices to the program.  It is safest
+to attach them to the start of each source file to most effectively
+state the exclusion of warranty; and each file should have at least
+the "copyright" line and a pointer to where the full notice is found.
+
+    <one line to give the program's name and a brief idea of what it does.>
+    Copyright (C) <year>  <name of author>
+
+    This program is free software: you can redistribute it and/or modify
+    it under the terms of the GNU General Public License as published by
+    the Free Software Foundation, either version 3 of the License, or
+    (at your option) any later version.
+
+    This program is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+    GNU General Public License for more details.
+
+    You should have received a copy of the GNU General Public License
+    along with this program.  If not, see <http://www.gnu.org/licenses/>.
+
+Also add information on how to contact you by electronic and paper mail.
+
+  If the program does terminal interaction, make it output a short
+notice like this when it starts in an interactive mode:
+
+    <program>  Copyright (C) <year>  <name of author>
+    This program comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
+    This is free software, and you are welcome to redistribute it
+    under certain conditions; type `show c' for details.
+
+The hypothetical commands `show w' and `show c' should show the appropriate
+parts of the General Public License.  Of course, your program's commands
+might be different; for a GUI interface, you would use an "about box".
+
+  You should also get your employer (if you work as a programmer) or school,
+if any, to sign a "copyright disclaimer" for the program, if necessary.
+For more information on this, and how to apply and follow the GNU GPL, see
+<http://www.gnu.org/licenses/>.
+
+  The GNU General Public License does not permit incorporating your program
+into proprietary programs.  If your program is a subroutine library, you
+may consider it more useful to permit linking proprietary applications with
+the library.  If this is what you want to do, use the GNU Lesser General
+Public License instead of this License.  But first, please read
+<http://www.gnu.org/philosophy/why-not-lgpl.html>.
diff --git a/applications/luci-pbx/Makefile b/applications/luci-pbx/Makefile
new file mode 100644
index 000000000..9a720f0d2
--- /dev/null
+++ b/applications/luci-pbx/Makefile
@@ -0,0 +1,4 @@
+PO = pbx
+
+include ../../build/config.mk
+include ../../build/module.mk
diff --git a/applications/luci-pbx/luasrc/controller/pbx.lua b/applications/luci-pbx/luasrc/controller/pbx.lua
new file mode 100644
index 000000000..127b40705
--- /dev/null
+++ b/applications/luci-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", "pbs-advanced"}, cbi("pbx-advanced"), "Advanced Settings", 5)
+end
diff --git a/applications/luci-pbx/luasrc/model/cbi/pbx-advanced.lua b/applications/luci-pbx/luasrc/model/cbi/pbx-advanced.lua
new file mode 100644
index 000000000..061982543
--- /dev/null
+++ b/applications/luci-pbx/luasrc/model/cbi/pbx-advanced.lua
@@ -0,0 +1,284 @@
+--[[
+    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 express 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 ~= "Express" 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", "Express")
+					commit = true
+				end
+			elseif s1._name == 'PBX-RTP' then
+				if s1.ports ~= rtprange or s1.target ~= "Express" 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", "Express")
+					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 which 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 at 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 SIP devices with. Please note that by default\
+	%s uses UDP port range %d to %d for RTP traffic (which carries voice), in case you need to configure\
+	NAT or QoS on another device.", appname, defaultrtpstart, defaultrtpend))
+
+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."))
+
+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 Name/Dynamic Domain Name"),
+	    translate("You should either have registered a domain name and have a static IP\
+	               address, or have configured Dynamic DNS on this router. Enter a\
+		       domain name which resolves to your external IP address."))
+h.datatype = "hostname"
+
+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) 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."))
+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\
+	    	       which 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-pbx/luasrc/model/cbi/pbx-calls.lua b/applications/luci-pbx/luasrc/model/cbi/pbx-calls.lua
new file mode 100644
index 000000000..029c5b06a
--- /dev/null
+++ b/applications/luci-pbx/luasrc/model/cbi/pbx-calls.lua
@@ -0,0 +1,359 @@
+--[[
+    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"
+
+-- This function builds and returns a table with all the entries in a given "module" and "section".
+function get_existing_entries(module, section)
+   i = 1
+   existing_entries = {}
+   m.uci:foreach(module, section,
+		 function(s1)
+		    existing_entries[i] = s1
+		    i = i + 1
+		 end)
+   return existing_entries
+end
+
+-- This function is used to build and return a table where the name field for
+-- every element in a table entries are the indexes to the table (used to check
+-- validity of user input.
