various cleanups

git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44627 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 24 insertions, 135 deletions

diff --git a/CHANGES b/CHANGES
index da3b23bfe..ddcc7bace 100644
--- a/CHANGES
+++ b/CHANGES
@@ -1,4 +1,4 @@
-Changes since Asterisk 1.2.0-beta1:
+Changes since Asterisk 1.2:
 
     * over 4,000 commits since 1.2
     * queue member naming
@@ -8,7 +8,7 @@ Changes since Asterisk 1.2.0-beta1:
     * chan_h323 update
     * multi-parking
     * RTP packetization
-    * SLA (Shared Line Appearance) support various apps (meetme, etc).
+    * SLA (Shared Line Appearance) support
     * T.38 Passthrough Support for faxing
     * Generic channel jitterbuffer (spawned from RTP)
     * VLDTMF for better DTMF compatibility
@@ -17,39 +17,36 @@ Changes since Asterisk 1.2.0-beta1:
       read: http://www.voip-info.org/wiki/view/Asterisk+AEL2
     * New sounds; English, Spanish, and French prompts, as well as music on hold files, in multiple Asterisk native formats.
     * IMAP storage of voicemail
-    * Jabber/Jingle
-    * New speech recognition API for interfacing to different Voice Recognition software packages.
-    * much more customizable build system
+    * Jabber/Jingle/GoogleTalk
+    * New speech recognition API for interfacing to different Voice Recognition software packages
+    * much more customizable and portable build system
           o also for asterisk-addons
     * Radius CDR logging
     * SNMP support
-    * STUN support in SIP
     * SMDI (Simplified Message Desk Interface) support
-    * Manager over http
+    * Manager over HTTP
     * Significant chan_skinny updates
     * Significant chan_misdn updates
-    * improved SIP transfers
-    * ChanSpy whisper mode (whisper Paging)
+    * Improved SIP transfers
+    * ChanSpy whisper mode (Whisper Paging)
     * Configurable language support for saying dates and times
     * Significant architecture improvements for memory usage and performance
-    * Partial IAX2 transfers
+    * Media-only IAX2 transfers
     * Updates to the Radio Repeater app code
-    * deprecation of agentcallbacklogin
+    * Deprecation of AgentCallbackLogin in favor of a dialplan-based solution
     * uClibc builds supported
-    * work done for cygwin portability
-    * work done for freeBSD portability
-    * a lot of work done for Solaris portability
+    * Work done for freeBSD portability
+    * Work done for Solaris portability
     * FreeTDS-based database can be used with Realtime
     * New internal data structure, stringfields, is implemented in IAX and SIP, reducing memory consumption by about 50%.
-    * for asterisk internal use, threadstorage is code to handle dynamically sized thread local buffers. Used in several places.
-    * New default echo canceler
-    * Reorganized files into docs/ main/ configs/, including name changes in some cases.
+    * Use of thread local storage for reduced memory allocation/freeing and lower stack consumption
+    * Reorganized files into docs/ main/ configs/, including name changes in some cases
     * Much effort was expended in arranging documentation in source files in doxygen format
     * Improved IP TOS support for IAX and SIP
-    * builtin mini-http server
+    * Builtin mini HTTP server
     * Added support for Sigma Designs cards.
-    * Frame Caching, an internal methodology to increase performance.
-    * using mpg123 to play MP3 files for music-on-hold will be deprecated in 1.4 (start using the "native support").
+    * Frame header caching to reduce memory allocation/freeing
+    * using mpg123 to play MP3 files for music-on-hold will be deprecated in 1.4 (start using the "native support")
     * New Apps:
          1. AMD() ;; Answering Machine Detection
          2. ChannelRedirect() ;; asynch goto, redirect chan to context/exten/priority
@@ -93,7 +90,7 @@ Changes since Asterisk 1.2.0-beta1:
         15. SetCIDnum -- use the function CALLERID(number) instead
         16. SetGroup -- use Set(GROUP=group) instead
         17. SetRDNIS -- use the function CALLERID(rdnis) instead
-        18. Sql_postgres -- ? Why was this dropped ??
+        18. Sql_postgres -- was deprecated in 1.2, now removed
         19. Txtcidname -- use the function TXTCIDNAME instead
     * New Funcs:
          1. ARRAY()
@@ -107,7 +104,7 @@ Changes since Asterisk 1.2.0-beta1:
          9. GLOBAL()
         10. IFTIME()
         11. KEYPADHASH()
-        12. ODBC interface;
+        12. ODBC()
         13. QUOTE()
         14. RAND()
         15. REALTIME()
@@ -159,7 +156,7 @@ Changes since Asterisk 1.2.0-beta1:
         19. WaitForSilence() -- new optional 3rd arg, time delay before returning.
     * Funcs that have changes to their interfaces:
          1. CDR -- new option: u
-         2. LANGUAGE -- DEPRECATED in 1.4, Use CHANNEL(language) instead.
+         2. LANGUAGE -- Deprecated. Use CHANNEL(language) instead.
          3. MUSICCLASS -- Deprecated. Use CHANNEL(musicclass) instead.
     * Config File Changes:
          1. NEW config files:
@@ -171,7 +168,6 @@ Changes since Asterisk 1.2.0-beta1:
                6. http.conf -- config for the builtin mini-http server in asterisk
                7. jabber.conf -- jabber interface
                8. jingle.conf -- jingle protocol interface config
-               9. muted.conf -- signal muted so you quiet down the sound card while you are on the phone.
               10. res_snmp.conf -- to enable snmp in asterisk, and define full/sub agent status
               11. say.conf -- define per-language rules for numbers, dates, etc.
               12. skinny.conf -- for those special skinny phones you want to use...
@@ -204,14 +200,12 @@ Changes since Asterisk 1.2.0-beta1:
                       o sections for csv and radius added, with variables usegmtime, loguniqueid, 
                         loguserfield, and radiuscfg variables.
                5. cdr_tds.conf
-                      o table variable addedextensions.ael
+                      o table variable added
