FS-10593: clean up whitespaces from several xml files.

conf/vanilla/autoload_configs/acl.conf.xml

@@ -1,6 +1,6 @@
 <configuration name="acl.conf" description="Network Lists">
   <network-lists>
-    <!-- 
+    <!--
 	 These ACL's are automatically created on startup.
 
 	 rfc1918.auto  - RFC1918 Space
@@ -15,9 +15,9 @@
     </list>
 
     <!--
-	This will traverse the directory adding all users 
+	This will traverse the directory adding all users
 	with the cidr= tag to this ACL, when this ACL matches
-	the users variables and params apply as if they 
+	the users variables and params apply as if they
 	digest authenticated.
     -->
     <list name="domains" default="deny">

conf/vanilla/autoload_configs/conference.conf.xml

@@ -1,4 +1,4 @@
-<!-- http://wiki.freeswitch.org/wiki/Mod_conference --> 
+<!-- http://wiki.freeswitch.org/wiki/Mod_conference -->
 <!-- None of these paths are real if you want any of these options you need to really set them up -->
 <configuration name="conference.conf" description="Audio Conference">
   <!-- Advertise certain presence on startup . -->
@@ -35,7 +35,7 @@
 	   absolute path means <value>/<confernece_uuid>.cdr.xml
       -->
       <!-- <param name="cdr-log-dir" value="auto"/> -->
-      
+
       <!-- Domain (for presence) -->
       <param name="domain" value="$${domain}"/>
       <!-- Sample Rate-->
@@ -251,7 +251,7 @@
       <param name="caller-id-number" value="$${outbound_caller_id}"/>
       <param name="comfort-noise" value="false"/>
       <param name="conference-flags" value="video-floor-only|rfc-4579|livearray-sync|minimize-video-encoding"/>
-      <param name="video-mode" value="mux"/> 
+      <param name="video-mode" value="mux"/>
       <param name="video-layout-name" value="3x3"/>
       <param name="video-layout-name" value="group:grid"/>
       <param name="video-canvas-size" value="1920x1080"/>
@@ -261,7 +261,7 @@
       <param name="video-fps" value="15"/>
     </profile>
 
-    
+
     <profile name="sla">
       <param name="domain" value="$${domain}"/>
       <param name="rate" value="16000"/>

conf/vanilla/dialplan/default.xml

@@ -1,10 +1,10 @@
 <?xml version="1.0" encoding="utf-8"?>
 <!--
     NOTICE:
-    
+
-    This context is usually accessed via authenticated callers on the sip profile on port 5060 
+    This context is usually accessed via authenticated callers on the sip profile on port 5060
     or transfered callers from the public context which arrived via the sip profile on port 5080.
-    
+
     Authenticated users will use the user_context variable on the user to determine what context
     they can access.  You can also add a user in the directory with the cidr= attribute acl.conf.xml
     will build the domains ACL using this value.
@@ -20,7 +20,7 @@
       </condition>
     </extension>
 
-    <!-- Example of doing things based on time of day. 
+    <!-- Example of doing things based on time of day.
 
 	 year = 4 digit year. Example year="2009"
 	 yday = 1-365
@@ -32,7 +32,7 @@
 	 hour = 0-23
 	 minute = 0-59
 	 minute-of-day = 1-1440
-	 
+
 	 Example:
 	 <condition minute-of-day="540-1080"> (9am to 6pm EVERY day)
 	 do something ...
@@ -135,7 +135,7 @@
 	<action application="sleep" data="10000"/>
       </condition>
       <!--
-	  This is an example of how to auto detect if telephone-event is missing and activate inband detection 
+	  This is an example of how to auto detect if telephone-event is missing and activate inband detection
       -->
       <!--
       <condition field="${switch_r_sdp}" expression="a=rtpmap:(\d+)\stelephone-event/8000" break="never">
@@ -189,7 +189,7 @@
 	<action application="transfer" data="3000"/>
       </condition>
     </extension>
-    
+
     <extension name="snom-demo-1">
       <condition field="destination_number" expression="^9000$">
 	<!--<key> <light> <label> <user> <host> <profile> <action_name> <action>-->
@@ -257,10 +257,10 @@
       </condition>
     </extension>
 
