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Remove code from checks to see if the extension is on the phone.

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Having the PBX check if the caller is already on the phone has not worked well at least not the approach that is getting removed in this commit. It is not the right way to solve the issue. Its better for the phone endpoint  to block the intercom or auto answer when it is already on a call.

One way to do this that worked in testing was to disable call waiting. Then the call is rejected and not allowed to interrupt. the call that already exists. Expect there are also other ways to instruct the phone not to interrupt active calls when it receives a SIP message to auto answer.
This commit is contained in:
FusionPBX
2022-05-27 14:45:41 -06:00
committed by GitHub
parent 10748a95c8
commit 5a89fa5081
1 changed files with 13 additions and 26 deletions
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39
app/scripts/resources/scripts/page.lua
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@@ -1,6 +1,6 @@
-- page.lua
-- Part of FusionPBX
-- Copyright (C) 2010 Mark J Crane <markjcrane@fusionpbx.com>
-- Copyright (C) 2010-2022 Mark J Crane <markjcrane@fusionpbx.com>
-- All rights reserved.
--
-- Redistribution and use in source and binary forms, with or without
@@ -152,10 +152,11 @@
end
end
--get the channels
--log the destinations
freeswitch.consoleLog("NOTICE", "[page] destinations "..destinations.." available\n");
--create the api object
api = freeswitch.API();
cmd_string = "show channels";
channel_result = api:executeString(cmd_string);
--originate the calls
destination_count = 0;
@@ -168,29 +169,15 @@
--prevent calling the user that initiated the page
if (sip_from_user ~= destination) then
--loop through channels to determine if 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';
end
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 {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;
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 {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;
end
end
end
end
end