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GNU GENERAL PUBLIC LICENSE
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7
applications/luci-pbx/CREDITS-SOUNDS Normal file
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This file pertains to the sounds files included in root/etc/pbx-asterisk/sounds
Recorded by:
Allison Smith (http://www.theivrvoice.com)
Financial Contributions by:
Digium, Inc. (http://www.digium.com)

312
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This file pertains to the sounds files included in root/etc/pbx-asterisk/sounds
LICENSE FOR VOICE PROMPT FILES
------------------------------
The voice prompt files distributed herewith are Copyright (C) 2003-2008
Allison Smith, and provided under terms of the following License. For
more information, or to purchase custom voice prompt files, please
visit:
http://www.digium.com/ivr or http://www.theasteriskvoice.com
LICENSE
-------
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c. "Derivative Work" means a work based upon the Work or upon the Work
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4
applications/luci-pbx/Makefile Normal file
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PO = pbx
include ../../build/config.mk
include ../../build/module.mk

29
applications/luci-pbx/luasrc/controller/pbx.lua Normal file
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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", "pbx-advanced"}, cbi("pbx-advanced"), "Advanced Settings", 6)
end

291
applications/luci-pbx/luasrc/model/cbi/pbx-advanced.lua Normal file
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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 priority service level, insert this rule.
commit = false
m.uci:foreach("qos", "classify",
function(s1)
if s1._name == 'PBX-SIP' then
if s1.ports ~= bindport or s1.target ~= "Priority" or s1.proto ~= "udp" then
m.uci:set("qos", s1['.name'], "ports", bindport)
m.uci:set("qos", s1['.name'], "proto", "udp")
m.uci:set("qos", s1['.name'], "target", "Priority")
commit = true
end
elseif s1._name == 'PBX-RTP' then
if s1.ports ~= rtprange or s1.target ~= "Priority" or s1.proto ~= "udp" then
m.uci:set("qos", s1['.name'], "ports", rtprange)
m.uci:set("qos", s1['.name'], "proto", "udp")
m.uci:set("qos", s1['.name'], "target", "Priority")
commit = true
end
end
end)
-- If something changed, then we commit the qos config.
if commit == true then m.uci:commit("qos") end
end
-- This function is a (so far) unsuccessful attempt to manipulate the firewall rules from here
-- Need to do more testing and eventually move to this mode.
function maintain_firewall_rules()
-- Get the network information
bindport, rtprange, ipaddr, externhost = get_network_info()
commit = false
-- Only if externhost is set, do we control firewall rules.
if externhost ~= nil and bindport ~= nil and rtprange ~= nil then
-- Insert empty PBX-SIP and PBX-RTP rules if not present.
insert_empty_sip_rtp_rules ("firewall", "rule")
-- Iterate through the firewall rules, and if the dest_port and dest_ip setting of the\
-- SIP and RTP rule do not match what we want configured, set all the entries in the rule\
-- appropriately.
m.uci:foreach("firewall", "rule",
function(s1)
if s1._name == 'PBX-SIP' then
if s1.dest_port ~= bindport then
m.uci:set("firewall", s1['.name'], "dest_port", bindport)
m.uci:set("firewall", s1['.name'], "src", "wan")
m.uci:set("firewall", s1['.name'], "proto", "udp")
m.uci:set("firewall", s1['.name'], "target", "ACCEPT")
commit = true
end
elseif s1._name == 'PBX-RTP' then
if s1.dest_port ~= rtprange then
m.uci:set("firewall", s1['.name'], "dest_port", rtprange)
m.uci:set("firewall", s1['.name'], "src", "wan")
m.uci:set("firewall", s1['.name'], "proto", "udp")
m.uci:set("firewall", s1['.name'], "target", "ACCEPT")
commit = true
end
end
end)
else
-- We delete the firewall rules if one or more of the necessary parameters are not set.
sip_rule_name=nil
rtp_rule_name=nil
-- First discover the configuration names of the rules.
m.uci:foreach("firewall", "rule",
function(s1)
if s1._name == 'PBX-SIP' then
sip_rule_name = s1['.name']
elseif s1._name == 'PBX-RTP' then
rtp_rule_name = s1['.name']
end
end)
-- Then, using the names, actually delete the rules.
if sip_rule_name ~= nil then
m.uci:delete("firewall", sip_rule_name)
commit = true
end
if rtp_rule_name ~= nil then
m.uci:delete("firewall", rtp_rule_name)
commit = true
end
end
-- If something changed, then we commit the firewall config.
if commit == true then m.uci:commit("firewall") end
end
m = Map (modulename, translate("Advanced Settings"),
translate("This section contains settings that do not need to be changed under \
normal circumstances. In addition, here you can configure your system \
for use with remote SIP devices, and resolve call quality issues by enabling \
the insertion of QoS rules."))
-- Recreate the voip server config, and restart necessary services after changes are commited
-- to the advanced configuration. The firewall must restart because of "Remote Usage".
function m.on_after_commit(self)
-- Make sure firewall rules are in place
maintain_firewall_rules()
-- If insertion of QoS rules is enabled
if m.uci:get(modulename, "advanced", "qos_enabled") == "yes" then
insert_qos_rules()
else
delete_qos_rules()
end
luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null")
luci.sys.call("/etc/init.d/" .. server .. " restart 1\>/dev/null 2\>/dev/null")
luci.sys.call("/etc/init.d/firewall restart 1\>/dev/null 2\>/dev/null")
end
-----------------------------------------------------------------------------
s = m:section(NamedSection, "advanced", "settings", translate("Advanced Settings"))
s.anonymous = true
s:tab("general", translate("General Settings"))
s:tab("remote_usage", translate("Remote Usage"),
translatef("You can use your SIP devices/softphones with this system from a remote location \
as well, as long as your Internet Service Provider gives you a public IP. \
You will be able to call other local users for free (e.g. other Analog Telephone Adapters (ATAs)) \
and use your VoIP providers to make calls as if you were local to the PBX. \
After configuring this tab, go back to where users are configured and see the new \
Server and Port setting you need to configure the remote SIP devices with. Please note that if this \
PBX is not running on your router/gateway, you will need to configure port forwarding (NAT) on your \
router/gateway. Please forward the ports below (SIP port and RTP range) to the IP address of the \
device running this PBX."))
s:tab("qos", translate("QoS Settings"),
translate("If you experience jittery or high latency audio during heavy downloads, you may want \
to enable QoS. QoS prioritizes traffic to and from your network for specified ports and IP \
addresses, resulting in better latency and throughput for sound in our case. If enabled below, \
a QoS rule for this service will be configured by the PBX automatically, but you must visit the \
QoS configuration page (Network->QoS) to configure other critical QoS settings like Download \
and Upload speed."))
ringtime = s:taboption("general", Value, "ringtime", translate("Number of Seconds to Ring"),
translate("Set the number of seconds to ring users upon incoming calls before hanging up \
or going to voicemail, if the voicemail is installed and enabled."))
ringtime.default = 30
ua = s:taboption("general", Value, "useragent", translate("User Agent String"),
translate("This is the name that the VoIP server will use to identify itself when \
registering to VoIP (SIP) providers. Some providers require this to a specific \
string matching a hardware SIP device."))
ua.default = appname
h = s:taboption("remote_usage", Value, "externhost", translate("Domain/IP Address/Dynamic Domain"),
translate("You can enter your domain name, external IP address, or dynamic domain name here \
Please keep in mind that if your IP address is dynamic and it changes your configuration \
will become invalid. Hence, it's recommended to set up Dynamic DNS in this case."))
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 \
that will be used for setting up RTP communication. It's usually OK to leave this \
at the default value."))
p.datatype = "port"
p.default = defaultrtpstart
p = s:taboption("remote_usage", Value, "rtpend", translate("RTP Port Range End"))
p.datatype = "port"
p.default = defaultrtpend
p = s:taboption("qos", ListValue, "qos_enabled", translate("Insert QoS Rules"))
p:value("yes", translate("Yes"))
p:value("no", translate("No"))
p.default = "yes"
return m

