Remove as much trailing whitespace as possible.

Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0

1 files changed, 38 insertions, 39 deletions

diff --git a/UPGRADE-1.4.txt b/UPGRADE-1.4.txt
index f99f937b0..74cb1e5b4 100644
--- a/UPGRADE-1.4.txt
+++ b/UPGRADE-1.4.txt
@@ -34,7 +34,7 @@ users, that will result in a similar build to what they would have had before
 the configure script was added to the build process (except for having to run
 'make' again after the configure script is run). Note that the configure script
 does NOT need to be re-run just to rebuild Asterisk; you only need to re-run it
-when your system configuration changes or you wish to build Asterisk with 
+when your system configuration changes or you wish to build Asterisk with
 different options.
 
 Build Process (module selection):
@@ -42,7 +42,7 @@ Build Process (module selection):
 The Asterisk source tree now includes a basic module selection and build option
 selection tool called 'menuselect'. Run 'make menuselect' to make your choices.
 In this tool, you can disable building of modules that you don't care about,
-turn on/off global options for the build and see which modules will not 
+turn on/off global options for the build and see which modules will not
 (and cannot) be built because your system does not have the required external
 dependencies installed.
 
@@ -55,7 +55,7 @@ interface.  In this case, the resulting file contains which CFLAGS are in use,
 not which ones are not in use.
 
 If you would like to save your choices and have them applied against all
-builds, the file can be copied to '~/.asterisk.makeopts' or 
+builds, the file can be copied to '~/.asterisk.makeopts' or
 '/etc/asterisk.makeopts'.
 
 Build Process (Makefile targets):
@@ -105,10 +105,10 @@ PBX Core:
 
 * The (very old and undocumented) ability to use BYEXTENSION for dialing
   instead of ${EXTEN} has been removed.
-  
+
 * Builtin (res_features) transfer functionality attempts to use the context
   defined in TRANSFER_CONTEXT variable of the transferer channel first. If
-  not set, it uses the transferee variable. If not set in any channel, it will 
+  not set, it uses the transferee variable. If not set in any channel, it will
   attempt to use the last non macro context. If not possible, it will default
   to the current context.
 
@@ -116,7 +116,7 @@ PBX Core:
   if your dialplan relies on the ability to 'run off the end' of an extension
   and wait for a new extension without using WaitExten() to accomplish that,
   you will need set autofallthrough to 'no' in your extensions.conf file.
- 
+
 Command Line Interface:
 
 * 'show channels concise', designed to be used by applications that will parse
@@ -155,10 +155,10 @@ Applications:
 
 * OSPAuth is added to authenticate OSP tokens in in_bound call setup messages.
 
-* The CONNECT event in the queue_log from app_queue now has a second field 
-  in addition to the holdtime field. It contains the unique ID of the 
-  queue member channel that is taking the call. This is useful when trying 
-  to link recording filenames back to a particular call from the queue.  
+* The CONNECT event in the queue_log from app_queue now has a second field
+  in addition to the holdtime field. It contains the unique ID of the
+  queue member channel that is taking the call. This is useful when trying
+  to link recording filenames back to a particular call from the queue.
 
 * The old/current behavior of app_queue has a serial type behavior
   in that the queue will make all waiting callers wait in the queue
@@ -170,14 +170,14 @@ Applications:
   waits while this happens. This cycle continues until there are
   no more available members or waiting callers, whichever comes first.
   The new behavior, enabled by setting autofill=yes in queues.conf
-  either at the [general] level to default for all queues or 
-  to set on a per-queue level, makes sure that when the waiting 
-  callers are connecting with available members in a parallel fashion 
+  either at the [general] level to default for all queues or
+  to set on a per-queue level, makes sure that when the waiting
+  callers are connecting with available members in a parallel fashion
   until there are no more available members or no more waiting callers,
   whichever comes first. This is probably more along the lines of how
-  one would expect a queue should work and in most cases, you will want 
-  to enable this new behavior. If you do not specify or comment out this 
-  option, it will default to "no" to keep backward compatability with the old 
+  one would expect a queue should work and in most cases, you will want
+  to enable this new behavior. If you do not specify or comment out this
+  option, it will default to "no" to keep backward compatability with the old
   behavior.
 
 * Queues depend on the channel driver reporting the proper state
@@ -197,7 +197,7 @@ Applications:
        call-limit=10
 
 
-* The app_queue application now has the ability to use MixMonitor to 
+* The app_queue application now has the ability to use MixMonitor to
   record conversations queue members are having with queue callers. Please
   see configs/queues.conf.sample for more information on this option.
 
@@ -212,9 +212,9 @@ Applications:
   other external program in order to mix the audio.
 
 * app_meetme: The 'm' option (monitor) is renamed to 'l' (listen only), and
-  the 'm' option now provides the functionality of "initially muted". 
+  the 'm' option now provides the functionality of "initially muted".
   In practice, most existing dialplans using the 'm' flag should not notice
-  any difference, unless the keypad menu is enabled, allowing the user 
+  any difference, unless the keypad menu is enabled, allowing the user
   to unmute themsleves.
 