+function get_valid_names(entries)
+   validnames={}
+   for index,value in ipairs(entries) do
+      validnames[entries[index].name] = true
+   end
+   return validnames
+end
+
+-- This function is used to build and return a table where the defaultuser field for
+-- every element in a table entries are the indexes to the table (used to check
+-- validity of user input.
+function get_valid_defaultusers(entries)
+   validnames={}
+   for index,value in ipairs(entries) do
+      validnames[entries[index].defaultuser] = true
+   end
+   return validnames
+end
+
+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 which 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
+
+----------------------------------------------------------------------------------------------------
+s = m:section(NamedSection, "outgoing_calls", "call_routing", translate("Outgoing Calls"),
+	translate("If you have more than one account which can make outgoing calls, you\
+		should enter a list of phone numbers and 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. 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."))
+s.anonymous = true
+
+m.uci:foreach(googlemodulename, "gtalk_jabber", 
+	      function(s1)
+		 if s1.username ~= nil and s1.name ~= nil and
+		    s1.make_outgoing_calls == "yes" then
+		    patt = s:option(DynamicList, s1.name, s1.username)
+
+		    -- If the saved field is empty, we return a string
+		    -- telling the user that this account would dial any exten.
+		    function patt.cfgvalue(self, section)
+		       value = self.map:get(section, self.option)
+
+		       if value == nil then
+			  return {"Dials any number"}
+		       else
+			  return value
+		       end
+		    end
+
+		    -- Write only valid extensions into the config file.
+		    function patt.write(self, section, value)
+		       newvalue = {}
+		       nindex = 1
+		       for index, field in ipairs(value) do
+			  if is_valid_extension(value[index]) == true then
+			     newvalue[nindex] = value[index]
+			     nindex = nindex + 1
+			  end
+		       end
+		       DynamicList.write(self, section, newvalue)
+		    end
+		 end
+	      end)
+
+m.uci:foreach(voipmodulename, "voip_provider", 
+	      function(s1)
+		 if s1.defaultuser ~= nil and s1.host ~= nil and 
+		    s1.name ~= nil        and s1.make_outgoing_calls == "yes" then
+		    patt = s:option(DynamicList, s1.name, s1.defaultuser .. "@" .. s1.host)
+
+		    -- If the saved field is empty, we return a string
+		    -- telling the user that this account would dial any exten.
+		    function patt.cfgvalue(self, section)
+		       value = self.map:get(section, self.option)
+
+		       if value == nil then
+			  return {"Dials any number"}
+		       else
+			  return value
+		       end
+		    end
+
+		    -- Write only valid extensions into the config file.
+		    function patt.write(self, section, value)
+		       newvalue = {}
+		       nindex = 1
+		       for index, field in ipairs(value) do
+			  if is_valid_extension(value[index]) == true then
+			     newvalue[nindex] = value[index]
+			     nindex = nindex + 1
+			  end
+		       end
+		       DynamicList.write(self, section, newvalue)
+		    end
+		 end
+	      end)
+
+----------------------------------------------------------------------------------------------------
+s = m:section(NamedSection, "incoming_calls", "call_routing", translate("Incoming Calls"),
+	translate("For each provider that receives calls, here you can restrict which users to ring\
+		on incoming calls. If the list is empty, the system will indicate that all users\
+		which are 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,	so this way, you can make users ring only for select providers.\
+		Entries can be made in a space-separated list, and/or one per\
+		line by hitting enter after every one."))
+s.anonymous = true
+
+m.uci:foreach(googlemodulename, "gtalk_jabber", 
+	      function(s1)
+		 if s1.username ~= nil and s1.register == "yes" then
+		    field_name=string.gsub(s1.username, "%W", "_")
+		    gtalkaccts = s:option(DynamicList, field_name, s1.username)
+		    
+		    -- If the saved field is empty, we return a string
+		    -- telling the user that this account would dial any exten.
+		    function gtalkaccts.cfgvalue(self, section)
+		       value = self.map:get(section, self.option)
+		       
+		       if value == nil then
+			  return {"Rings all users"}
+		       else
+			  return value
+		       end
+		    end
+
+		    -- Write only valid user names.