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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"
allvalidaccounts = {}
nallvalidaccounts = 0
validoutaccounts = {}
nvalidoutaccounts = 0
validinaccounts = {}
nvalidinaccounts = 0
allvalidusers = {}
nallvalidusers = 0
validoutusers = {}
nvalidoutusers = 0
-- Checks whether the entered extension is valid syntactically.
function is_valid_extension(exten)
return (exten:match("[#*+0-9NXZ]+$") ~= nil)
end
m = Map (modulename, translate("Call Routing"),
translate("This is where you indicate which Google/SIP accounts are used to call what \
country/area codes, which users can use what SIP/Google accounts, how incoming \
calls are routed, what numbers can get into this PBX with a password, and what \
numbers are blacklisted."))
-- Recreate the config, and restart services after changes are commited to the configuration.
function m.on_after_commit(self)
luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null")
luci.sys.call("/etc/init.d/" .. server .. " restart 1\>/dev/null 2\>/dev/null")
end
-- Add Google accounts to all valid accounts, and accounts valid for incoming and outgoing calls.
m.uci:foreach(googlemodulename, "gtalk_jabber",
function(s1)
-- Add this provider to list of valid accounts.
if s1.username ~= nil and s1.name ~= nil then
allvalidaccounts[s1.name] = s1.username
nallvalidaccounts = nallvalidaccounts + 1
if s1.make_outgoing_calls == "yes" then
-- Add provider to the associative array of valid outgoing accounts.
validoutaccounts[s1.name] = s1.username
nvalidoutaccounts = nvalidoutaccounts + 1
end
if s1.register == "yes" then
-- Add provider to the associative array of valid outgoing accounts.
validinaccounts[s1.name] = s1.username
nvalidinaccounts = nvalidinaccounts + 1
end
end
end)
-- Add SIP accounts to all valid accounts, and accounts valid for incoming and outgoing calls.
m.uci:foreach(voipmodulename, "voip_provider",
function(s1)
-- Add this provider to list of valid accounts.
if s1.defaultuser ~= nil and s1.host ~= nil and s1.name ~= nil then
allvalidaccounts[s1.name] = s1.defaultuser .. "@" .. s1.host
nallvalidaccounts = nallvalidaccounts + 1
if s1.make_outgoing_calls == "yes" then
-- Add provider to the associative array of valid outgoing accounts.
validoutaccounts[s1.name] = s1.defaultuser .. "@" .. s1.host
nvalidoutaccounts = nvalidoutaccounts + 1
end
if s1.register == "yes" then
-- Add provider to the associative array of valid outgoing accounts.
validinaccounts[s1.name] = s1.defaultuser .. "@" .. s1.host
nvalidinaccounts = nvalidinaccounts + 1
end
end
end)
-- Add Local User accounts to all valid users, and users allowed to make outgoing calls.
m.uci:foreach(usersmodulename, "local_user",
function(s1)
-- Add user to list of all valid users.
if s1.defaultuser ~= nil then
allvalidusers[s1.defaultuser] = true
nallvalidusers = nallvalidusers + 1
if s1.can_call == "yes" then
validoutusers[s1.defaultuser] = true
nvalidoutusers = nvalidoutusers + 1
end
end
end)
----------------------------------------------------------------------------------------------------
-- If there are no accounts configured, or no accounts enabled for outgoing calls, display a warning.
-- Otherwise, display the usual help text within the section.
if nallvalidaccounts == 0 then
text = translate("NOTE: There are no Google or SIP provider accounts configured.")
elseif nvalidoutaccounts == 0 then
text = translate("NOTE: There are no Google or SIP provider accounts enabled for outgoing calls.")
else
text = translate("If you have more than one account that can make outgoing calls, you \
should enter a list of phone numbers and/or prefixes in the following fields for each \
provider listed. Invalid prefixes are removed silently, and only 0-9, X, Z, N, #, *, \
and + are valid characters. The letter X matches 0-9, Z matches 1-9, and N matches 2-9. \
For example to make calls to Germany through a provider, you can enter 49. To make calls \
to North America, you can enter 1NXXNXXXXXX. If one of your providers can make \"local\" \
calls to an area code like New York's 646, you can enter 646NXXXXXX for that \
provider. You should leave one account with an empty list to make calls with \
it by default, if no other provider's prefixes match. The system will automatically \
replace an empty list with a message that the provider dials all numbers not matched by another \
provider's prefixes. Be as specific as possible (i.e. 1NXXNXXXXXX is better than 1). Please note \
all international dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a \
space-separated list, and/or one per line by hitting enter after every one.")
end
s = m:section(NamedSection, "outgoing_calls", "call_routing", translate("Outgoing Calls"), text)
s.anonymous = true
for k,v in pairs(validoutaccounts) do
patterns = s:option(DynamicList, k, v)
-- If the saved field is empty, we return a string
-- telling the user that this account would dial any exten.
function patterns.cfgvalue(self, section)
value = self.map:get(section, self.option)
if value == nil then
return {translate("Dials numbers unmatched elsewhere")}
else
return value
end
end
-- Write only valid extensions into the config file.
function patterns.write(self, section, value)
newvalue = {}
nindex = 1
for index, field in ipairs(value) do
val = luci.util.trim(value[index])
if is_valid_extension(val) == true then
newvalue[nindex] = val
nindex = nindex + 1
end
end
DynamicList.write(self, section, newvalue)
end
end
----------------------------------------------------------------------------------------------------
-- If there are no accounts configured, or no accounts enabled for incoming calls, display a warning.
-- Otherwise, display the usual help text within the section.
if nallvalidaccounts == 0 then
text = translate("NOTE: There are no Google or SIP provider accounts configured.")
elseif nvalidinaccounts == 0 then
text = translate("NOTE: There are no Google or SIP provider accounts enabled for incoming calls.")
else
text = translate("For each provider enabled for incoming calls, here you can restrict which users to\
ring on incoming calls. If the list is empty, the system will indicate that all users \
enabled for incoming calls will ring. Invalid usernames will be rejected \
silently. Also, entering a username here overrides the user's setting to not receive \
incoming calls. This way, you can make certain users ring only for specific providers. \
Entries can be made in a space-separated list, and/or one per line by hitting enter after \
every one.")
end
s = m:section(NamedSection, "incoming_calls", "call_routing", translate("Incoming Calls"), text)
s.anonymous = true
for k,v in pairs(validinaccounts) do
users = s:option(DynamicList, k, v)
-- If the saved field is empty, we return a string
-- telling the user that this account would dial any exten.
function users.cfgvalue(self, section)
value = self.map:get(section, self.option)
if value == nil then
return {translate("Rings users enabled for incoming calls")}
else
return value
end
end
-- Write only valid user names.
function users.write(self, section, value)
newvalue = {}
nindex = 1
for index, field in ipairs(value) do
trimuser = luci.util.trim(value[index])
if allvalidusers[trimuser] == true then
newvalue[nindex] = trimuser
nindex = nindex + 1
end
end
DynamicList.write(self, section, newvalue)
end
end
----------------------------------------------------------------------------------------------------
-- If there are no user accounts configured, no user accounts enabled for outgoing calls,
-- display a warning. Otherwise, display the usual help text within the section.
if nallvalidusers == 0 then
text = translate("NOTE: There are no local user accounts configured.")
elseif nvalidoutusers == 0 then
text = translate("NOTE: There are no local user accounts enabled for outgoing calls.")
else
text = translate("For each user enabled for outgoing calls you can restrict what providers the user \
can use for outgoing calls. By default all users can use all providers. To show up in the list \
below the user should be allowed to make outgoing calls in the \"User Accounts\" page. Enter VoIP \
providers in the format username@some.host.name, as listed in \"Outgoing Calls\" above. It's \
easiest to copy and paste the providers from above. Invalid entries, including providers not \
enabled for outgoing calls, will be rejected silently. Entries can be made in a space-separated \
list, and/or one per line by hitting enter after every one.")
end
s = m:section(NamedSection, "providers_user_can_use", "call_routing",
translate("Providers Used for Outgoing Calls"), text)
s.anonymous = true
for k,v in pairs(validoutusers) do
providers = s:option(DynamicList, k, k)
-- If the saved field is empty, we return a string
-- telling the user that this account would dial any exten.
function providers.cfgvalue(self, section)
value = self.map:get(section, self.option)
if value == nil then
return {translate("Uses providers enabled for outgoing calls")}
else
newvalue = {}
-- Convert internal names to user@host values.
for i,v in ipairs(value) do
newvalue[i] = validoutaccounts[v]
end
return newvalue
end
end
-- Cook the new values prior to entering them into the config file.
-- Also, enter them only if they are valid.
function providers.write(self, section, value)
cookedvalue = {}
cindex = 1
for index, field in ipairs(value) do
cooked = string.gsub(luci.util.trim(value[index]), "%W", "_")
if validoutaccounts[cooked] ~= nil then
cookedvalue[cindex] = cooked
cindex = cindex + 1
end
end
DynamicList.write(self, section, cookedvalue)
end
end
----------------------------------------------------------------------------------------------------
s = m:section(TypedSection, "callthrough_numbers", translate("Call-through Numbers"),
translate("Designate numbers that are allowed to call through this system and which user's \
privileges 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 out with this user's providers. \
Invalid usernames, including users not enabled for outgoing calls, are dropped silently. \
Please verify that the entry was accepted."))
function user.write(self, section, value)
trimuser = luci.util.trim(value)
if allvalidusers[trimuser] == true then
Value.write(self, section, trimuser)
end
end
pwd = s:option(Value, "pin", translate("PIN"),
translate("Your PIN disappears when saved for your protection. It will be changed \
only when you enter a value different from the saved one. Leaving the PIN \
empty is possible, but please beware of the security implications."))
pwd.password = true
pwd.rmempty = false
-- We skip reading off the saved value and return nothing.
function pwd.cfgvalue(self, section)
return ""
end
-- We check the entered value against the saved one, and only write if the entered value is
-- something other than the empty string, and it differes from the saved value.
function pwd.write(self, section, value)
local orig_pwd = m:get(section, self.option)
if value and #value > 0 and orig_pwd ~= value then
Value.write(self, section, value)
end
end
----------------------------------------------------------------------------------------------------
s = m:section(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). Please \
make at least one voice call using the Google Talk plugin installable through the \
GMail interface, and then log out from your account everywhere. Click \"Add\" \
to add as many accounts as you wish."))
-- Recreate the config, and restart services after changes are commited to the configuration.
function m.on_after_commit(self)
-- Create a field "name" for each account that identifies the account in the backend.
commit = false
m.uci:foreach(modulename, "gtalk_jabber",
function(s1)
if s1.username ~= nil then
name=string.gsub(s1.username, "%W", "_")
if s1.name ~= name then
m.uci:set(modulename, s1['.name'], "name", name)
commit = true
end
end
end)
if commit == true then m.uci:commit(modulename) end
luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null")
luci.sys.call("/etc/init.d/asterisk restart 1\>/dev/null 2\>/dev/null")
end
-----------------------------------------------------------------------------
s = m:section(TypedSection, "gtalk_jabber", translate("Google Voice/Talk Accounts"))
s.anonymous = true
s.addremove = true
s:option(Value, "username", translate("Email"))
pwd = s:option(Value, "secret", translate("Password"),
translate("When your password is saved, it disappears from this field and is not displayed \
for your protection. The previously saved password will be changed only when you \
enter a value different from the saved one."))
pwd.password = true
pwd.rmempty = false
-- We skip reading off the saved value and return nothing.
function pwd.cfgvalue(self, section)
return ""
end
-- We check the entered value against the saved one, and only write if the entered value is
-- something other than the empty string, and it differes from the saved value.
function pwd.write(self, section, value)
local orig_pwd = m:get(section, self.option)
if value and #value > 0 and orig_pwd ~= value then
Value.write(self, section, value)
end
end
p = s:option(ListValue, "register",
translate("Enable Incoming Calls (set Status below)"),
translate("When somebody starts voice chat with your GTalk account or calls the GVoice, \
number (if you have Google Voice), the call will be forwarded to any users \
that are online (registered using a SIP device or softphone) and permitted to \
receive the call. If you have Google Voice, you must go to your GVoice settings and \
forward calls to Google chat in order to actually receive calls made to your \
GVoice number. If you have trouble receiving calls from GVoice, experiment \
with the Call Screening option in your GVoice Settings. Finally, make sure no other \
client is online with this account (browser in gmail, mobile/desktop Google Talk \
App) as it may interfere."))
p:value("yes", translate("Yes"))
p:value("no", translate("No"))
p.default = "yes"
p = s:option(ListValue, "make_outgoing_calls", translate("Enable Outgoing Calls"),
translate("Use this account to make outgoing calls as configured in the \"Call Routing\" section."))
p:value("yes", translate("Yes"))
p:value("no", translate("No"))
p.default = "yes"
st = s:option(ListValue, "status", translate("Google Talk Status"))
st:depends("register", "yes")
st:value("dnd", translate("Do Not Disturb"))
st:value("away", translate("Away"))
st:value("available", translate("Available"))
st.default = defaultstatus
stm = s:option(Value, "statusmessage", translate("Google Talk Status Message"),
translate("Avoid using anything but alpha-numeric characters, space, comma, and period."))
stm:depends("register", "yes")
stm.default = defaultstatusmessage
return m