-    <!-- 
+    <!--
-	 dial the extension (1000-1019) for 30 seconds and go to voicemail if the 
+	 dial the extension (1000-1019) for 30 seconds and go to voicemail if the
 	 call fails (continue_on_fail=true), otherwise hang up after a successful
-	 bridge (hangup_after_bridge=true) 
+	 bridge (hangup_after_bridge=true)
     -->
     <extension name="Local_Extension">
       <condition field="destination_number" expression="^(10[01][0-9])$">
@@ -336,11 +336,11 @@
       <condition field="destination_number" expression="^vmain$|^4000$|^\*98$">
 	<action application="answer"/>
 	<action application="sleep" data="1000"/>
-	<action application="voicemail" data="check default ${domain_name}"/>  
+	<action application="voicemail" data="check default ${domain_name}"/>
       </condition>
     </extension>
 
-    <!-- 
+    <!--
 	 This extension is used by mod_portaudio so you can pa call sip:someone@example.com
 	 mod_portaudio will pass the entire string to the dialplan for routing.
     -->
@@ -352,7 +352,7 @@
 
     <!--
 	start a dynamic conference with the settings of the "default" conference profile in conference.conf.xml
-    -->                                                                                                                                                       
+    -->
     <extension name="nb_conferences">
       <condition field="destination_number" expression="^(30\d{2})$">
 	<action application="answer"/>
@@ -418,7 +418,7 @@
     <extension name="freeswitch_public_conf_via_sip">
       <condition field="destination_number" expression="^9(888|8888|1616|3232)$">
 	<action application="export" data="hold_music=silence"/>
-	<!-- 
+	<!--
 	     This will take the SAS from the b-leg and send it to the display on the a-leg phone.
 	     Known working with Polycom and Snom maybe others.
 	-->
@@ -432,7 +432,7 @@
 
     <!--
 	This extension will start a conference and invite a group.
-	At anytime the participant can dial *2 to bridge directly to the boss. 
+	At anytime the participant can dial *2 to bridge directly to the boss.
 	All other callers are then hung up on.
     -->
     <extension name="mad_boss_intercom">
@@ -450,7 +450,7 @@
 
     <!--
 	This extension will start a conference and invite a few of people.
-	At anytime the participant can dial *2 to bridge directly to the boss. 
+	At anytime the participant can dial *2 to bridge directly to the boss.
 	All other callers are then hung up on.
     -->
     <extension name="mad_boss_intercom">
@@ -490,7 +490,7 @@
       </condition>
     </extension>
 
-    <!-- Create a conference on the fly and pull someone in at the same time. --> 
+    <!-- Create a conference on the fly and pull someone in at the same time. -->
     <extension name="dynamic_conference">
       <condition field="destination_number" expression="^5001$">
 	<action application="conference" data="bridge:mydynaconf:sofia/${use_profile}/1234@conference.freeswitch.org"/>
@@ -504,7 +504,7 @@
       </condition>
     </extension>
 
-    <!-- 
+    <!--
 	 Parking extensions... transferring calls to 5900 will park them in a queue.
     -->
     <extension name="park">
@@ -514,7 +514,7 @@
       </condition>
     </extension>
 
-    <!-- 
+    <!--
 	 Parking pickup extension.  Calling 5901 will pickup the call.
     -->
     <extension name="unpark">
@@ -524,7 +524,7 @@
       </condition>
     </extension>
 
-    <!-- 
+    <!--
 	 Valet park retrieval, works with valet_park extension below.
 	 Retrieve a valet parked call by dialing 6000 + park number + #
     -->
@@ -535,7 +535,7 @@
       </condition>
     </extension>
 