+		    function gtalkaccts.write(self, section, value)
+		       users=get_valid_defaultusers(get_existing_entries(usersmodulename, "local_user"))
+		       newvalue = {}
+		       nindex = 1
+		       for index, field in ipairs(value) do
+			  if users[value[index]] == true then
+			     newvalue[nindex] = value[index]
+			     nindex = nindex + 1
+			  end
+		       end
+		       DynamicList.write(self, section, newvalue)
+		    end
+		 end
+	      end)
+
+
+m.uci:foreach(voipmodulename, "voip_provider", 
+	      function(s1)
+		 if s1.defaultuser ~= nil and s1.host ~= nil and s1.register == "yes" then
+		    field_name=string.gsub(s1.defaultuser .. "_" .. s1.host, "%W", "_")
+		    voipaccts = s:option(DynamicList, field_name, s1.defaultuser .. "@" .. s1.host)
+
+		    -- If the saved field is empty, we return a string
+		    -- telling the user that this account would dial any exten.
+		    function voipaccts.cfgvalue(self, section)
+		       value = self.map:get(section, self.option)
+		       
+		       if value == nil then
+			  return {"Rings all users"}
+		       else
+			  return value
+		       end
+		    end
+
+		    -- Write only valid user names.
+		    function voipaccts.write(self, section, value)
+		       users=get_valid_defaultusers(get_existing_entries(usersmodulename, "local_user"))
+		       newvalue = {}
+		       nindex = 1
+		       for index, field in ipairs(value) do
+			  if users[value[index]] == true then
+			     newvalue[nindex] = value[index]
+			     nindex = nindex + 1
+			  end
+		       end
+		       DynamicList.write(self, section, newvalue)
+		    end
+		 end
+	      end)
+
+----------------------------------------------------------------------------------------------------
+s = m:section(NamedSection, "providers_user_can_use", "call_routing",
+      translate("Providers Used for Outgoing Calls"),
+      translate("If you would like, you could restrict which providers users are allowed to 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 will be rejected silently. Also, any entries automatically\
+	change to this PBX's internal naming scheme, with \"_\" replacing all non-alphanumeric characters.\
+	Entries can be made in a space-separated list, and/or one per line by hitting enter after every\
+	one."))
+s.anonymous = true
+
+m.uci:foreach(usersmodulename, "local_user", 
+	      function(s1)
+		 if s1.defaultuser ~= nil and s1.can_call == "yes" then
+		    providers = s:option(DynamicList, s1.defaultuser, s1.defaultuser)
+		    
+		    -- If the saved field is empty, we return a string
+		    -- telling the user that this account would dial any exten.
+		    function providers.cfgvalue(self, section)
+		       value = self.map:get(section, self.option)
+		       
+		       if value == nil then
+			  return {"Uses all provider accounts"}
+		       else
+			  return value
+		       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)
+		       validvoip=get_valid_names(get_existing_entries(voipmodulename, "voip_provider"))
+		       validgoog=get_valid_names(get_existing_entries(googlemodulename, "gtalk_jabber"))
+		       cookedvalue = {}
+		       cindex = 1
+		       for index, field in ipairs(value) do
+			  cooked = string.gsub(value[index], "%W", "_")
+			  if validvoip[cooked] == true or validgoog[cooked] == true then
+			     cookedvalue[cindex] = string.gsub(value[index], "%W", "_")
+			     cindex = cindex + 1
+			  end
+		       end
+		       DynamicList.write(self, section, cookedvalue)
+		    end
+		 end
+	      end)
+
+----------------------------------------------------------------------------------------------------
+s = m:section(TypedSection, "callthrough_numbers", translate("Call-through Numbers"),
+	translate("Designate numbers which will be allowed to call through this system and which user's\
+		  privileges it will have."))
+s.anonymous = true
+s.addremove = true
+
+num = s:option(DynamicList, "callthrough_number_list", translate("Call-through Numbers"))
+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 outwith this user's providers.\
+		   Invalid usernames are dropped silently, please verify that the entry was accepted."))
+function user.write(self, section, value)
+   users=get_valid_defaultusers(get_existing_entries(usersmodulename, "local_user"))
+   if users[value] == true then
+      Value.write(self, section, value)
+   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(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-pbx/luasrc/model/cbi/pbx-google.lua b/applications/luci-pbx/luasrc/model/cbi/pbx-google.lua
new file mode 100644
index 000000000..128c7a307
--- /dev/null
+++ b/applications/luci-pbx/luasrc/model/cbi/pbx-google.lua
@@ -0,0 +1,120 @@
+--[[
+    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). 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 which 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 (See Status, Message 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("Account 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("Account 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-pbx/luasrc/model/cbi/pbx-users.lua b/applications/luci-pbx/luasrc/model/cbi/pbx-users.lua
new file mode 100644
index 000000000..7c25308cd
--- /dev/null
+++ b/applications/luci-pbx/luasrc/model/cbi/pbx-users.lua
@@ -0,0 +1,148 @@
+--[[
+    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, which you\
+		will use to register with this service. Use this account either in an analog telephony\
+		adapter (ATA), or in a SIP softphone like CSipSimple, Linphone, or Sipdroid on your\
+		Android smartphone, or X-lite or Ekiga 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)
+	--allusers = ""
+	--ringusers = ""
+	--
+	---- Create two lists of users - one of all users and one of users enabled for incoming calls.