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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, that you \
will use to register with this service. Use this account either in an Analog Telephony \
Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid on your \
smartphone, or Ekiga, Linphone, or X-Lite on your computer. By default, all SIP accounts \
will ring simultaneously if a call is made to one of your VoIP provider accounts or GV \
numbers."))
-- Recreate the config, and restart services after changes are commited to the configuration.
function m.on_after_commit(self)
luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null")
luci.sys.call("/etc/init.d/" .. server .. " restart 1\>/dev/null 2\>/dev/null")
end
externhost = m.uci:get(modulenameadvanced, "advanced", "externhost")
bindport = m.uci:get(modulenameadvanced, "advanced", "bindport")
ipaddr = m.uci:get("network", "lan", "ipaddr")
-----------------------------------------------------------------------------
s = m:section(NamedSection, "server", "user", translate("Server Setting"))
s.anonymous = true
if ipaddr == nil or ipaddr == "" then
ipaddr = "(IP address not static)"
end
if bindport ~= nil then
just_ipaddr = ipaddr
ipaddr = ipaddr .. ":" .. bindport
end
s:option(DummyValue, "ipaddr", translate("Server Setting for Local SIP Devices"),
translate("Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices you will \
use ONLY locally and never from a remote location.")).default = ipaddr
if externhost ~= nil then
if bindport ~= nil then
just_externhost = externhost
externhost = externhost .. ":" .. bindport
end
s:option(DummyValue, "externhost", translate("Server Setting for Remote SIP Devices"),
translate("Enter this hostname (or hostname:port) in the Server/Registrar setting of SIP \
devices you will use from a remote location (they will work locally too).")
).default = externhost
end
if bindport ~= nil then
s:option(DummyValue, "bindport", translate("Port Setting for SIP Devices"),
translatef("If setting Server/Registrar to %s or %s does not work for you, try setting \
it to %s or %s and entering this port number in a separate field that specifies the \
Server/Registrar port number. Beware that some devices have a confusing \
setting that sets the port where SIP requests originate from on the SIP \
device itself (the bind port). The port specified on this page is NOT this bind port \
but the port this service listens on.",
ipaddr, externhost, just_ipaddr, just_externhost)).default = bindport
end
-----------------------------------------------------------------------------
s = m:section(TypedSection, "local_user", translate("SIP Device/Softphone Accounts"))
s.anonymous = true
s.addremove = true
s:option(Value, "fullname", translate("Full Name"),
translate("You can specify a real name to show up in the Caller ID here."))
du = s:option(Value, "defaultuser", translate("User Name"),
translate("Use (four to five digit) numeric user name if you are connecting normal telephones \
with ATAs to this system (so they can dial user names)."))
du.datatype = "uciname"
pwd = s:option(Value, "secret", translate("Password"),
translate("Your password disappears when saved for your protection. It will be changed \
only when you enter a value different from the saved one."))
pwd.password = true
pwd.rmempty = false
-- We skip reading off the saved value and return nothing.
function pwd.cfgvalue(self, section)
return ""
end
-- We check the entered value against the saved one, and only write if the entered value is
-- something other than the empty string, and it differes from the saved value.
function pwd.write(self, section, value)
local orig_pwd = m:get(section, self.option)
if value and #value > 0 and orig_pwd ~= value then
Value.write(self, section, value)
end
end
p = s:option(ListValue, "ring", translate("Receives Incoming Calls"))
p:value("yes", translate("Yes"))
p:value("no", translate("No"))
p.default = "yes"
p = s:option(ListValue, "can_call", translate("Makes Outgoing Calls"))
p:value("yes", translate("Yes"))
p:value("no", translate("No"))
p.default = "yes"
return m

