(14.3) FreeSWITCH VRS Configurations

1- Recording Script

• The first step we need to take on freeswitch is to download file.

• Connect to the VM where free switch is installed.

• Place it in the "scripts" directory of freeswitch. It can be found under "/usr/local/freeswitch/scripts/" or "/usr/share/freeswitch/scripts/" depending on whether you installed freeswitch from source or package, respectively.

• Create a direcotry/folder in the path /usr/share/freeswitch with the name cucmRecording. now enter to this cucmRecording folder and create two more folders/directories with names “streams“ and “sessions”.
  The directory/folder can be created with

mkdir <directry or folder name>

• Edit/open the recording script (record.lua) and assign the correct paths to the "recording_dir" and "mixedRecordingDir" variables.

• The correct path can be found via "pwd” command after checking into each directory, the “pwd” command will print the working direcotry path.

local recording_dir = '/usr/share/freeswitch/cucmRecording/streams'
local mixedRecordingDir = '/usr/share/freeswitch/cucmRecording/sessions'

• Make sure to assign all user permissions to both directories used in these variables.

#chmod 777 -R "full/address/of/directory"

2 - Dialplan Configuration

• Switch to the dialplan direcotry inside the freeswitch.

• Add the following lines of code in the public .xml file of both freeswitches.

<extension name="outside_call" continue="true">
	<condition>
		<action application="set" data="outside_call=true"/>
        <action application="log" data="INFO PP-----${sip_from_host}--------------Starting Record Dialplan --------------12124"/>
        <action application="export" data="RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}"/>
	</condition>
</extension>

<extension name="CUCM Recording Profile">
	<action application="log" data="INFO Entering Call from CUCM"/>
	<condition field="${sip_from_host}" expression="192.168.1.26">
		<action application="lua" data="record.lua"/>
	</condition>
</extension>

3 - SIP Profiles

• Switch to the sip_profiles directory.

• Edit the internal.xml and external.xml SIP profiles and enable or uncomment the Third Party Call Control option in both of them.

• Change the value flag to "true" if it is set to "false" or "proxy".

<param name="enable-3pcc" value="true"/>

• We also need to add one more line to the internal.xml file if it doesn't already exist.

<param name="parse-all-invite-headers" value="true"/>