+	--m.uci:foreach(modulename, "local_user",
+	--	function(s1)
+	--		allusers = allusers .. " " .. s1.defaultuser
+	--		if s1.ring == "yes" then
+	--			ringusers = ringusers .. " " .. s1.defaultuser
+	--		end
+	--	end)
+	--
+	--m.uci:set (modulenamecalls, "valid_users", "allusers",  allusers)
+	--m.uci:set (modulenamecalls, "valid_users", "ringusers", ringusers)
+	--m.uci:commit (modulenamecalls)
+
+	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 which specifies the\
+	Server/Registrar port number. Beware that some devices have a confusing\
+	setting which sets the port where SIP requests originate from on the SIP\
+	device itself (bind port). The port specified on this page is NOT this bind port\
+	but the 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-pbx/luasrc/model/cbi/pbx-voip.lua b/applications/luci-pbx/luasrc/model/cbi/pbx-voip.lua
new file mode 100644
index 000000000..0be3deab6
--- /dev/null
+++ b/applications/luci-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 which 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-pbx/luasrc/model/cbi/pbx.lua b/applications/luci-pbx/luasrc/model/cbi/pbx.lua
new file mode 100644
index 000000000..f73930eee
--- /dev/null
+++ b/applications/luci-pbx/luasrc/model/cbi/pbx.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"
+
+function mysplit(inputstr, sep)
+        if sep == nil then
+                sep = "%s"
+        end
+        t={} ; i=1
+        for str in string.gmatch(inputstr, "([^"..sep.."]+)") do
+                t[i] = str
+                i = i + 1
+        end
+        return t
+end
+
+function format_two_indices(string, ind1, ind2)
+	lines=mysplit(string, "\n")
+
+	words={}
+	for index,value in ipairs(lines) do
+	        words[index]=mysplit(value)
+	end
+
+	output = ""
+	for index,value in ipairs(words) do
+	        if value[ind1] ~= nil and value[ind2] ~= nil then
+	                output = output .. string.format("%-40s \t %-20s\n", value[ind1], value[ind2])
+	        end
+	end
+	return output
+end
+
+function format_one_index(string, ind1)
+	lines=mysplit(string, "\n")
+
+	words={}
+	for index,value in ipairs(lines) do
+	        words[index]=mysplit(value)
+	end
+
+	output = ""
+	for index,value in ipairs(words) do
+	        if value[ind1] ~= nil then
+	                output = output .. string.format("%-40s\n", value[ind1])
+	        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 with, and sharing multiple Google and SIP (like Sipgate,\
+                   SipSorcery, and Betamax) accounts 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 configure at least one local SIP account\
+      on this PBX, to make and receive calls with your Google/SIP accounts.\
+      Configuring multiple users will allow you to make free calls between 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("Service Control and Connection 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
+      reg  = luci.sys.exec("asterisk -rx 'sip show registry' | sed 's/peer-//'")
+      jab  = 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_two_indices(reg, 1, 5) .. format_two_indices(jab, 2, 4) .. "\n" 
+             .. format_one_index(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-pbx/root/etc/config/pbx b/applications/luci-pbx/root/etc/config/pbx
new file mode 100644
index 000000000..ca7c1669d
--- /dev/null
+++ b/applications/luci-pbx/root/etc/config/pbx
@@ -0,0 +1 @@
+config 'main' 'connection_status'
diff --git a/applications/luci-pbx/root/etc/config/pbx-advanced b/applications/luci-pbx/root/etc/config/pbx-advanced
new file mode 100644
index 000000000..e7ac0e221
--- /dev/null
+++ b/applications/luci-pbx/root/etc/config/pbx-advanced
@@ -0,0 +1 @@
+config 'settings' 'advanced'
diff --git a/applications/luci-pbx/root/etc/config/pbx-calls b/applications/luci-pbx/root/etc/config/pbx-calls
new file mode 100644
index 000000000..822bd4a1b
--- /dev/null
+++ b/applications/luci-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-pbx/root/etc/config/pbx-google b/applications/luci-pbx/root/etc/config/pbx-google
new file mode 100644
index 000000000..e69de29bb
--- /dev/null
+++ b/applications/luci-pbx/root/etc/config/pbx-google
diff --git a/applications/luci-pbx/root/etc/config/pbx-users b/applications/luci-pbx/root/etc/config/pbx-users
new file mode 100644
index 000000000..a4277b1bf
--- /dev/null
+++ b/applications/luci-pbx/root/etc/config/pbx-users
@@ -0,0 +1 @@
+config 'user' 'server'
diff --git a/applications/luci-pbx/root/etc/config/pbx-voip b/applications/luci-pbx/root/etc/config/pbx-voip
new file mode 100644
index 000000000..e69de29bb
--- /dev/null
+++ b/applications/luci-pbx/root/etc/config/pbx-voip
diff --git a/applications/luci-pbx/root/etc/init.d/pbx-asterisk b/applications/luci-pbx/root/etc/init.d/pbx-asterisk
new file mode 100644
index 000000000..d96eb4f08
--- /dev/null
+++ b/applications/luci-pbx/root/etc/init.d/pbx-asterisk
@@ -0,0 +1,638 @@
+#!/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/>.