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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 that identifies the account in the backend.
m.uci:foreach(modulename, "voip_provider",
function(s1)
if s1.defaultuser ~= nil and s1.host ~= nil then
name=string.gsub(s1.defaultuser.."_"..s1.host, "%W", "_")
if s1.name ~= name then
m.uci:set(modulename, s1['.name'], "name", name)
commit = true
end
end
end)
if commit == true then m.uci:commit(modulename) end
luci.sys.call("/etc/init.d/pbx-" .. server .. " restart 1\>/dev/null 2\>/dev/null")
luci.sys.call("/etc/init.d/" .. server .. " restart 1\>/dev/null 2\>/dev/null")
end
-----------------------------------------------------------------------------
s = m:section(TypedSection, "voip_provider", translate("SIP Provider Accounts"))
s.anonymous = true
s.addremove = true
s:option(Value, "defaultuser", translate("User Name"))
pwd = s:option(Value, "secret", translate("Password"),
translate("When your password is saved, it disappears from this field and is not displayed \
for your protection. The previously saved password will be changed only when you \
enter a value different from the saved one."))
pwd.password = true
pwd.rmempty = false
-- We skip reading off the saved value and return nothing.
function pwd.cfgvalue(self, section)
return ""
end
-- We check the entered value against the saved one, and only write if the entered value is
-- something other than the empty string, and it differes from the saved value.
function pwd.write(self, section, value)
local orig_pwd = m:get(section, self.option)
if value and #value > 0 and orig_pwd ~= value then
Value.write(self, section, value)
end
end
h = s:option(Value, "host", translate("SIP Server/Registrar"))
h.datatype = "host"
p = s:option(ListValue, "register", translate("Enable Incoming Calls (Register via SIP)"),
translate("This option should be set to \"Yes\" if you have a DID \(real telephone number\) \
associated with this SIP account or want to receive SIP uri calls through this \
provider."))
p:value("yes", translate("Yes"))
p:value("no", translate("No"))
p.default = "yes"
p = s:option(ListValue, "make_outgoing_calls", translate("Enable Outgoing Calls"),
translate("Use this account to make outgoing calls."))
p:value("yes", translate("Yes"))
p:value("no", translate("No"))
p.default = "yes"
from = s:option(Value, "fromdomain",
translate("SIP Realm (needed by some providers)"))
from.optional = true
from.datatype = "host"
port = s:option(Value, "port", translate("SIP Server/Registrar Port"))
port.optional = true
port.datatype = "port"
op = s:option(Value, "outboundproxy", translate("Outbound Proxy"))
op.optional = true
op.datatype = "host"
return m

