dbxe
/
fusionpbx
2
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity

Improve page.lua add a way to control mute from a variable and if no one is called because they are on a call produce a busy tone for the caller.

This commit is contained in:
Mark Crane 2012-06-07 19:43:43 +00:00
parent 4d943f4697
commit a5e6b9f758
1 changed files with 22 additions and 3 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

25
includes/install/scripts/page.lua
View File

@ -54,6 +54,7 @@ if ( session:ready() ) then
caller_id_number = session:getVariable("caller_id_number");
extension_table = explode(",",extension_list);
sip_from_user = session:getVariable("sip_from_user");
mute = session:getVariable("mute");
--set the sounds path for the language, dialect and voice
default_language = session:getVariable("default_language");
@ -77,6 +78,17 @@ if ( session:ready() ) then
caller_id_number = effective_caller_id_number;
end
--set conference flags
if (mute) then
if (mute == "false") then
flags = "flags{}";
else
flags = "flags{mute}";
end
else
flags = "flags{mute}";
end
--if the pin number is provided then require it
if (pin_number) then
min_digits = string.len(pin_number);
@ -91,6 +103,7 @@ if ( session:ready() ) then
end
end
destination_count = 0;
api = freeswitch.API();
for index,value in pairs(extension_table) do
if (string.find(value, "-") == nill) then
@ -109,8 +122,9 @@ if ( session:ready() ) then
--this extension is the caller that initated the page
else
--originate the call
cmd_string = "bgapi originate {sip_auto_answer=true,hangup_after_bridge=false,origination_caller_id_name='"..caller_id_name.."',origination_caller_id_number="..caller_id_number.."}user/"..extension.."@"..domain_name.." conference:page-"..destination_number.."@page+flags{mute} inline";
cmd_string = "bgapi originate {sip_auto_answer=true,hangup_after_bridge=false,origination_caller_id_name='"..caller_id_name.."',origination_caller_id_number="..caller_id_number.."}user/"..extension.."@"..domain_name.." conference:page-"..destination_number.."@page+"..flags.." inline";
api:executeString(cmd_string);
destination_count = destination_count + 1;
end
--freeswitch.consoleLog("NOTICE", "cmd_string "..cmd_string.."\n");
else
@ -123,8 +137,9 @@ if ( session:ready() ) then
--this extension is the caller that initated the page
else
--originate the call
cmd_string = "bgapi originate {sip_auto_answer=true,hangup_after_bridge=false,origination_caller_id_name='"..caller_id_name.."',origination_caller_id_number="..caller_id_number.."}user/"..extension.."@"..domain_name.." conference:page-"..destination_number.."@page+flags{mute} inline";
cmd_string = "bgapi originate {sip_auto_answer=true,hangup_after_bridge=false,origination_caller_id_name='"..caller_id_name.."',origination_caller_id_number="..caller_id_number.."}user/"..extension.."@"..domain_name.." conference:page-"..destination_number.."@page+"..flags.." inline";
api:executeString(cmd_string);
destination_count = destination_count + 1;
end
end
end
@ -133,6 +148,10 @@ if ( session:ready() ) then
end
--send main call to the conference room
session:execute("conference", "page-"..destination_number.."@page+flags{endconf}");
if (destination_count > 0) then
session:execute("conference", "page-"..destination_number.."@page+flags{endconf}");
else
session:execute("playback", "tone_stream://%(500,500,480,620);loops=3");
end
end