-    <!-- 
+    <!--
 	 Valet park 6001-6099.  Blind x-fer to 6001, 6002, etc. to valet park the call.
 	 Dial 6001, 6002, etc. to retrieve a call that is already valet parked.
 	 After call is retrieved, park extension is free for another call.
@@ -549,10 +549,10 @@
 
 
     <!--
-	This extension is used with Snom phones.  
+	This extension is used with Snom phones.
-	
+
 	Set a function key to park+lot (lot being a number or name.)
-	Set type to Park+Orbit.  You can then park and pickup using 
+	Set type to Park+Orbit.  You can then park and pickup using
 	the softkey on the phone.  Should work with other phones.
     -->
     <extension name="park">
@@ -560,9 +560,9 @@
       <condition field="destination_number" expression="park\+(\d+)">
 	<action application="fifo" data="$1@${domain_name} in undef $${hold_music}"/>
       </condition>
-    </extension> 
+    </extension>
     <!--
-	The extension is parking pickup with a to param of the fifo we are calling 
+	The extension is parking pickup with a to param of the fifo we are calling
 	Some phones send things like orbit= and you can extract that info.
     -->
     <extension name="unpark">
@@ -590,7 +590,7 @@
 	<action application="fifo" data="$1@${domain_name} in undef $${hold_music}"/>
       </condition>
     </extension>
-    
+
     <!--
        This extension is used with Linksys phones.
 
@@ -610,7 +610,7 @@
     <!--
 	Here are some examples of how to override the ringback heard by the
 	far end.  You have two variables that you can use to override this.
-	
+
 	ringback          - used when a call isn't answered. (early media)
 	transfer_ringback - used when the call is already answered. (post answer)
     -->
@@ -626,7 +626,7 @@
 	<action application="hangup"/>
       </condition>
     </extension>
-    
+
     <extension name="fax_receive">
       <condition field="destination_number" expression="^9178$">
 	<action application="answer" />
@@ -786,7 +786,7 @@
 	You can place files in the default directory to get included.
     -->
     <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
-    
+
     <!--
     <extension name="refer">
       <condition field="${sip_refer_to}">
@@ -810,7 +810,7 @@
 	<action application="bridge" data="sofia/${ruri_profile}/${ruri_user}${regex(${sofia_contact(${ruri_contact})}|^[^\@]+(.*)|%1)}"/>
       </condition>
     </extension>
-    
+
     <extension name="7004">
       <condition field="destination_number" expression="^7004$">
 	<action application="set" data="ruri_profile=default"/>

conf/vanilla/dialplan/default/01_Talking_Clock.xml

@@ -8,7 +8,7 @@
       <action application="hangup"/>
     </condition>
   </extension>
-  
+
   <extension name="Talking Clock Date" ><!--e.g. March 8, 2011-->
     <condition field="destination_number" expression="^9171$">
       <action application="answer"/>
@@ -18,7 +18,7 @@
       <action application="hangup"/>
     </condition>
   </extension>
-  
+
   <extension name="Talking Clock Date and Time" ><!--e.g. March 8, 2011
 						     10:56pm-->
     <condition field="destination_number" expression="^9172$">

conf/vanilla/dialplan/public/00_inbound_did.xml

@@ -4,12 +4,12 @@
       <!--
 	  If you're hosting multiple domains you will want to set the
 	  target_domain on these calls so they hit the proper domain after you
-	  transfer the caller into the default context. 
+	  transfer the caller into the default context.
-	  
+
 	  $${domain} is the default domain set from vars.xml but you can set it
 	  to any domain you have setup in your user directory.
 