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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/>.
]]--
modulename = "pbx"
if nixio.fs.access("/etc/init.d/asterisk") then
server = "asterisk"
elseif nixio.fs.access("/etc/init.d/freeswitch") then
server = "freeswitch"
else
server = ""
end
-- Returns formatted output of string containing only the words at the indices
-- specified in the table "indices".
function format_indices(string, indices)
if indices == nil then
return "Error: No indices to format specified.\n"
end
-- Split input into separate lines.
lines = luci.util.split(luci.util.trim(string), "\n")
-- Split lines into separate words.
splitlines = {}
for lpos,line in ipairs(lines) do
splitlines[lpos] = luci.util.split(luci.util.trim(line), "%s+", nil, true)
end
-- For each split line, if the word at all indices specified
-- to be formatted are not null, add the formatted line to the
-- gathered output.
output = ""
for lpos,splitline in ipairs(splitlines) do
loutput = ""
for ipos,index in ipairs(indices) do
if splitline[index] ~= nil then
loutput = loutput .. string.format("%-40s", splitline[index])
else
loutput = nil
break
end
end
if loutput ~= nil then
output = output .. loutput .. "\n"
end
end
return output
end
m = Map (modulename, translate("PBX Main Page"),
translate("This configuration page allows you to configure a phone system (PBX) service which \
permits making phone calls through multiple Google and SIP (like Sipgate, \
SipSorcery, and Betamax) accounts and sharing them among many SIP devices. \
Note that Google accounts, SIP accounts, and local user accounts are configured in the \
\"Google Accounts\", \"SIP Accounts\", and \"User Accounts\" sub-sections. \
You must add at least one User Account to this PBX, and then configure a SIP device or \
softphone to use the account, in order to make and receive calls with your Google/SIP \
accounts. Configuring multiple users will allow you to make free calls between all users, \
and share the configured Google and SIP accounts. If you have more than one Google and SIP \
accounts set up, you should probably configure how calls to and from them are routed in \
the \"Call Routing\" page. If you're interested in using your own PBX from anywhere in the \
world, then visit the \"Remote Usage\" section in the \"Advanced Settings\" page."))
-----------------------------------------------------------------------------------------
s = m:section(NamedSection, "connection_status", "main",
translate("PBX Service Status"))
s.anonymous = true
s:option (DummyValue, "status", translate("Service Status"))
sts = s:option(DummyValue, "_sts")
sts.template = "cbi/tvalue"
sts.rows = 20
function sts.cfgvalue(self, section)
if server == "asterisk" then
regs = luci.sys.exec("asterisk -rx 'sip show registry' | sed 's/peer-//'")
jabs = luci.sys.exec("asterisk -rx 'jabber show connections' | grep onnected")
usrs = luci.sys.exec("asterisk -rx 'sip show users'")
chan = luci.sys.exec("asterisk -rx 'core show channels'")
return format_indices(regs, {1, 5}) ..
format_indices(jabs, {2, 4}) .. "\n" ..
format_indices(usrs, {1} ) .. "\n" .. chan
elseif server == "freeswitch" then
return "Freeswitch is not supported yet.\n"
else
return "Neither Asterisk nor FreeSwitch discovered, please install Asterisk, as Freeswitch is not supported yet.\n"
end
end
return m

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config 'main' 'connection_status'

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config 'settings' 'advanced'
option 'useragent' 'PBX'
option 'ringtime' '30'
option 'rtpstart' '19850'
option 'rtpend' '19900'

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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'

