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GNU GENERAL PUBLIC LICENSE
Version 3, 29 June 2007
Copyright (C) 2007 Free Software Foundation, Inc. <http://fsf.org/>
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PO = pbx
include ../../build/config.mk
include ../../build/module.mk

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--[[
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

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--[[
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

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--[[
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

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--[[
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

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--[[
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

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--[[
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

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--[[
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

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applications/luci-pbx/root/etc/config/pbx Normal file
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config 'main' 'connection_status'

1
applications/luci-pbx/root/etc/config/pbx-advanced Normal file
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config 'settings' 'advanced'

7
applications/luci-pbx/root/etc/config/pbx-calls Normal file
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config 'call_routing' 'outgoing_calls'
config 'call_routing' 'incoming_calls'
config 'call_routing' 'providers_user_can_use'
config 'call_routing' 'blacklisting'

0
applications/luci-pbx/root/etc/config/pbx-google Normal file
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1
applications/luci-pbx/root/etc/config/pbx-users Normal file
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config 'user' 'server'

0
applications/luci-pbx/root/etc/config/pbx-voip Normal file
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638
applications/luci-pbx/root/etc/init.d/pbx-asterisk Normal file
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#!/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
}

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applications/luci-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE Normal file
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@ -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

30
applications/luci-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE Normal file
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@ -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

1
applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist.conf.TEMPLATE Normal file
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@ -0,0 +1 @@
exten => s,n,Gotoif($[ "${CALLERID(NUM)}" = "|BLACKLISTITEM|" ]?context-user-hangup,s,1)

2
applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist_footer.conf.TEMPLATE Normal file
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exten => s,n,Goto(${SOURCECONTEXT},${SOURCEEXTEN},doneblacklist)

3
applications/luci-pbx/root/etc/pbx-asterisk/extensions_blacklist_header.conf.TEMPLATE Normal file
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@ -0,0 +1,3 @@
[blacklist-call-context]
exten => s,1,Noop()

1
applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check.conf.TEMPLATE Normal file
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@ -0,0 +1 @@
exten => s,n,Gotoif($[ "${CALLERID(NUM)}" =~ ".*|NUMBER|" ]?disa-call-context,|NUMBER|,1)

1
applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check_footer.conf.TEMPLATE Normal file
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exten => s,n,Goto(${SOURCECONTEXT},${SOURCEEXTEN},donedisacheck)

2
applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa-check_header.conf.TEMPLATE Normal file
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[disa-check-call-context]
exten => s,1,Noop()

4
applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa.conf.TEMPLATE Normal file
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exten => |NUMBER|,1,Noop()
exten => |NUMBER|,n,Authenticate(|PIN|)
exten => |NUMBER|,n,DISA(no-password,context-user-|DEFAULTUSER|)

1
applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa_header.conf.TEMPLATE Normal file
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[disa-call-context]

11
applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk.conf.TEMPLATE Normal file
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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))

1
applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_gtalk_header.conf.TEMPLATE Normal file
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@ -0,0 +1 @@
[context-incoming-gtalk]

9
applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE Normal file
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@ -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)

2
applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_context_header.conf.TEMPLATE Normal file
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[context-outgoing-|NAME|]

1
applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_dial_local_user.conf.TEMPLATE Normal file
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exten => |DEFAULTUSER|,1,Dial(SIP/|DEFAULTUSER|,60,r)

9
applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_gtalk.conf.TEMPLATE Normal file
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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)

2
applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_gtalk.conf.TEMPLATE Normal file
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exten => |PATTERN|,1,Dial(Gtalk/gtalk-|NAME|/${EXTEN|SYMBOLSTOREMOVE|}@voice.google.com,60)

1
applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_pattern_sip.conf.TEMPLATE Normal file
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exten => |PATTERN|,1,Dial(SIP/${EXTEN|SYMBOLSTOREMOVE|}@peer-|NAME|,60,r)

8
applications/luci-pbx/root/etc/pbx-asterisk/extensions_outgoing_sip.conf.TEMPLATE Normal file
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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)

1
applications/luci-pbx/root/etc/pbx-asterisk/extensions_user_context_footer.conf.TEMPLATE Normal file
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include => context-catch-all

3
applications/luci-pbx/root/etc/pbx-asterisk/extensions_user_context_header.conf.TEMPLATE Normal file
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[context-user-|DEFAULTUSER|]

2
applications/luci-pbx/root/etc/pbx-asterisk/firewall.home-pbx.TEMPLATE Normal file
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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

10
applications/luci-pbx/root/etc/pbx-asterisk/gtalk.conf.TEMPLATE Normal file
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[general]
context=context-incoming-gtalk
allowguest=yes
allowguests=yes
bindaddr=0.0.0.0
[guest]
disallow=all
allow=ulaw
context=context-incoming-gtalk

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applications/luci-pbx/root/etc/pbx-asterisk/indications.conf.TEMPLATE Normal file
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; 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

4
applications/luci-pbx/root/etc/pbx-asterisk/jabber.conf.TEMPLATE Normal file
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[general]
autoregister=yes
#include jabber_users.conf

8
applications/luci-pbx/root/etc/pbx-asterisk/jabber_users.conf.TEMPLATE Normal file
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[gtalk-|NAME|]
type=client
serverhost=talk.google.com
username=|USERNAME|/Talk
secret=|SECRET|
timeout=150
status=|STATUS|
statusmessage=|STATUSMESSAGE|

7
applications/luci-pbx/root/etc/pbx-asterisk/logger.conf.TEMPLATE Normal file
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[general]
queue_log = no
event_log = no
[logfiles]
console => notice,warning,error
messages => error

7
applications/luci-pbx/root/etc/pbx-asterisk/manager.conf.TEMPLATE Normal file
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[general]
enabled = no
port = 5038
bindaddr = 0.0.0.0

31
applications/luci-pbx/root/etc/pbx-asterisk/modules.conf.TEMPLATE Normal file
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[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

6
applications/luci-pbx/root/etc/pbx-asterisk/rtp.conf.TEMPLATE Normal file
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[general]
rtpstart=|RTPSTART|
rtpend=|RTPEND|
rtpchecksums=no
dtmftimeout=3000
rtcpinterval = 2000

37
applications/luci-pbx/root/etc/pbx-asterisk/sip.conf.TEMPLATE Normal file
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[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

13
applications/luci-pbx/root/etc/pbx-asterisk/sip_peer.TEMPLATE Normal file
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[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|

2
applications/luci-pbx/root/etc/pbx-asterisk/sip_registration.TEMPLATE Normal file
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register => |DEFAULTUSER|:|SECRET|@peer-|NAME|

11
applications/luci-pbx/root/etc/pbx-asterisk/sip_user.TEMPLATE Normal file
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[|DEFAULTUSER|]
fullname = |FULLNAME|
defaultuser = |DEFAULTUSER|
secret = |SECRET|
hassip = yes
hasvoicemail = no
host = dynamic
type = friend
context = context-user-|CONTEXTNAME|
qualify = no