mirror of
https://github.com/fusionpbx/fusionpbx.git
synced 2025-12-30 00:53:50 +00:00
added delayed paging (#6771)
* added delayed paging * Modified the code to use sched_api as requested * added the beep
This commit is contained in:
Antonio Fernandez
committed by
GitHub
GitHub
parent
c44d1825cf
commit
a4bfd1f20f
@@ -6,6 +6,7 @@
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<action application="set" data="destinations=101-103,105" />
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<action application="set" data="moderator=false" />
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<action application="set" data="mute=true" />
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<action application="set" data="delay=false" />
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<action application="set" data="check_destination_status=true" enabled="false" />
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<action application="set" data="api_hangup_hook=conference page-${destination_number}@${domain_name} hup all" enabled="false"/>
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<action application="set" data="execute_on_answer=sched_hangup +80 allotted_timeout" enabled="false"/>
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@@ -25,176 +25,218 @@
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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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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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require "resources.functions.trim";
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--define the explode function
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require "resources.functions.explode";
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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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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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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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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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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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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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--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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delay = session:getVariable("delay");
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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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--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 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 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 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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--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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--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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--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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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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--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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--if annouce delay is active then prompt for recording
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if (delay == "true") then
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--callback function for the delayed recording
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function onInputCBF(s, _type, obj, arg)
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local k, v = nil, nil
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if (_type == "dtmf") then
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dtmf_entered = 1; --set this variable to know that the user entered DTMF
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return 'break'
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else
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return ''
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end
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end
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--sleep
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session:sleep(500);
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--set variables for page recording
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recording_dir = '/tmp/';
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filename = "page-"..destination_number.."@"..domain_name..".wav";
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recording_filename = string.format('%s%s', recording_dir, filename);
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max_len_secs = 20;
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silence_threshold = 500;
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silence_secs = 1;
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dtmf_entered = 0;
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silence_triggered = 0;
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--ask user to record
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session:execute("playback", sounds_dir.."/"..default_language.."/"..default_dialect.."/"..default_voice.."/voicemail/vm-record_message.wav")
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session:streamFile("tone_stream://L=1;%(1000, 0, 640)");
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--set callback function for when a user clicks DTMF
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session:setInputCallback('onInputCBF', '');
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--time before starting the recording
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startUTCTime = os.time(os.date('!*t'));
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--record the page message
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silence_triggered = session:recordFile(recording_filename, max_len_secs, silence_threshold, silence_secs);
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--time after starting the recording
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endUTCTime = os.time(os.date('!*t'));
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--total recording time
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recording_length = endUTCTime - startUTCTime;
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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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--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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end
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--log the destinations
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freeswitch.consoleLog("NOTICE", "[page] destinations "..destinations.." available\n");
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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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--create the api object
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api = freeswitch.API();
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--log the destinations
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freeswitch.consoleLog("NOTICE", "[page] destinations "..destinations.." available\n");
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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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--create the api object
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api = freeswitch.API();
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--originate the calls
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destination_count = 0;
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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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if (delay ~= "true" or (dtmf_entered == 1 or silence_triggered == 1)) then
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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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@@ -242,17 +284,33 @@
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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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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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--set moderator flag
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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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--check if delay is true
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if (delay == "true" and (dtmf_entered == 1 or silence_triggered == 1)) then
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--play the recorded file into the page/conference. Need to wait for the page/conference to actually be started before we can end it.
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response = api:executeString("sched_api +2 none conference "..conference_name.." play "..recording_filename);
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end
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--wait for recording to finish then end page/conference
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response = api:executeString("sched_api +"..tostring(recording_length+4).." none conference "..conference_name.." hup all");
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else
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--join the moderator into the page
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session:execute("conference", conference_bridge.."+flags{endconf,mintwo"..moderator_flag.."}");
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end
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else
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--error tone due to no destinations
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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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