                6. extensions.ael
                       o Many upgrades. See the info at http://www.voip-info.org/wiki/view/Asterisk+AEL2
                7. extensions.conf
                       o autofallthru now set to "yes" by default
                       o userscontext variable added
-                      o global and environment variables can no longer be reached directly (via ${varname} references. 
-                        You have to use ${GLOBAL(varname)} and ${ENV(varname)} now.
                       o added info/examples on paging and hints.
                8. features.conf
                       o parkedplay variable added (who to beep at)
@@ -275,7 +269,6 @@ Changes since Asterisk 1.2.0-beta1:
                       o autofill variable added
                       o autopause variable added
                       o setinterfacevar variable added
-                      o monitor-type variable added
                       o ringinuse variable added
               19. res_odbc.conf
                       o pooling variable added
@@ -297,7 +290,7 @@ Changes since Asterisk 1.2.0-beta1:
                       o t1min variable added
                       o musicclass variable REMOVED
                       o mohinterpret variable added
-                      o mohmaxcallbitratesuggest variable added
+                      o maxcallbitratesuggest variable added
                       o allowsubscribe variable added
                       o videosupport variable added
                       o maxcallbitrate variable added
@@ -305,7 +298,7 @@ Changes since Asterisk 1.2.0-beta1:
                       o dumphistory variable added
                       o allowsubscribe variable added
                       o t38pt_udptl variable added
-                      o canreinvite variable can also now be set to 'nonat' and 'update'
+                      o canreinvite variable can also now be set to 'nonat'
                       o rtsavesysname variable added
                       o JitterBuffer support added
               23. skinny.conf
@@ -335,111 +328,7 @@ Changes since Asterisk 1.2.0-beta1:
                       o mohsuggest variable added
                       o JitterBuffer support added
     * Removed Codecs/Channels:
-         1. codec_g723 was removed because the actual codec implementation it was designed to use is not available
+         1. codec_g723 was removed because the actual codec implementation it was designed to use is not distributable
          2. chan_modem_* stuff is gone because the kernel support for those interfaces is old, buggy and unsupported
     * New Utils:
          1. aelparse -- compile .ael files outside of asterisk
-         2. muted -- turn down the volume on the sound card when certain phones are ringing or off-hook... automagically.
-
-Changes since Asterisk 1.0:
-
-This list currently only containts changes made from the end of November until
-March 26, 2005.
-
-   * Add new applications:
-     -- AgentMonitorOutgoing
-     -- Curl
-     -- ExecIf
-     -- ExecIfTime
-     -- IAX2Provision
-     -- MacroExit
-     -- MacroIf
-     -- PauseQueueMember
-     -- ReadFile
-     -- SetRDNIS
-     -- SIPAddHeader
-     -- SIPGetHeader
-     -- StartMusicOnHold
-     -- StopMusicOnHold
-     -- UnpauseQueueMember
-     -- WaitForSilence
-     -- While / EndWhile
-   * app Answer
-     -- added delay option
-   * app ChanIsAvail
-     -- added 's' option
-   * app Dial
-     -- add option to specify the class for musiconhold with m option
-   * app EnumLookup
-     -- added "reload enum" for configuration
-   * app Goto
-     -- added relative priorities
-   * app GotoIf
-     -- added relative priorities
-   * app MeetMe
-     -- added 'i' option
-     -- added 'r' option
-     -- added 'T' option
-     -- added 'P' option
-     -- added 'c' option
-     -- added adminpin to meetme.conf
-     -- added reload command
-   * app PrivacyManager
-     -- add config file privacy.conf
-   * app queue
-     -- queues.conf
-        -- added persistentmembers option to queues.conf
-        -- changed music option to musiconhold
-        -- added weight option
-        -- added note about why agent groups probably shouldn't be used
-        -- added timeoutrestart option
-   * app Read
-     -- added attempts parameter
-     -- added timeout parameter
-   * app Record
-     -- added 'q' option
-   * app SendDTMF
-     -- add timeout option
-   * app SMS
-     -- document alternative syntax for queueing messages
-   * app Voicemail
-     -- add info about VM_CATEGORY
-     -- voicemail.conf
-        -- added usedirectory option
-        -- added VM_CIDNUM and VM_CIDNAME in message config
-   * chan IAX2
-     -- new jitterbuffer
-     -- added setvar option
-     -- added regex to iax2 show peers/users
-     -- allow multiple bindaddr lines in iax.conf
-     -- added reload command
-     -- added forcejitterbuffer option
-     -- added note about specifying bindport before bindaddr
-     -- added trunktimestamps option
-   * chan Agent
-     -- added agent logoff CLI command
-   * chan OSS
-     -- added Flash CLI command
-   * chan SIP
-     -- added setvar option
-     -- added compactheaders option
-     -- added usereqphone option
-     -- added registertimeout option
-     -- added externhost option
-     -- added sip notify CLI command
-     -- added sip_notify.conf
-     -- added allowguest option
-   * chan Zap
-     -- added hanguponplarityswitch option
-     -- added sendcalleridafter option
-     -- added priresetinterval option
-     -- added TON/NPI config options (the ones right above the resetinterval option)
-     -- added answeronpolarityswitch option
-     -- added "never" for resetinterval
-   * extensions
-     -- allow '*' when including files (#include "sip-*.conf")
-     -- added eswitch
-   * General
-     -- added #exec syntax for including output from a command
-     -- added show features CLI command
-     -- added configuration templates for category inheritance