Minor tweaks and spelling fixes for CHANGES and UPGRADE.txt.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203960 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CHANGES

@@ -50,12 +50,12 @@ IAX2 Changes
 Applications
 ------------
  * Added progress option to the app_dial D() option.  When progress DTMF is
-   present, those values are sent immediatly upon receiving a PROGRESS message
+   present, those values are sent immediately upon receiving a PROGRESS message
    regardless if the call has been answered or not.
  * Added functionality to the app_dial F() option to continue with execution
    at the current location when no parameters are provided.
  * Added c() option to app_chanspy. This option allows custom DTMF to be set
-   to cycle through the next avaliable channel.  By default this is still '*'.
+   to cycle through the next available channel.  By default this is still '*'.
  * Added x() option to app_chanspy.  This option allows DTMF to be set to
    exit the application.
  * The Voicemail application has been improved to automatically ignore messages
@@ -87,7 +87,7 @@ Dialplan Functions
 
    The possible values are:
 
-     on - normal mode (the echo canceller is actually reinitalized)
+     on - normal mode (the echo canceller is actually reinitialized)
      off - disabled
      fax - FAX/data mode (NLP disabled if possible, otherwise completely
            disabled)
@@ -166,12 +166,6 @@ Asterisk Manager Interface
    reflect this change.  Previous options such as 'sslenable' still work,
    but options with the 'tls' prefix are preferred.
 
-Logger
-------
- * The rarely used 'event_log' and LOG_EVENT channel have been removed; the few
-   users of this channel in the tree have been converted to LOG_NOTICE or removed
-   (in cases where the same message was already generated to another channel).
-
 Channel Event Logging
 ---------------------
  * A new interface, CEL, is introduced here. CEL logs single events, much like
@@ -505,12 +499,8 @@ SIP Changes
    as well as periodically updating the IP address. These properties allow for
    better performance as well as recovery in the event of an IP change.
  * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve 
-   load/reload of large numbers of peers/users by ~40x (for large lists of peers.
-   Initially, we saw 4x improvement in call setup/destruction, but at the time
-   of merging, this gain has disappeared; further research will be done to try
-   and restore this performance improvement. Astobj2 refcounting is now used
-   for users, peers, and dialogs.  Users are encouraged to assist in regression
-   testing and problem reporting!
+   load/reload of large numbers of peers/users by ~40x (for large lists of peers).
+   These changes also provide performance improvements for call setup and tear down.
  * Added ability to specify registration expiry time on a per registration basis in
    the register line.
  * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
@@ -1146,11 +1136,11 @@ Logger changes
 --------------
   * Added rotatestrategy option to logger.conf, along with two new options:
      "timestamp" which will use the time to name the logger files instead of
-     sequence number; and "rotate", which rotates the names of the logfiles,
+     sequence number; and "rotate", which rotates the names of the log files,
      similar to the way syslog rotates files.
   * Added exec_after_rotate option to logger.conf, which allows a system
      command to be run after rotation.  This is primarily useful with
-     rotatestrategry=rotate, to allow a limit on the number of logfiles kept
+     rotatestrategry=rotate, to allow a limit on the number of log files kept
      and to ensure that the oldest log file gets deleted.
   * Added realtime support for the queue log
 

UPGRADE.txt

@@ -20,11 +20,16 @@
 
 From 1.6.2 to 1.6.3:
 
+* The rarely used 'event_log' and LOG_EVENT channel have been removed; the few
+  users of this channel in the tree have been converted to LOG_NOTICE or removed
+  (in cases where the same message was already generated to another channel).
+
 * The usage of RTP inside of Asterisk has now become modularized. This means
   the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk.
   If you are not using autoload=yes in modules.conf you will need to ensure
   it is set to load. If not, then any module which uses RTP (such as chan_sip)
   will not be able to send or receive calls.
+
 * The app_dahdiscan.c file has been removed, but the dialplan app DAHDIScan still 
   remains. It now exists within app_chanspy.c and retains the exact same 
   functionality as before.