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

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applications/luci-pbx/root/etc/config/pbx-voip Normal file
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applications/luci-pbx/root/etc/init.d/pbx-asterisk Executable 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"
HANGUPCNTXT=hangup-call-context
GTALKUNVL=unavailable
ASTUSER=nobody
ASTGROUP=nogroup
ASTDIRSRECURSIVE="/var/run/asterisk /var/log/asterisk /var/spool/asterisk"
ASTDIRS="/usr/lib/asterisk"
ASTSOUNDSDIR="/usr/lib/asterisk/sounds"
TEMPLATEDIR=/etc/${MODULENAME}-asterisk
PBXSOUNDSDIR=$TEMPLATEDIR/sounds
VMTEMPLATEDIR=/etc/${MODULENAME}-voicemail
VMSOUNDSDIR=$VMTEMPLATEDIR/sounds
ASTERISKDIR=/etc/asterisk
WORKDIR=/tmp/$MODULENAME.$$
MD5SUMSFILE=/tmp/$MODULENAME-sums.$$
TMPL_ASTERISK=$TEMPLATEDIR/asterisk.conf.TEMPLATE
TMPL_GTALK=$TEMPLATEDIR/gtalk.conf.TEMPLATE
TMPL_INDICATIONS=$TEMPLATEDIR/indications.conf.TEMPLATE
TMPL_LOGGER=$TEMPLATEDIR/logger.conf.TEMPLATE
TMPL_MANAGER=$TEMPLATEDIR/manager.conf.TEMPLATE
TMPL_MODULES=$TEMPLATEDIR/modules.conf.TEMPLATE
TMPL_RTP=$TEMPLATEDIR/rtp.conf.TEMPLATE
TMPL_EXTCTHRUCHECKHDR=$TEMPLATEDIR/extensions_disa-check_header.conf.TEMPLATE
TMPL_EXTCTHRUCHECK=$TEMPLATEDIR/extensions_disa-check.conf.TEMPLATE
TMPL_EXTCTHRUCHECKFTR=$TEMPLATEDIR/extensions_disa-check_footer.conf.TEMPLATE
TMPL_EXTCTHRUHDR=$TEMPLATEDIR/extensions_disa_header.conf.TEMPLATE
TMPL_EXTCTHRU=$TEMPLATEDIR/extensions_disa.conf.TEMPLATE
TMPL_EXTENSIONS=$TEMPLATEDIR/extensions.conf.TEMPLATE
TMPL_EXTVMDISABLED=$TEMPLATEDIR/extensions_voicemail_disabled.conf.TEMPLATE
TMPL_EXTVMENABLED=$TEMPLATEDIR/extensions_voicemail_enabled.conf.TEMPLATE
TMPL_EXTBLKLIST=$TEMPLATEDIR/extensions_blacklist.conf.TEMPLATE
TMPL_EXTBLKLISTFTR=$TEMPLATEDIR/extensions_blacklist_footer.conf.TEMPLATE
TMPL_EXTBLKLISTHDR=$TEMPLATEDIR/extensions_blacklist_header.conf.TEMPLATE
TMPL_EXTDEFAULT=$TEMPLATEDIR/extensions_default.conf.TEMPLATE
TMPL_EXTDEFAULTUSER=$TEMPLATEDIR/extensions_default_user.conf.TEMPLATE
TMPL_EXTINCNTXTSIP=$TEMPLATEDIR/extensions_incoming_context_sip.conf.TEMPLATE
TMPL_EXTINCNTXTGTALKHDR=$TEMPLATEDIR/extensions_incoming_context_gtalk_header.conf.TEMPLATE
TMPL_EXTINCNTXTGTALK=$TEMPLATEDIR/extensions_incoming_context_gtalk.conf.TEMPLATE
TMPL_EXTUSERCNTXT=$TEMPLATEDIR/extensions_user_context.conf.TEMPLATE
TMPL_EXTUSERCNTXTFTR=$TEMPLATEDIR/extensions_user_context_footer.conf.TEMPLATE
TMPL_EXTUSERCNTXTHDR=$TEMPLATEDIR/extensions_user_context_header.conf.TEMPLATE
TMPL_EXTOUTHDR=$TEMPLATEDIR/extensions_default_outgoing_header.conf.TEMPLATE
TMPL_EXTOUTGTALK=$TEMPLATEDIR/extensions_outgoing_gtalk.conf.TEMPLATE
TMPL_EXTOUTLOCAL=$TEMPLATEDIR/extensions_outgoing_dial_local_user.conf.TEMPLATE
TMPL_EXTOUTSIP=$TEMPLATEDIR/extensions_outgoing_sip.conf.TEMPLATE
TMPL_JABBER=$TEMPLATEDIR/jabber.conf.TEMPLATE
TMPL_JABBERUSER=$TEMPLATEDIR/jabber_users.conf.TEMPLATE
TMPL_SIP=$TEMPLATEDIR/sip.conf.TEMPLATE
TMPL_SIPPEER=$TEMPLATEDIR/sip_peer.TEMPLATE
TMPL_SIPREG=$TEMPLATEDIR/sip_registration.TEMPLATE
TMPL_SIPUSR=$TEMPLATEDIR/sip_user.TEMPLATE
TMPL_MSMTPDEFAULT=$VMTEMPLATEDIR/pbx-msmtprc-defaults.TEMPLATE
TMPL_MSMTPACCOUNT=$VMTEMPLATEDIR/pbx-msmtprc-account.TEMPLATE
TMPL_MSMTPAUTH=$VMTEMPLATEDIR/pbx-msmtprc-account-auth.TEMPLATE
TMPL_MSMTPACCTDFLT=$VMTEMPLATEDIR/pbx-msmtprc-account-default.TEMPLATE
INCLUDED_FILES="$WORKDIR/extensions_blacklist.conf $WORKDIR/extensions_callthrough.conf\
$WORKDIR/extensions_incoming.conf $WORKDIR/extensions_incoming_gtalk.conf\
$WORKDIR/extensions_user.conf $WORKDIR/jabber_users.conf\
$WORKDIR/sip_peers.conf $WORKDIR/sip_registrations.conf\
$WORKDIR/sip_users.conf $WORKDIR/extensions_voicemail.conf\
$WORKDIR/extensions_default.conf"
# In this string, we concatenate all local users enabled to receive calls
# readily formatted for the Dial command.
localusers_to_ring=""
# In this string, we keep a list of all users that are enabled for outgoing
# calls. It is used at the end to create the user contexts.
localusers_can_dial=""
# In this string, we put together a space-separated list of provider names
# (alphanumeric, with all non-alpha characters replaced with underscores),
# which will be used to dial out by default (whose outgoing contexts will
# be included in users' contexts by default.
outbound_providers=""
# Copies the template files which we don't edit.
copy_unedited_templates_over()
{
cp $TMPL_ASTERISK $WORKDIR/asterisk.conf
cp $TMPL_GTALK $WORKDIR/gtalk.conf
cp $TMPL_INDICATIONS $WORKDIR/indications.conf
cp $TMPL_LOGGER $WORKDIR/logger.conf
cp $TMPL_MANAGER $WORKDIR/manager.conf
cp $TMPL_MODULES $WORKDIR/modules.conf
# If this file isn't present at this stage, voicemail is disabled.
[ ! -f $WORKDIR/extensions_voicemail.conf ] && \
cp $TMPL_EXTVMDISABLED $WORKDIR/extensions_voicemail.conf
}
# Touches all the included files, to prevent asterisk from refusing to
# start if a config item is missing and an included config file isn't created.
create_included_files()
{
touch $INCLUDED_FILES
}
# Puts together all the extensions.conf related configuration.
pbx_create_extensions_config()
{
local ringtime
config_get ringtime advanced ringtime
sed "s/|RINGTIME|/$ringtime/" $TMPL_EXTENSIONS > $WORKDIR/extensions.conf
mv $WORKDIR/inext.TMP $WORKDIR/extensions_incoming.conf
cp $TMPL_EXTINCNTXTGTALKHDR $WORKDIR/extensions_incoming_gtalk.conf
cat $WORKDIR/outextgtalk.TMP >> $WORKDIR/extensions_incoming_gtalk.conf 2>/dev/null
rm -f $WORKDIR/outextgtalk.TMP
mv $WORKDIR/blacklist.TMP $WORKDIR/extensions_blacklist.conf
mv $WORKDIR/userext.TMP $WORKDIR/extensions_user.conf
cp $TMPL_EXTCTHRUHDR $WORKDIR/extensions_callthrough.conf
cat $WORKDIR/callthrough.TMP >> $WORKDIR/extensions_callthrough.conf 2>/dev/null
rm -f $WORKDIR/callthrough.TMP