+
+. /etc/functions.sh
+
+START=60
+
+# Some global variables
+MODULENAME=pbx
+USERAGENT="PBX"
+DEFAULTOUTCNTXT=default-outgoing-call-context
+HANGUPCNTXT=hangup
+GTALKUNVL=unavailable
+
+ASTUSER=nobody
+ASTGROUP=nogroup
+ASTDIRSRECURSIVE="/var/run/asterisk /var/log/asterisk /var/spool/asterisk"
+ASTDIRS="/usr/lib/asterisk"
+
+FIREWALL_PATH="/etc/asterisk/firewall.$MODULENAME"
+
+TEMPLATEDIR=/etc/${MODULENAME}-asterisk
+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_EXTENSIONS=$TEMPLATEDIR/extensions.conf.TEMPLATE
+
+TMPL_EXTBLKLIST=$TEMPLATEDIR/extensions_blacklist.conf.TEMPLATE
+TMPL_EXTBLKLISTFTR=$TEMPLATEDIR/extensions_blacklist_footer.conf.TEMPLATE
+TMPL_EXTBLKLISTHDR=$TEMPLATEDIR/extensions_blacklist_header.conf.TEMPLATE
+
+TMPL_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_FIREWALL=$TEMPLATEDIR/firewall.$MODULENAME.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
+
+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"
+
+
+# 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=""
+
+# 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
+}
+
+# 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 $INCLUDED_FILES
+}
+
+# Puts together all the extensions.conf related configuration.
+pbx_create_extensions_config()
+{
+	sed "s/|LOCALUSERS|/$localusers_to_ring/g" $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
+	rm -f $WORKDIR/outext-*.TMP
+	rm -f $WORKDIR/localext.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 secret
+	local ring
+	local can_call
+
+	config_get fullname $1 fullname
+	config_get defaultuser $1 defaultuser
+	config_get secret $1 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 $defaultuser"
+		sed -i "s/|CONTEXTNAME|/$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
+
+	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
+	config_get secret $1 secret
+	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
+	config_get statusmessage $1 statusmessage
+
+	[ -z "$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 $@ ; do users_to_ring=$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 $numprefix ; do
+			sed "s/|NUMPREFIX|/$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
+	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
+	config_get secret $1 secret
+	config_get host $1 host
+	config_get port $1 port
+	config_get outbountproxy $1 outboundproxy
+	config_get fromdomain $1 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 $numprefix ; do
+			sed "s/|NUMPREFIX|/$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
+	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 $localusers_can_dial ; do
+		sed "s/|DEFAULTUSER|/$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 $blacklist2 ; do
+		sed "s/|BLACKLISTITEM|/$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
+
+	config_get callthrough_number_list $1 callthrough_number_list
+	config_get defaultuser $1 defaultuser
+	config_get pin $1 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 >> $WORKDIR/callthroughcheck.TMP
+
+		sed "s/|NUMBER|/$callthrough_number/g" $TMPL_EXTCTHRU |\
+		    sed "s/|DEFAULTUSER|/$defaultuser/"               |\
+		        sed "s/|PIN|/$pin/"                                >> $WORKDIR/callthrough.TMP
+	done
+}
+
+# Creates sip.conf from its template.
