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Update page.lua with more options check_destination_status, auto answer and alert info.

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FusionPBX 2022-06-11 00:58:00 -06:00 committed by GitHub
parent d0d0346b42
commit af6b00bf66
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1 changed files with 68 additions and 7 deletions
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75
app/scripts/resources/scripts/page.lua
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@ -68,6 +68,7 @@
if ( session:ready() ) then
--answer the call
session:answer();
--get the dialplan variables and set them as local variables
destination_number = session:getVariable("destination_number");
pin_number = session:getVariable("pin_number");
@ -84,6 +85,31 @@
sip_from_user = session:getVariable("sip_from_user");
mute = session:getVariable("mute");
--determine whether to check if the destination is available
check_destination_status = session:getVariable("check_destination_status");
if (not check_destination_status) then check_destination_status = 'false'; end
--set the type of auto answer
auto_answer = session:getVariable("auto_answer");
if (not auto_answer) then auto_answer = 'call_info'; end
if (auto_answer == 'call_info') then
auto_answer = "sip_h_Call-Info=<sip:"..domain_name..">;answer-after=0";
end
if (auto_answer == 'sip_auto_answer') then
auto_answer = "sip_auto_answer=true";
end
--set sip header Alert-Info
alert_info = session:getVariable("alert_info");
if (not alert_info) then alert_info = 'ring_answer'; end
if (alert_info == 'auto_answer') then
alert_info = "sip_h_Alert-Info='Auto Answer'";
elseif (alert_info == 'ring_answer') then
alert_info = "sip_h_Alert-Info='Ring Answer'";
else
alert_info = "sip_h_Alert-Info='"..alert_info.."'";
end
--set the sounds path for the language, dialect and voice
default_language = session:getVariable("default_language");
default_dialect = session:getVariable("default_dialect");
@ -128,10 +154,12 @@
if (pin_number) then
--sleep
session:sleep(500);
--get the user pin number
min_digits = 2;
max_digits = 20;
digits = session:playAndGetDigits(min_digits, max_digits, max_tries, digit_timeout, "#", "phrase:voicemail_enter_pass:#", "", "\\d+");
--validate the user pin number
pin_number_table = explode(",",pin_number);
for index,pin_number in pairs(pin_number_table) do
@ -144,6 +172,7 @@
break;
end
end
--if not authorized play a message and then hangup
if (not auth) then
session:streamFile("phrase:voicemail_fail_auth:#");
@ -158,6 +187,12 @@
--create the api object
api = freeswitch.API();
--get the channels
if (check_destination_status == 'true') then
cmd_string = "show channels";
channel_result = api:executeString(cmd_string);
end
--originate the calls
destination_count = 0;
@ -169,14 +204,40 @@
--prevent calling the user that initiated the page
if (sip_from_user ~= destination) then
freeswitch.consoleLog("NOTICE", "[page] destination "..destination.." available\n");
if destination == sip_from_user then
--this destination is the caller that initated the page
if (check_destination_status == 'true') then
--detect if the destination is available or busy
destination_status = 'available';
channel_array = explode("\n", channel_result);
for index,row in pairs(channel_array) do
if string.find(row, destination..'@'..domain_name, nil, true) then
destination_status = 'busy';
break;
end
end
--if available then page then originate the call with auto answer
if (destination_status == 'available') then
freeswitch.consoleLog("NOTICE", "[page] destination "..destination.." available\n");
if destination == sip_from_user then
--this destination is the caller that initated the page
else
--originate the call
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";
api:executeString(cmd_string);
destination_count = destination_count + 1;
end
end
else
--originate the call
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";
api:executeString(cmd_string);
destination_count = destination_count + 1;
--endpoint determines what to do with the call when the destination is active
freeswitch.consoleLog("NOTICE", "[page] endpoint determines what to do if the it has an active call.\n");
if destination == sip_from_user then
--this destination is the caller that initated the page
else
--originate the call
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";
api:executeString(cmd_string);
destination_count = destination_count + 1;
end
end
end
end