cat $TMPL_EXTCTHRUCHECKHDR >> $WORKDIR/extensions_callthrough.conf 2>/dev/null
cat $WORKDIR/callthroughcheck.TMP >> $WORKDIR/extensions_callthrough.conf 2>/dev/null
rm -f $WORKDIR/callthroughcheck.TMP
cat $TMPL_EXTCTHRUCHECKFTR >> $WORKDIR/extensions_callthrough.conf 2>/dev/null
rm -f $WORKDIR/outext-*.TMP
rm -f $WORKDIR/localext.TMP
sed "s/|LOCALUSERS|/$localusers_to_ring/g" $TMPL_EXTDEFAULT \
> $WORKDIR/extensions_default.conf
cat $WORKDIR/inextuser.TMP >> $WORKDIR/extensions_default.conf
rm -f $WORKDIR/inextuser.TMP
}
# Puts together all the sip.conf related configuration.
pbx_create_sip_config()
{
mv $WORKDIR/sip_regs.TMP $WORKDIR/sip_registrations.conf
mv $WORKDIR/sip_peers.TMP $WORKDIR/sip_peers.conf
mv $WORKDIR/sip_users.TMP $WORKDIR/sip_users.conf
}
# Creates the jabber.conf related config
pbx_create_jabber_config()
{
cp $TMPL_JABBER $WORKDIR/jabber.conf
mv $WORKDIR/jabber.TMP $WORKDIR/jabber_users.conf
}
# Gets rid of any config files from $ASTERISKDIR not found in $WORKDIR.
clean_up_asterisk_config_dir()
{
for f in $ASTERISKDIR/* ; do
basef="`basename $f`"
if [ ! -e "$WORKDIR/$basef" ] ; then
rm -rf "$f"
fi
done
}
# Compares md5sums of the config files in $WORKDIR to those
# in $ASTERISKDIR, and copies only changed files over to reduce
# wear on flash in embedded devices.
compare_configs_and_copy_changed()
{
# First, compute md5sums of the config files in $WORKDIR.
cd $WORKDIR/
md5sum * > $MD5SUMSFILE
# Now, check the files in $ASTERISKDIR against the md5sums.
cd $ASTERISKDIR/
changed_files="`md5sum -c $MD5SUMSFILE 2>/dev/null | fgrep ": FAILED" | awk -F: '{print $1}'`"
rm -f $MD5SUMSFILE
[ -z "$changed_files" ] && return
# Now copy over the changed files.
for f in $changed_files ; do
cp "$WORKDIR/$f" "$ASTERISKDIR/$f"
done
}
# Calls the functions that create the final config files
# Calls the function which compares which files have changed
# Puts the final touches on $ASTERISKDIR
# Gets rid of $WORKDIR
pbx_assemble_and_copy_config()
{
mkdir -p $ASTERISKDIR
copy_unedited_templates_over
create_included_files
pbx_create_extensions_config
pbx_create_sip_config
pbx_create_jabber_config
touch $WORKDIR/features.conf
# At this point, $WORKDIR should contain a complete, working config.
clean_up_asterisk_config_dir
compare_configs_and_copy_changed
[ ! -d $ASTERISKDIR/manager.d ] && mkdir -p $ASTERISKDIR/manager.d/
# Get rid of the working directory
rm -rf $WORKDIR/
}
# Creates configuration for a user and adds it to the temporary file that holds
# all users configured so far.
pbx_add_user()
{
local fullname
local defaultuser
local 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 >> $WORKDIR/localext.TMP
# Add configuration which puts calls to users through the default
# context, so that blacklists and voicemail take effect for this user.
sed "s/|DEFAULTUSER|/$defaultuser/g" $TMPL_EXTDEFAULTUSER >> $WORKDIR/inextuser.TMP
rm -f $WORKDIR/sip_user.tmp
}
# Creates configuration for a Google account, and adds it to the temporary file that holds
# all accounts configured so far.
# Also creates the outgoing extensions which are used in users' outgoing contexts.
pbx_add_jabber()
{
local username
local secret
local numprefix
local register
local make_outgoing_calls
local name
local users_to_ring
local status
local statusmessage
config_get username $1 username
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
}
# Links any sound files found in $PBXSOUNDSDIR and $VMSOUNDSDIR
# into $ASTSOUNDSDIR for use by Asterisk. Does not overwrite files.
pbx_link_sounds()
{
mkdir -p $ASTSOUNDSDIR
for dir in $PBXSOUNDSDIR $VMSOUNDSDIR ; do
if [ -d $dir ] ; then
for f in $dir/* ; do
ln -s $f $ASTSOUNDSDIR 2>/dev/null
done
fi
done
}
# Makes sure the ownership of specified directories is proper.
pbx_fix_ownership()
{
chown $ASTUSER:$ASTGROUP $ASTDIRS
chown $ASTUSER:$ASTGROUP -R $ASTDIRSRECURSIVE
}
# Creates voicemail config if installed and enabled.
pbx_configure_voicemail()
{
local enabled
local global_timeout
local global_email_addresses
local smtp_tls
local smtp_server
local smtp_port
local smtp_auth
local smtp_user
local smtp_password
config_get enabled global_voicemail enabled
# First check if voicemail is enabled.
[ "$enabled" != "yes" ] && return
config_get global_timeout global_voicemail global_timeout
#config_get global_email_addresses global_voicemail global_email_addresses
config_get smtp_auth voicemail_smtp smtp_auth
config_get smtp_tls voicemail_smtp smtp_tls
config_get smtp_server voicemail_smtp smtp_server
config_get smtp_port voicemail_smtp smtp_port
config_get smtp_user voicemail_smtp smtp_user
config_get smtp_password voicemail_smtp smtp_password
sed "s/|AUTH|/$smtp_auth/" $TMPL_MSMTPDEFAULT |\
sed "s/|TLS|/$smtp_tls/" > $WORKDIR/pbx-msmtprc
sed "s/|HOST|/$smtp_server/" $TMPL_MSMTPACCOUNT |\
sed "s/|PORT|/$smtp_port/" >> $WORKDIR/pbx-msmtprc
if [ "$smtp_auth" = "on" ] ; then
sed "s/|USER|/$smtp_user/" $TMPL_MSMTPAUTH |\
sed "s/|PASSWORD|/$smtp_password/" >> $WORKDIR/pbx-msmtprc
fi
cat $TMPL_MSMTPACCTDFLT >> $WORKDIR/pbx-msmtprc
[ ! -f /etc/pbx-msmtprc ] && cp $WORKDIR/pbx-msmtprc /etc/pbx-msmtprc
diff -q $WORKDIR/pbx-msmtprc /etc/pbx-msmtprc 1>/dev/null \
|| mv $WORKDIR/pbx-msmtprc /etc/pbx-msmtprc
chmod 600 /etc/pbx-msmtprc
# Copy over the extensions file which has voicemail enabled.
cp $TMPL_EXTVMENABLED $WORKDIR/extensions_voicemail.conf
}
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
# Prepare voicemail config.
config_unset
config_load ${MODULENAME}-voicemail
pbx_configure_voicemail
# Assemble the configuration, and copy changed files over.
config_unset
config_load ${MODULENAME}-advanced
pbx_assemble_and_copy_config
# Link sound files
pbx_link_sounds
# Enforce ownership of specified files and directories.
pbx_fix_ownership
}

