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Update page.lua

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Update page.lua add support for rtp_secure_media=true.
This commit is contained in:
FusionPBX
2016-07-06 14:48:00 -06:00
committed by GitHub
parent ec26328577
commit 0f84dbc9cd
1 changed files with 112 additions and 103 deletions
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resources/install/scripts/page.lua
+112 -103
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@@ -58,6 +58,7 @@
domain_name = session:getVariable("domain_name"); domain_name = session:getVariable("domain_name");
sounds_dir = session:getVariable("sounds_dir"); sounds_dir = session:getVariable("sounds_dir");
destinations = session:getVariable("destinations"); destinations = session:getVariable("destinations");
rtp_secure_media = session:getVariable("rtp_secure_media");
if (destinations == nil) then if (destinations == nil) then
destinations = session:getVariable("extension_list"); destinations = session:getVariable("extension_list");
end end
@@ -66,7 +67,7 @@
caller_id_number = session:getVariable("caller_id_number"); caller_id_number = session:getVariable("caller_id_number");
sip_from_user = session:getVariable("sip_from_user"); sip_from_user = session:getVariable("sip_from_user");
mute = session:getVariable("mute"); mute = session:getVariable("mute");
--set the sounds path for the language, dialect and voice --set the sounds path for the language, dialect and voice
default_language = session:getVariable("default_language"); default_language = session:getVariable("default_language");
default_dialect = session:getVariable("default_dialect"); default_dialect = session:getVariable("default_dialect");
@@ -74,123 +75,131 @@
if (not default_language) then default_language = 'en'; end if (not default_language) then default_language = 'en'; end
if (not default_dialect) then default_dialect = 'us'; end if (not default_dialect) then default_dialect = 'us'; end
if (not default_voice) then default_voice = 'callie'; end if (not default_voice) then default_voice = 'callie'; end
local conference_name = "page-"..destination_number.."-"..domain_name.."@page"
if (caller_id_name) then
--caller id name provided do nothing
else
effective_caller_id_name = session:getVariable("effective_caller_id_name");
caller_id_name = effective_caller_id_name;
end
if (caller_id_number) then
--caller id number provided do nothing
else
effective_caller_id_number = session:getVariable("effective_caller_id_number");
caller_id_number = effective_caller_id_number;
end
--set conference flags
if (mute == "true") then
flags = "flags{mute}";
else
flags = "flags{}";
end
--if the pin number is provided then require it
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
if (digits == pin_number) then
--set the variable to true
auth = true;
--set the authorized pin number that was used
session:setVariable("pin_number", pin_number);
--end the loop
break;
end
end
--if not authorized play a message and then hangup
if (not auth) then
session:streamFile("phrase:voicemail_fail_auth:#");
session:hangup("NORMAL_CLEARING");
return;
end
end
destination_count = 0; --set rtp_secure_media to an empty string if not provided.