+pbx_cook_sip_template()
+{
+	local useragent
+	local externhost
+	local bindport
+
+	config_get useragent advanced useragent
+	config_get externhost advanced externhost
+	config_get bindport advanced bindport
+
+	[ -z "$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
+}
+
+# Makes sure the ownership of specified directories is proper.
+pbx_fix_ownership()
+{
+	chown $ASTUSER:$ASTGROUP    $ASTDIRS
+	chown $ASTUSER:$ASTGROUP -R $ASTDIRSRECURSIVE
+}
+
+# Creates firewall configuration. However, since this functionality is now
+# taken over by pbx-advanced.lua, this function is unused.
+create_firewall_config()
+{
+	local bindport
+	local rtpstart
+	local rtpend
+	local externhost
+
+	config_get bindport advanced bindport
+	config_get externhost advanced externhost
+	config_get rtpstart advanced rtpstart
+	config_get rtpend   advanced rtpend
+
+	# We need all of these parameters to be set in order to consider inserting firewall rules.
+	if [ -z "$externhost" -o -z "$bindport" -o -z "$rtpstart" -o -z "$rtpend" ] ; then
+		rm -f $FIREWALL_PATH
+		return
+	fi
+
+	sed "s/|SIPPORT|/$bindport/g" $TMPL_FIREWALL |\
+	     sed "s/|RTPRANGE|/$rtpstart:$rtpend/g" > $WORKDIR/firewall.$MODULENAME
+
+	# Make sure there is an include section in the firewall configuration
+	# to include the file we just created.
+	i=0 ; found_path=0
+	while p=`uci get firewall.@include[$i].path 2>/dev/null` ; do
+		if [ "$p" = "$FIREWALL_PATH" ] ; then
+			found_path=1
+			break
+		fi
+		i=`expr $i + 1`
+	done
+
+	# If no include section was found which mentions $FIREWALL_PATH, add one.
+	if [ $found_path -eq 0 ] ; then
+		uci add firewall include   1>/dev/null 2>/dev/null
+		uci set firewall.@include[-1].path="$FIREWALL_PATH"
+		uci commit firewall
+	fi
+}
+
+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, and phone blacklist.
+	config_unset
+	config_load ${MODULENAME}-calls
+	pbx_create_user_contexts
+	pbx_add_blacklist
+	config_foreach pbx_add_callthrough callthrough_numbers
+
+	# Prepare sip.conf using settings from the "advanced" section.
+	config_unset
+	config_load ${MODULENAME}-advanced
+	pbx_cook_sip_template
+	pbx_cook_rtp_template
+
+	# Assemble the configuration, and copy changed files over.
+	pbx_assemble_and_copy_config
+
+	# Enforce ownership of specified files and directories.
+	pbx_fix_ownership
+}
+
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE
new file mode 100644
index 000000000..ae6340af8
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE
@@ -0,0 +1,17 @@
+[directories]
+astetcdir => /etc/asterisk
+astmoddir => /usr/lib/asterisk/modules
+astvarlibdir => /var/lib/asterisk
+astdbdir => /usr/lib/asterisk
+astkeydir => /var/lib/asterisk
+astdatadir => /usr/share/asterisk
+astagidir => /usr/share/asterisk/agi-bin
+astspooldir => /var/spool/asterisk
+astrundir => /var/run/asterisk
+astlogdir => /var/log/asterisk
+
+[options]
+languageprefix = yes
+dumpcore = no
+runuser = nobody
+rungroup = nogroup
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE
new file mode 100644
index 000000000..f3edac384
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE
@@ -0,0 +1,30 @@
+[general]
+static = yes
+writeprotect = yes
+clearglobalvars = no
+
+[globals]
+
+[default]
+
+[context-user-hangup]
+exten =>   s,1,Hangup()
+exten => _X.,1,Hangup()
+
+[context-catch-all]
+exten   => _[!-~].,1,Dial(SIP/${EXTEN},60,r)
+
+[default-incoming-call-context]
+exten => s,1,NoOp(${CALLERID})
+exten => s,n,Set(SOURCECONTEXT=default-incoming-call-context)
+exten => s,n,Set(SOURCEEXTEN=s)
+exten => s,n,Goto(blacklist-call-context,s,1)
+exten => s,n(doneblacklist),NoOp()
+exten => s,n,Goto(disa-check-call-context,s,1)
+exten => s,n(donedisacheck),Dial(|LOCALUSERS|,60,r)
+
+#include extensions_incoming.conf
+#include extensions_incoming_gtalk.conf
+#include extensions_blacklist.conf
+#include extensions_callthrough.conf
+#include extensions_user.conf
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist.conf.TEMPLATE
new file mode 100644
index 000000000..54ee989b0
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist.conf.TEMPLATE
@@ -0,0 +1 @@
+exten => s,n,Gotoif($[ "${CALLERID(NUM)}" = "|BLACKLISTITEM|" ]?context-user-hangup,s,1)
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist_footer.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist_footer.conf.TEMPLATE
new file mode 100644
index 000000000..da964f238
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist_footer.conf.TEMPLATE
@@ -0,0 +1,2 @@
+exten => s,n,Goto(${SOURCECONTEXT},${SOURCEEXTEN},doneblacklist)
+
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist_header.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist_header.conf.TEMPLATE
new file mode 100644
index 000000000..de0e98465
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist_header.conf.TEMPLATE
@@ -0,0 +1,3 @@