-      --> 
+      -->
       <action application="set" data="domain_name=$${domain}"/>
       <!-- This example maps the DID 5551212 to ring 1000 in the default context -->
       <action application="transfer" data="1000 XML default"/>

conf/vanilla/dialplan/skinny-patterns.xml

@@ -25,6 +25,6 @@
 	You can place files in the skinny-patterns directory to get included.
     -->
     <X-PRE-PROCESS cmd="include" data="skinny-patterns/*.xml"/>
-    
+
   </context>
 </include>

conf/vanilla/directory/default.xml

@@ -1,9 +1,9 @@
 <!--
-    NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE 
+    NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
-    
+
     FreeSWITCH works off the concept of users and domains just like email.
     You have users that are in domains for example 1000@domain.com.
-    
+
     When freeswitch gets a register packet it looks for the user in the directory
     based on the from or to domain in the packet depending on how your sofia profile
     is configured.  Out of the box the default domain will be the IP address of the
@@ -11,10 +11,10 @@
     CLI.  You will register your phones to the IP and not the hostname by default.
     If you wish to register using the domain please open vars.xml in the root conf
     directory and set the default domain to the hostname you desire.  Then you would
-    use the domain name in the client instead of the IP address to register 
+    use the domain name in the client instead of the IP address to register
     with FreeSWITCH.
-    
+
-    NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE 
+    NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
 -->
 
 <include>
@@ -44,7 +44,7 @@
       <group name="sales">
 	<users>
 	  <!--
-	      type="pointer" is a pointer so you can have the 
+	      type="pointer" is a pointer so you can have the
 	      same user in multiple groups.  It basically means
 	      to keep searching for the user in the directory.
 	  -->

conf/vanilla/extensions.conf

@@ -15,7 +15,7 @@ exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,bridge(${enum_auto_route})
 
 ; instead of exten, put anything about the call you would rather match on.
 ; either the names of a field in caller_profile or a string of variables to expand.
-caller_id_number => 2137991400,n,Goto(default|music) 
+caller_id_number => 2137991400,n,Goto(default|music)
 ${sip_from_user} => bill,n,Goto(default|music)
 
 

conf/vanilla/freeswitch.xml

@@ -1,24 +1,24 @@
 <?xml version="1.0"?>
 <!--
-    NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE 
+    NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
-    
+
-    This is the FreeSWITCH default config.  Everything you see before you now traverses 
+    This is the FreeSWITCH default config.  Everything you see before you now traverses
     down into all the directories including files which include more files.  The default
     config comes out of the box already working in most situations as a PBX.  This will
     allow you to get started testing and playing with various things in FreeSWITCH.
-    
+
     Before you start to modify this default please visit this wiki page:
-    
+
     http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Some_stuff_to_try_out.21
-    
+
     If all else fails you can read our FAQ located at:
-    
+
     http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ
-    
+
-    NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE 
+    NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
 -->
 <document type="freeswitch/xml">
-  <!--#comment 
+  <!--#comment
       All comments starting with #command will be preprocessed and never sent to the xml parser
       Valid instructions:
       #include ==> Include another file to this exact point
@@ -26,10 +26,10 @@
       #set     ==> Set a global variable (can be expanded during preprocessing with $$ variables)
                    (note the double $$ which denotes preprocessor variables)
       #comment ==> A general comment such as this
-      
+
       The preprocessor will compile the full xml document to ${prefix}/log/freeswitch.xml.fsxml
       Don't modify it while freeswitch is running cos it is mem mapped in most cases =D
-      
+
       The same can be achieved with the <X-PRE-PROCESS> tag where the attrs 'cmd' and 'data' are
       parsed in the same way.
   -->
@@ -41,7 +41,7 @@
   <section name="configuration" description="Various Configuration">
     <X-PRE-PROCESS cmd="include" data="autoload_configs/*.xml"/>
   </section>
-  
+
   <section name="dialplan" description="Regex/XML Dialplan">
     <X-PRE-PROCESS cmd="include" data="dialplan/*.xml"/>
   </section>
@@ -51,7 +51,7 @@
   </section>
 
