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https://github.com/fusionpbx/fusionpbx.git
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Move the scripts to app/scripts/resources/scripts
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markjcrane
215
app/scripts/resources/scripts/page.lua
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215
app/scripts/resources/scripts/page.lua
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-- page.lua
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-- Part of FusionPBX
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-- Copyright (C) 2010 Mark J Crane <markjcrane@fusionpbx.com>
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-- All rights reserved.
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--
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-- Redistribution and use in source and binary forms, with or without
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-- modification, are permitted provided that the following conditions are met:
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--
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-- 1. Redistributions of source code must retain the above copyright notice,
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-- this list of conditions and the following disclaimer.
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--
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-- 2. Redistributions in binary form must reproduce the above copyright
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-- notice, this list of conditions and the following disclaimer in the
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-- documentation and/or other materials provided with the distribution.
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--
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-- THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES,
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-- INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY
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-- AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE
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-- AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY,
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-- OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
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-- SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
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-- INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
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-- CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
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-- ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
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-- POSSIBILITY OF SUCH DAMAGE.
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--set default settings
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pin_number = "";
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max_tries = "3";
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digit_timeout = "3000";
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--define the trim function
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require "resources.functions.trim";
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--define the explode function
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require "resources.functions.explode";
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--define the split function
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require "resources.functions.split";
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--iterator over numbers.
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local function each_number(value)
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local begin_value, end_value = split_first(value, "-", true)
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if (not end_value) or (begin_value == end_value) then
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return function()
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local result = begin_value
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begin_value = nil
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return result
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end
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end
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if string.find(begin_value, "^0") then
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assert(#begin_value == #end_value, "number in range with leading `0` should have same length")
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end
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local number_length = ("." .. tostring(#begin_value))
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begin_value, end_value = tonumber(begin_value), tonumber(end_value)
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assert(begin_value and end_value and (begin_value <= end_value), "Invalid range: " .. value)
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return function()
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value, begin_value = begin_value, begin_value + 1
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if value > end_value then return end
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return string.format("%" .. number_length .. "d", value)
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end
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end
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--make sure the session is ready
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if ( session:ready() ) then
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--answer the call
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session:answer();
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--get the dialplan variables and set them as local variables
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destination_number = session:getVariable("destination_number");
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pin_number = session:getVariable("pin_number");
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domain_name = session:getVariable("domain_name");
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sounds_dir = session:getVariable("sounds_dir");
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destinations = session:getVariable("destinations");
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rtp_secure_media = session:getVariable("rtp_secure_media");
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if (destinations == nil) then
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destinations = session:getVariable("extension_list");
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end
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destination_table = explode(",",destinations);
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caller_id_name = session:getVariable("caller_id_name");
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caller_id_number = session:getVariable("caller_id_number");
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sip_from_user = session:getVariable("sip_from_user");
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mute = session:getVariable("mute");
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--set the sounds path for the language, dialect and voice
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default_language = session:getVariable("default_language");
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default_dialect = session:getVariable("default_dialect");
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default_voice = session:getVariable("default_voice");
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if (not default_language) then default_language = 'en'; end
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if (not default_dialect) then default_dialect = 'us'; end
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if (not default_voice) then default_voice = 'callie'; end
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--set rtp_secure_media to an empty string if not provided.
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if (rtp_secure_media == nil) then
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rtp_secure_media = 'false';
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end
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--define the conference name
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local conference_profile = "page";
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local conference_name = "page-"..destination_number.."@"..domain_name;
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local conference_bridge = conference_name.."@"..conference_profile;
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--set the caller id
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if (caller_id_name) then
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--caller id name provided do nothing
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else
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effective_caller_id_name = session:getVariable("effective_caller_id_name");
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caller_id_name = effective_caller_id_name;
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end
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if (caller_id_number) then
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--caller id number provided do nothing
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else
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effective_caller_id_number = session:getVariable("effective_caller_id_number");
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caller_id_number = effective_caller_id_number;
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end
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--set conference flags
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if (mute == "true") then
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flags = "flags{mute}";
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else
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flags = "flags{}";
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end
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--if the pin number is provided then require it
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if (pin_number) then
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--sleep
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session:sleep(500);
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--get the user pin number
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min_digits = 2;
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max_digits = 20;
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digits = session:playAndGetDigits(min_digits, max_digits, max_tries, digit_timeout, "#", "phrase:voicemail_enter_pass:#", "", "\\d+");
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--validate the user pin number
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pin_number_table = explode(",",pin_number);
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for index,pin_number in pairs(pin_number_table) do
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if (digits == pin_number) then
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--set the variable to true
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auth = true;
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--set the authorized pin number that was used
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session:setVariable("pin_number", pin_number);
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--end the loop
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break;
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end
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end
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--if not authorized play a message and then hangup
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if (not auth) then
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session:streamFile("phrase:voicemail_fail_auth:#");
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session:hangup("NORMAL_CLEARING");
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return;
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end
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end
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--originate the calls
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destination_count = 0;
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api = freeswitch.API();
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for index,value in pairs(destination_table) do
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for destination in each_number(value) do
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--get the destination required for number-alias
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destination = api:execute("user_data", destination .. "@" .. domain_name .. " attr id");
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--prevent calling the user that initiated the page
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if (sip_from_user ~= destination) then
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--cmd = "username_exists id "..destination.."@"..domain_name;
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--reply = trim(api:executeString(cmd));
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--if (reply == "true") then
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destination_status = "show channels like "..destination.."@";
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reply = trim(api:executeString(destination_status));
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if (reply == "0 total.") then
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freeswitch.consoleLog("NOTICE", "[page] destination "..destination.." available\n");
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if destination == sip_from_user then
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--this destination is the caller that initated the page
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else
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--originate the call
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cmd_string = "bgapi originate {sip_auto_answer=true,sip_h_Alert-Info='Ring Answer',hangup_after_bridge=false,rtp_secure_media="..rtp_secure_media..",origination_caller_id_name='"..caller_id_name.."',origination_caller_id_number="..caller_id_number.."}user/"..destination.."@"..domain_name.." conference:"..conference_bridge.."+"..flags.." inline";
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api:executeString(cmd_string);
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destination_count = destination_count + 1;
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end
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--freeswitch.consoleLog("NOTICE", "cmd_string "..cmd_string.."\n");
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else
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--look inside the reply to check for the correct domain_name
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if string.find(reply, domain_name, nil, true) then
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--found: user is busy
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else
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--not found
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if (destination == tonumber(sip_from_user)) then
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--this destination is the caller that initated the page
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else
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--originate the call
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cmd_string = "bgapi originate {sip_auto_answer=true,hangup_after_bridge=false,rtp_secure_media="..rtp_secure_media..",origination_caller_id_name='"..caller_id_name.."',origination_caller_id_number="..caller_id_number.."}user/"..destination.."@"..domain_name.." conference:"..conference_bridge.."+"..flags.." inline";
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api:executeString(cmd_string);
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destination_count = destination_count + 1;
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end
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end
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end
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--end
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end
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end
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end
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--send main call to the conference room
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if (destination_count > 0) then
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if (session:getVariable("moderator") == "true") then
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moderator_flag = ",moderator";
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else
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moderator_flag = "";
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end
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session:execute("conference", conference_bridge.."+flags{endconf"..moderator_flag.."}");
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else
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session:execute("playback", "tone_stream://%(500,500,480,620);loops=3");
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end
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end
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