+
+[blacklist-call-context]
+exten => s,1,Noop()
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check.conf.TEMPLATE
new file mode 100644
index 000000000..ba2379b73
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check.conf.TEMPLATE
@@ -0,0 +1 @@
+exten => s,n,Gotoif($[ "${CALLERID(NUM)}" =~ ".*|NUMBER|" ]?disa-call-context,|NUMBER|,1)
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check_footer.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check_footer.conf.TEMPLATE
new file mode 100644
index 000000000..74056fa01
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check_footer.conf.TEMPLATE
@@ -0,0 +1 @@
+exten => s,n,Goto(${SOURCECONTEXT},${SOURCEEXTEN},donedisacheck)
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check_header.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check_header.conf.TEMPLATE
new file mode 100644
index 000000000..e0d67b802
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check_header.conf.TEMPLATE
@@ -0,0 +1,2 @@
+[disa-check-call-context]
+exten => s,1,Noop()
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa.conf.TEMPLATE
new file mode 100644
index 000000000..e799d5210
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa.conf.TEMPLATE
@@ -0,0 +1,4 @@
+exten => |NUMBER|,1,Noop()
+exten => |NUMBER|,n,Authenticate(|PIN|)
+exten => |NUMBER|,n,DISA(no-password,context-user-|DEFAULTUSER|)
+
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa_header.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa_header.conf.TEMPLATE
new file mode 100644
index 000000000..a74227114
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa_header.conf.TEMPLATE
@@ -0,0 +1 @@
+[disa-call-context]
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk.conf.TEMPLATE
new file mode 100644
index 000000000..dc0c68add
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk.conf.TEMPLATE
@@ -0,0 +1,11 @@
+exten => |USERNAME|,1,NoOp(${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)=${CALLERID(name)})
+same => n,Set(SOURCECONTEXT=context-incoming-gtalk)
+same => n,Set(SOURCEEXTEN=|USERNAME|)
+same => n,Goto(blacklist-call-context,s,1)
+same => n(doneblacklist),NoOp()
+same => n,Goto(disa-check-call-context,s,1)
+same => n(donedisacheck),Dial(|LOCALUSERS|,60,D(:w11111111))
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk_header.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk_header.conf.TEMPLATE
new file mode 100644
index 000000000..f6e44a5bf
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk_header.conf.TEMPLATE
@@ -0,0 +1 @@
+[context-incoming-gtalk]
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE
new file mode 100644
index 000000000..12ecefc80
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE
@@ -0,0 +1,9 @@
+
+[context-incoming-|NAME|]
+exten => s,1,NoOp(${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(disa-check-call-context,s,1)
+exten => s,n(donedisacheck),Dial(|LOCALUSERS|,60,r)
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_context_header.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_context_header.conf.TEMPLATE
new file mode 100644
index 000000000..a7f36cf91
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_context_header.conf.TEMPLATE
@@ -0,0 +1,2 @@
+
+[context-outgoing-|NAME|]
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_dial_local_user.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_dial_local_user.conf.TEMPLATE
new file mode 100644
index 000000000..b81da9eab
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_dial_local_user.conf.TEMPLATE
@@ -0,0 +1 @@
+exten =>     |DEFAULTUSER|,1,Dial(SIP/|DEFAULTUSER|,60,r)
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_gtalk.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_gtalk.conf.TEMPLATE
new file mode 100644
index 000000000..259c2ceaa
--- /dev/null
+++ b/applications/luci-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-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_gtalk.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_gtalk.conf.TEMPLATE
new file mode 100644
index 000000000..1fa7713e2
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_gtalk.conf.TEMPLATE
@@ -0,0 +1,2 @@
+exten => |PATTERN|,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN|SYMBOLSTOREMOVE|}@voice.google.com,60)
+
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_sip.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_sip.conf.TEMPLATE
new file mode 100644
index 000000000..178b6deaa
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_sip.conf.TEMPLATE
@@ -0,0 +1 @@
+exten => |PATTERN|,1,Dial(SIP/${EXTEN|SYMBOLSTOREMOVE|}@peer-|NAME|,60,r)
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_sip.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_sip.conf.TEMPLATE
new file mode 100644
index 000000000..9b1d9addc
--- /dev/null