   <!-- mod_dingaling is reliant on the vcard data in the "directory" section. -->
-  <!-- mod_sofia is reliant on the user data for authorization --> 
+  <!-- mod_sofia is reliant on the user data for authorization -->
   <section name="directory" description="User Directory">
     <X-PRE-PROCESS cmd="include" data="directory/*.xml"/>
   </section>

conf/vanilla/ivr_menus/demo_ivr.xml

@@ -27,7 +27,7 @@
     <entry action="menu-sub" digits="6" param="demo_ivr_submenu"/>                  <!-- demo sub menu -->
     <!-- Using a regex in the digits tag lets you define a dial pattern for the caller
          You may define multiple regexes if you need a different pattern for some reason -->
-    <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/> 
+    <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/>
     <entry action="menu-top" digits="9"/>          <!-- Repeat this menu -->
   </menu>
 

conf/vanilla/sip_profiles/internal.xml

@@ -310,7 +310,7 @@
 
     <!-- for sip over secure websocket support -->
     <!-- You need wss.pem in $${certs_dir} for wss or one will be created for you -->
-    <param name="wss-binding" value=":7443"/>    
+    <param name="wss-binding" value=":7443"/>
 
     <!--<param name="delete-subs-on-register" value="false"/>-->
 

conf/vanilla/vars.xml

@@ -1,26 +1,26 @@
 <include>
   <!-- Preprocessor Variables
-       These are introduced when configuration strings must be consistent across modules. 
+       These are introduced when configuration strings must be consistent across modules.
        NOTICE: YOU CAN NOT COMMENT OUT AN X-PRE-PROCESS line, Remove the line instead.
-       
+
-       WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 
+       WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
-       
+
        YOU SHOULD CHANGE THIS default_password value if you don't want to be subject to any
        toll fraud in the future.  It's your responsibility to secure your own system.
-       
+
        This default config is used to demonstrate the feature set of FreeSWITCH.
-       
+
-       WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 
+       WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
   -->
   <X-PRE-PROCESS cmd="set" data="default_password=1234"/>
   <!-- Did you change it yet? -->
   <!--
-      The following variables are set dynamically - calculated if possible by freeswitch - and 
+      The following variables are set dynamically - calculated if possible by freeswitch - and
       are available to the config as $${variable}.  You can see their calculated value via fs_cli
       by entering eval $${variable}
-      
+
       hostname
-      local_ip_v4 
+      local_ip_v4
       local_mask_v4
       local_ip_v6
       switch_serial
@@ -45,7 +45,7 @@
       nat_public_addr
       nat_private_addr
       nat_type
-      
+
   -->
 
 
@@ -54,8 +54,8 @@
 
   <!--
       This setting is what sets the default domain FreeSWITCH will use if all else fails.
-      
+
-      FreeSWICH will default to $${local_ip_v4} unless changed.  Changing this setting does 
+      FreeSWICH will default to $${local_ip_v4} unless changed.  Changing this setting does
       affect the sip authentication.  Please review conf/directory/default.xml for more
       information on this topic.
   -->
@@ -66,7 +66,7 @@
   <X-PRE-PROCESS cmd="set" data="rtp_sdes_suites=AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH"/>
   <!--
       Enable ZRTP globally you can override this on a per channel basis
-      
+
       http://wiki.freeswitch.org/wiki/ZRTP (on how to enable zrtp)
   -->
   <X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/>
@@ -169,13 +169,13 @@
       and only use rtp_secure_media=[optional|mandatory|false|true] without having
       to dictate the suite list with the rtp_secure_media* variables.
   -->
-  <!-- 
+  <!--
        Examples of codec options: (module must be compiled and loaded)
-       
+
        codecname[@8000h|16000h|32000h[@XXi]]
-       
+
        XX is the frame size must be multples allowed for the codec
-       FreeSWITCH can support 10-120ms on some codecs. 
+       FreeSWITCH can support 10-120ms on some codecs.
        We do not support exceeding the MTU of the RTP packet.
 