 * ast_play_and_record would attempt to cancel the recording if a DTMF
@@ -255,10 +255,10 @@ Applications:
   previously used only by EXTENDED_ODBC_STORAGE. This means that you will need to update
   your table format using the schema provided in doc/odbcstorage.txt
 
-* app_waitforsilence: Fixes have been made to this application which changes the 
+* app_waitforsilence: Fixes have been made to this application which changes the
   default behavior with how quickly it returns. You can maintain "old-style" behavior
   with the addition/use of a third "timeout" parameter.
-  Please consult the application documentation and make changes to your dialplan 
+  Please consult the application documentation and make changes to your dialplan
   if appropriate.
 
 Manager:
@@ -270,9 +270,9 @@ Manager:
   until after the release of 1.4, when it will be removed.  Please use the time
   during the 1.4 release to make this transition.
 
-* The AgentConnect event now has an additional field called "BridgedChannel" 
-  which contains the unique ID of the queue member channel that is taking the 
-  call. This is useful when trying to link recording filenames back to 
+* The AgentConnect event now has an additional field called "BridgedChannel"
+  which contains the unique ID of the queue member channel that is taking the
+  call. This is useful when trying to link recording filenames back to
   a particular call from the queue.
 
 * app_userevent has been modified to always send Event: UserEvent with the
@@ -298,17 +298,17 @@ Variables:
   functions.  You are encouraged to move towards the associated dialplan
   function, as these variables will be removed in a future release.
 
-* The CDR-CSV variables uniqueid, userfield, and basing time on GMT are now 
+* The CDR-CSV variables uniqueid, userfield, and basing time on GMT are now
   adjustable from cdr.conf, instead of recompiling.
 
 * OSP applications exports several new variables, ${OSPINHANDLE},
   ${OSPOUTHANDLE}, ${OSPINTOKEN}, ${OSPOUTTOKEN}, ${OSPCALLING},
   ${OSPINTIMELIMIT}, and ${OSPOUTTIMELIMIT}
-  
+
 * Builtin transfer functionality sets the variable ${TRANSFERERNAME} in the new
   created channel. This variables holds the channel name of the transferer.
 
-* The dial plan variable PRI_CAUSE will be removed from future versions 
+* The dial plan variable PRI_CAUSE will be removed from future versions
   of Asterisk.
   It is replaced by adding a cause value to the hangup() application.
 
@@ -322,8 +322,8 @@ Functions:
   modules.conf file then you will need to explicitly load the modules that
   contain the functions you want to use.
 
-* The ENUMLOOKUP() function with the 'c' option (for counting the number of 
-  records), but the lookup fails to match any records, the returned value will 
+* The ENUMLOOKUP() function with the 'c' option (for counting the number of
+  records), but the lookup fails to match any records, the returned value will
   now be "0" instead of blank.
 
 * The REALTIME() function is now available in version 1.4 and app_realtime has
@@ -345,7 +345,7 @@ The IAX2 channel:
   which user you want based on host.
 
   If you would like to go ahead and use the new behavior which doesn't use the
-  order in the config file to influence matching order, then change the 
+  order in the config file to influence matching order, then change the
   MAX_PEER_BUCKETS define in chan_iax2.c to a value greater than one.  An
   example is provided there.  By changing this, you will get *much* better
   performance on systems that do a lot of peer and user lookups as they will be
@@ -360,12 +360,12 @@ The IAX2 channel:
 
 The SIP channel:
 
-* The "incominglimit" setting is replaced by the "call-limit" setting in 
+* The "incominglimit" setting is replaced by the "call-limit" setting in
   sip.conf.
 
-* OSP support code is removed from SIP channel to OSP applications. ospauth 
+* OSP support code is removed from SIP channel to OSP applications. ospauth
   option in sip.conf is removed to osp.conf as authpolicy. allowguest option
-  in sip.conf cannot be set as osp anymore. 
+  in sip.conf cannot be set as osp anymore.
 
 * The Asterisk RTP stack has been changed in regards to RFC2833 reception
   and transmission. Packets will now be sent with proper duration instead of all
@@ -419,10 +419,10 @@ The G726-32 codec:
 Installation:
 
 * On BSD systems, the installation directories have changed to more "FreeBSDish"
-  directories. On startup, Asterisk will look for the main configuration in 
+  directories. On startup, Asterisk will look for the main configuration in
   /usr/local/etc/asterisk/asterisk.conf
-  If you have an old installation, you might want to remove the binaries and 
-  move the configuration files to the new locations. The following directories 
+  If you have an old installation, you might want to remove the binaries and
+  move the configuration files to the new locations. The following directories
   are now default:
 	ASTLIBDIR	/usr/local/lib/asterisk
 	ASTVARLIBDIR	/usr/local/share/asterisk
@@ -476,7 +476,7 @@ CDR Records:
   set.  When using the "callerid" option for various channel drivers, some
   would set ANI and some would not.  This has been cleared up so that all
   channel drivers set ANI.  If you would like to change the callerid number
-  on the channel from the dialplan and have that change also show up in the 
+  on the channel from the dialplan and have that change also show up in the
   CDR, then you *must* set CALLERID(ANI) as well as CALLERID(num).
 
 API:
@@ -492,7 +492,6 @@ Formats:
 * format_wav: The GAIN preprocessor definition has been changed from 2 to 0
   in Asterisk 1.4.  This change was made in response to user complaints of
   choppiness or the clipping of loud signal peaks.  The GAIN preprocessor
-  definition will be retained in Asterisk 1.4, but will be removed in a 
+  definition will be retained in Asterisk 1.4, but will be removed in a
   future release.  The use of GAIN for the increasing of voicemail message
   volume should use the 'volgain' option in voicemail.conf
-