+++ b/applications/luci-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-pbx/root/etc/pbx-asterisk/extensions_user_context_footer.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_user_context_footer.conf.TEMPLATE
new file mode 100644
index 000000000..7ce1060dc
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_user_context_footer.conf.TEMPLATE
@@ -0,0 +1 @@
+include => context-catch-all
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/extensions_user_context_header.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_user_context_header.conf.TEMPLATE
new file mode 100644
index 000000000..5931eaf28
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/extensions_user_context_header.conf.TEMPLATE
@@ -0,0 +1,3 @@
+
+[context-user-|DEFAULTUSER|]
+
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/firewall.home-pbx.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/firewall.home-pbx.TEMPLATE
new file mode 100644
index 000000000..91e9f66d8
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/firewall.home-pbx.TEMPLATE
@@ -0,0 +1,2 @@
+iptables -I INPUT 3 -p udp -s 0/0 -d 0/0 --dport |SIPPORT|  -j ACCEPT
+iptables -I INPUT 3 -p udp -s 0/0 -d 0/0 --dport |RTPRANGE| -j ACCEPT
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/gtalk.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/gtalk.conf.TEMPLATE
new file mode 100644
index 000000000..4f07a7166
--- /dev/null
+++ b/applications/luci-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-pbx/root/etc/pbx-asterisk/indications.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/indications.conf.TEMPLATE
new file mode 100644
index 000000000..d7088db7c
--- /dev/null
+++ b/applications/luci-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-pbx/root/etc/pbx-asterisk/jabber.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/jabber.conf.TEMPLATE
new file mode 100644
index 000000000..cf71e1f8f
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/jabber.conf.TEMPLATE
@@ -0,0 +1,4 @@
+[general]
+autoregister=yes
+
+#include jabber_users.conf
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/jabber_users.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/jabber_users.conf.TEMPLATE
new file mode 100644
index 000000000..3ee2463ed
--- /dev/null
+++ b/applications/luci-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-pbx/root/etc/pbx-asterisk/logger.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/logger.conf.TEMPLATE
new file mode 100644
index 000000000..e57325013
--- /dev/null
+++ b/applications/luci-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-pbx/root/etc/pbx-asterisk/manager.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/manager.conf.TEMPLATE
new file mode 100644
index 000000000..2ac2f0033
--- /dev/null
+++ b/applications/luci-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-pbx/root/etc/pbx-asterisk/modules.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/modules.conf.TEMPLATE
new file mode 100644
index 000000000..3594dad43
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/modules.conf.TEMPLATE
@@ -0,0 +1,31 @@
+[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 => 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_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
+
+[global]
+chan_modem.so=no
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/rtp.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/rtp.conf.TEMPLATE
new file mode 100644
index 000000000..10d577d3a
--- /dev/null
+++ b/applications/luci-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-pbx/root/etc/pbx-asterisk/sip.conf.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/sip.conf.TEMPLATE
new file mode 100644
index 000000000..6069554e0
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/sip.conf.TEMPLATE
@@ -0,0 +1,37 @@
+[general]
+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|
+
+#include sip_registrations.conf
+
+[authentication]
+
+#include sip_peers.conf
+#include sip_users.conf
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/sip_peer.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/sip_peer.TEMPLATE
new file mode 100644
index 000000000..30abaadd5
--- /dev/null
+++ b/applications/luci-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-pbx/root/etc/pbx-asterisk/sip_registration.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/sip_registration.TEMPLATE
new file mode 100644
index 000000000..e139d43f0
--- /dev/null
+++ b/applications/luci-pbx/root/etc/pbx-asterisk/sip_registration.TEMPLATE
@@ -0,0 +1,2 @@
+register => |DEFAULTUSER|:|SECRET|@peer-|NAME|
+
diff --git a/applications/luci-pbx/root/etc/pbx-asterisk/sip_user.TEMPLATE b/applications/luci-pbx/root/etc/pbx-asterisk/sip_user.TEMPLATE
new file mode 100644
index 000000000..61a8b0b86
--- /dev/null
+++ b/applications/luci-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