 
@@ -205,22 +205,22 @@
        AAL2-G726-40     - Same as G726-40 but using AAL2 packing. (multiples of 10)
        LPC              - LPC10 using 90ms ptime (only supports 90ms at this time in FreeSWITCH)
        L16              - L16 isn't recommended for VoIP but you can do it. L16 can exceed the MTU rather quickly.
-       
+
        These are the passthru audio codecs:
-       
+
        G729             - G729 in passthru mode. (mod_g729)
        G723             - G723.1 in passthru mode. (mod_g723_1)
        AMR              - AMR in passthru mode. (mod_amr)
-       
+
        These are the passthru video codecs: (mod_h26x)
-       
+
        H261             - H.261 Video
        H263             - H.263 Video
        H263-1998        - H.263-1998 Video
        H263-2000        - H.263-2000 Video
        H264             - H.264 Video
-       
+
-       RTP Dynamic Payload Numbers currently used in FreeSWITCH and what for.
+       RTP Dynamic Payload Numbers currently used in FreeSWITCH and their purpose.
 
        96  - AMR
        97  - iLBC (30)
@@ -229,9 +229,9 @@
        100 -
        101 - telephone-event
        102 -
-       103 - 
+       103 -
-       104 - 
+       104 -
-       105 - 
+       105 -
        106 - BV16
        107 - G722.1 (16kHz)
        108 -
@@ -251,7 +251,7 @@
        122 - AAL2-G726-32 && G726-32
        123 - AAL2-G726-24 && G726-24
        124 - AAL2-G726-16 && G726-16
-       125 - 
+       125 -
        126 -
        127 - BV32
 
@@ -261,20 +261,20 @@
 
   <!--
       xmpp_client_profile and xmpp_server_profile
-      xmpp_client_profile can be any string. 
+      xmpp_client_profile can be any string.
       xmpp_server_profile is appended to "dingaling_" to form the database name
       containing the "subscriptions" table.
-      used by: dingaling.conf.xml enum.conf.xml 
+      used by: dingaling.conf.xml enum.conf.xml
-  --> 
+  -->
 
   <X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/>
   <X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/>
-  <!-- 
+  <!--
        THIS IS ONLY USED FOR DINGALING
 
        bind_server_ip
 
-       Can be an ip address, a dns name, or "auto". 
+       Can be an ip address, a dns name, or "auto".
        This determines an ip address available on this host to bind.
        If you are separating RTP and SIP traffic, you will want to have
        use different addresses where this variable appears.
@@ -283,7 +283,7 @@
   <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
 
   <!-- NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
-       
+
        If you're going to load test FreeSWITCH please input real IP addresses
        for external_rtp_ip and external_sip_ip
   -->
@@ -315,7 +315,7 @@
 
   <!-- unroll-loops
        Used to turn on sip loopback unrolling.
-  --> 
+  -->
   <X-PRE-PROCESS cmd="set" data="unroll_loops=true"/>
 
   <!-- outbound_caller_id and outbound_caller_name
@@ -370,11 +370,11 @@
 
   <!--
        Digits Dialed filter: (FS-6940)
-       
+
        The digits stream may contain valid credit card numbers or social security numbers, These digit
-       filters will allow you to make a valant effort to stamp out sensitive information for 
+       filters will allow you to make a valant effort to stamp out sensitive information for
        PCI/HIPPA compliance. (see xml_cdr dialed_digits)
-       
+
        df_us_ssn   = US Social Security Number pattern
        df_us_luhn  = Visa, MasterCard, American Express, Diners Club, Discover and JCB
   -->
@@ -386,7 +386,7 @@
   <!--
       Setting up your default sip provider is easy.
       Below are some values that should work in most cases.
-      
+
       These are for conf/directory/default/example.com.xml
   -->
   <X-PRE-PROCESS cmd="set" data="default_provider=example.com"/>
@@ -399,7 +399,7 @@
 
   <!--
      SIP and TLS settings. http://wiki.freeswitch.org/wiki/Tls
-     
+
      valid options: sslv2,sslv3,sslv23,tlsv1,tlsv1.1,tlsv1.2
 
      default: tlsv1,tlsv1.1,tlsv1.2
@@ -416,7 +416,7 @@
      Will show you what is available in your verion of openssl.
   -->
   <X-PRE-PROCESS cmd="set" data="sip_tls_ciphers=ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH"/>
-  
+
   <!-- Internal SIP Profile -->
   <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
   <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>