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applications/luci-pbx/root/etc/pbx-asterisk/asterisk.conf.TEMPLATE Normal file
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[directories]
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /usr/lib/asterisk
astdbdir => /usr/lib/asterisk
astkeydir => /usr/lib/asterisk
astdatadir => /usr/lib/asterisk
astagidir => /usr/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk
[options]
languageprefix = yes
dumpcore = no
runuser = nobody
rungroup = nogroup

24
applications/luci-pbx/root/etc/pbx-asterisk/extensions.conf.TEMPLATE Normal file
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[general]
static = yes
writeprotect = yes
clearglobalvars = no
[globals]
RINGTIME => |RINGTIME|
[default]
[context-user-hangup-call-context]
exten => s,1,Hangup()
exten => _X.,1,Hangup()
[context-catch-all]
exten => _[!-~].,1,Dial(SIP/${EXTEN},60,r)
#include extensions_default.conf
#include extensions_voicemail.conf
#include extensions_incoming.conf
#include extensions_incoming_gtalk.conf
#include extensions_blacklist.conf
#include extensions_callthrough.conf
#include extensions_user.conf

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

9
applications/luci-pbx/root/etc/pbx-asterisk/extensions_default.conf.TEMPLATE Normal file
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[default-incoming-call-context]
exten => s,1,NoOp(${CALLERID})
exten => s,n,Set(SOURCECONTEXT=default-incoming-call-context)
exten => s,n,Set(SOURCEEXTEN=s)
exten => s,n,Goto(blacklist-call-context,s,1)
exten => s,n(doneblacklist),NoOp()
exten => s,n,Goto(disa-check-call-context,s,1)
exten => s,n(donedisacheck),Dial(|LOCALUSERS|,${RINGTIME},r)
exten => s,n,Goto(context-voicemail,s,1)

1
applications/luci-pbx/root/etc/pbx-asterisk/extensions_default_user.conf.TEMPLATE Normal file
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exten => |DEFAULTUSER|,1,Goto(default-incoming-call-context,s,1)

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()

6
applications/luci-pbx/root/etc/pbx-asterisk/extensions_disa.conf.TEMPLATE Normal file
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exten => |NUMBER|,1,Noop()
exten => |NUMBER|,n,Set(TIMEOUT(digit)=7)
exten => |NUMBER|,n,Set(TIMEOUT(response)=21)
exten => |NUMBER|,n,Authenticate(|PIN|)
exten => |NUMBER|,n,DISA(no-password,context-user-|DEFAULTUSER|)

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

13
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|,${RINGTIME},D(:w11111111))
same => n,Goto(context-voicemail,s,1)

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

10
applications/luci-pbx/root/etc/pbx-asterisk/extensions_incoming_context_sip.conf.TEMPLATE Normal file
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[context-incoming-|NAME|]
exten => s,1,NoOp(${CALLERID})
exten => s,n,Set(SOURCECONTEXT=context-incoming-|NAME|)
exten => s,n,Set(SOURCEEXTEN=s)
exten => s,n,Goto(blacklist-call-context,s,1)
exten => s,n(doneblacklist),NoOp()
exten => s,n,Goto(disa-check-call-context,s,1)
exten => s,n(donedisacheck),Dial(|LOCALUSERS|,${RINGTIME},r)
exten => s,n,Goto(context-voicemail,s,1)

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|,${RINGTIME},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)

2
applications/luci-pbx/root/etc/pbx-asterisk/extensions_user_context_footer.conf.TEMPLATE Normal file
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include => context-voicemail-record-greeting
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|]

4
applications/luci-pbx/root/etc/pbx-asterisk/extensions_voicemail_disabled.conf.TEMPLATE Normal file
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[context-voicemail-record-greeting]
[context-voicemail]
exten => s,1,Hangup()

15
applications/luci-pbx/root/etc/pbx-asterisk/extensions_voicemail_enabled.conf.TEMPLATE Normal file
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[context-voicemail-record-greeting]
exten => *789,1,Wait(2)
exten => *789,n,Playback(/etc/pbx-voicemail/greeting)
exten => *789,n,Wait(5)
exten => *789,n,Record(/etc/pbx-voicemail/greeting:WAV,20,90,k)
exten => *789,n,Wait(2)
exten => *789,n,Playback(/etc/pbx-voicemail/greeting)
[context-voicemail]
exten => s,1,Wait(2)
exten => s,2,Playback(/etc/pbx-voicemail/greeting)
exten => s,3,Wait(2)
exten => s,n,Record(/tmp/voicemail%d:WAV,20,180,k)
exten => h,1,System(/etc/pbx-voicemail/pbx-send-voicemail '${RECORDED_FILE}.WAV' '${CALLERID(all)}' &)

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

733
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

34
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 => format_wav_gsm.so
load => app_dial.so ; Dialing Application - Requires res_features.so, res_musiconhold.so
load => app_parkandannounce.so ; Call Parking and Announce Application - Requires res_features.so
load => app_playback.so ; Sound File Playback Application - Requires N/A
load => app_record.so ; Sound File Record Application - Requires N/A
load => app_system.so ; Execute a system command - Requires N/A
load => app_disa.so ; Direct Inward System Access
load => app_authenticate.so ; Authenticate via pin
load => app_senddtmf.so ; Ability to send DTMF tones on the line.
load => func_cut.so ; To manipulate strings
load => func_timeout.so ; Used for DISA timeouts
[global]
chan_modem.so=no

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

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