api = freeswitch.API(); if (rtp_secure_media == nil) then
for index,value in pairs(destination_table) do rtp_secure_media = 'false';
if (string.find(value, "-") == nil) then
value = value..'-'..value;
end end
sub_table = explode("-",value);
zeros = leading_zeros(sub_table[1]);
for destination=sub_table[1],sub_table[2] do
--add the leading zeros back again --define the conference name
destination = zeros .. destination; local conference_name = "page-"..destination_number.."-"..domain_name.."@page"
--get the destination required for number-alias --set the caller id
destination = api:execute("user_data", destination .. "@" .. domain_name .. " attr id"); if (caller_id_name) then
--caller id name provided do nothing
--prevent calling the user that initiated the page else
if (sip_from_user ~= destination) then effective_caller_id_name = session:getVariable("effective_caller_id_name");
--cmd = "username_exists id "..destination.."@"..domain_name; caller_id_name = effective_caller_id_name;
--reply = trim(api:executeString(cmd)); end
--if (reply == "true") then
destination_status = "show channels like "..destination.."@"; if (caller_id_number) then
reply = trim(api:executeString(destination_status)); --caller id number provided do nothing
if (reply == "0 total.") then else
freeswitch.consoleLog("NOTICE", "[page] destination "..destination.." available\n"); effective_caller_id_number = session:getVariable("effective_caller_id_number");
if (destination == tonumber(sip_from_user)) then caller_id_number = effective_caller_id_number;
--this destination is the caller that initated the page end
else
--originate the call --set conference flags
cmd_string = "bgapi originate {sip_auto_answer=true,sip_h_Alert-Info='Ring Answer',hangup_after_bridge=false,origination_caller_id_name='"..caller_id_name.."',origination_caller_id_number="..caller_id_number.."}user/"..destination.."@"..domain_name.." conference:"..conference_name.."+"..flags.." inline"; if (mute == "true") then
api:executeString(cmd_string); flags = "flags{mute}";
destination_count = destination_count + 1; else
end flags = "flags{}";
--freeswitch.consoleLog("NOTICE", "cmd_string "..cmd_string.."\n"); end
else
--look inside the reply to check for the correct domain_name --if the pin number is provided then require it
if string.find(reply, domain_name) then if (pin_number) then
--found: user is busy --sleep
else session:sleep(500);
--not found --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
if (digits == pin_number) then
--set the variable to true
auth = true;
--set the authorized pin number that was used
session:setVariable("pin_number", pin_number);
--end the loop
break;
end
end
--if not authorized play a message and then hangup
if (not auth) then
session:streamFile("phrase:voicemail_fail_auth:#");
session:hangup("NORMAL_CLEARING");
return;
end
end
--originate the calls
destination_count = 0;
api = freeswitch.API();
for index,value in pairs(destination_table) do
if (string.find(value, "-") == nil) then
value = value..'-'..value;
end
sub_table = explode("-",value);
zeros = leading_zeros(sub_table[1]);
for destination=sub_table[1],sub_table[2] do
--add the leading zeros back again
destination = zeros .. destination;
--get the destination required for number-alias
destination = api:execute("user_data", destination .. "@" .. domain_name .. " attr id");
--prevent calling the user that initiated the page
if (sip_from_user ~= destination) then
--cmd = "username_exists id "..destination.."@"..domain_name;
--reply = trim(api:executeString(cmd));
--if (reply == "true") then
destination_status = "show channels like "..destination.."@";
reply = trim(api:executeString(destination_status));
if (reply == "0 total.") then
freeswitch.consoleLog("NOTICE", "[page] destination "..destination.." available\n");
if (destination == tonumber(sip_from_user)) then if (destination == tonumber(sip_from_user)) then
--this destination is the caller that initated the page --this destination is the caller that initated the page
else else
--originate the call --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/"..destination.."@"..domain_name.." conference:"..conference_name.."+"..flags.." inline"; 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_name.."+"..flags.." inline";
api:executeString(cmd_string); api:executeString(cmd_string);
destination_count = destination_count + 1; destination_count = destination_count + 1;
end end
--freeswitch.consoleLog("NOTICE", "cmd_string "..cmd_string.."\n");
else
--look inside the reply to check for the correct domain_name
if string.find(reply, domain_name) then
--found: user is busy
else
--not found
if (destination == tonumber(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,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_name.."+"..flags.." inline";
api:executeString(cmd_string);
destination_count = destination_count + 1;
end
end
end end
end --end
--end end
end end
end end
end
--send main call to the conference room --send main call to the conference room
if (destination_count > 0) then if (destination_count > 0) then
if (session:getVariable("moderator") == "true") then if (session:getVariable("moderator") == "true") then
moderator_flag = ",moderator"; moderator_flag = ",moderator";
else
moderator_flag = "";
end
session:execute("conference", conference_name.."+flags{endconf"..moderator_flag.."}");
else else
moderator_flag = ""; session:execute("playback", "tone_stream://%(500,500,480,620);loops=3");
end end
session:execute("conference", conference_name.."+flags{endconf"..moderator_flag.."}");
else
session:execute("playback", "tone_stream://%(500,500,480,620);loops=3");
end
end end