mirror of
https://github.com/fusionpbx/fusionpbx.git
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ae98ad2d49
Adding "mintwo" flag will tell the system that there has to be a minimum of two people in the conference to keep it open. This fixes the problem of when a user initiates a *8[ext] to someone and the far end hangs up and the initiator does not hang up the line they will hear hold music till they hang up. Adding this will disconnect the intercom page if either party hangs up the call.
259 lines
9.6 KiB
Lua
259 lines
9.6 KiB
Lua
-- page.lua
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-- Part of FusionPBX
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-- Copyright (C) 2010-2022 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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--determine whether to check if the destination is available
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check_destination_status = session:getVariable("check_destination_status");
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if (not check_destination_status) then check_destination_status = 'false'; end
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--set the type of auto answer
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auto_answer = session:getVariable("auto_answer");
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if (not auto_answer) then auto_answer = 'call_info'; end
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if (auto_answer == 'call_info') then
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auto_answer = "sip_h_Call-Info=<sip:"..domain_name..">;answer-after=0";
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end
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if (auto_answer == 'sip_auto_answer') then
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auto_answer = "sip_auto_answer=true";
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end
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--set sip header Alert-Info
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alert_info = session:getVariable("alert_info");
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if (not alert_info) then alert_info = 'ring_answer'; end
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if (alert_info == 'auto_answer') then
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alert_info = "sip_h_Alert-Info='Auto Answer'";
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elseif (alert_info == 'ring_answer') then
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alert_info = "sip_h_Alert-Info='Ring Answer'";
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else
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alert_info = "sip_h_Alert-Info='"..alert_info.."'";
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end
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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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--log the destinations
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freeswitch.consoleLog("NOTICE", "[page] destinations "..destinations.." available\n");
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--create the api object
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api = freeswitch.API();
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--get the channels
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if (check_destination_status == 'true') then
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cmd_string = "show channels";
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channel_result = api:executeString(cmd_string);
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end
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--originate the calls
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destination_count = 0;
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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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if (check_destination_status == 'true') then
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--detect if the destination is available or busy
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destination_status = 'available';
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channel_array = explode("\n", channel_result);
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for index,row in pairs(channel_array) do
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if string.find(row, destination..'@'..domain_name, nil, true) then
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destination_status = 'busy';
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break;
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end
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end
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--if available then page then originate the call with auto answer
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if (destination_status == 'available') 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 {"..auto_answer..","..alert_info..",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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else
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--endpoint determines what to do with the call when the destination is active
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freeswitch.consoleLog("NOTICE", "[page] endpoint determines what to do if the it has an active call.\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 {"..auto_answer..","..alert_info..",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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--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,mintwo"..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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