I put the accumulated changes from the commit logs and inspection, into CHANGES. Hope everyone approves\!

git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44466 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 340 insertions, 22 deletions

diff --git a/CHANGES b/CHANGES
index 49e14a4fa..da3b23bfe 100644
--- a/CHANGES
+++ b/CHANGES
@@ -1,27 +1,345 @@
-Changes since Asterisk 1.2.0-beta2:
-
-   * Cygwin build system portability
-   * Optional generation of outbound silence during channel recording
-
 Changes since Asterisk 1.2.0-beta1:
 
-   * Many, many bug fixes
-   * Documentation and sample configuration updates
-   * Vastly improved presence/subscription support in the SIP channel driver
-   * A new (experimental) mISDN channel driver
-   * A new monitoring application (MixMonitor)
-   * More portability fixes for non-Linux platforms
-   * New dialplan functions replacing old applications
-   * Significant deadlock and performance upgrades for the Manager interface
-   * An upgrade to the 'new' dialplan expression parser for all users
-   * New Zaptel echo cancellers with improved performance
-   * Support for the latest OSP toolkit from TransNexus
-   * Support user-controlled volume adjustment in MeetMe application
-   * More dialplan applications now return status variables instead of priority jumping
-   * Much more powerful ENUM support in the dialplan
-   * SIP domain support for authentication and virtual hosting
-   * Many PRI protocol updates and fixes, including more complete Q.SIG support
-   * New applications: Pickup() and Page()
+    * over 4,000 commits since 1.2
+    * queue member naming
+    * CLI commands rework
+          o Change the way CLI commands are structured.
+          o Most commands are now <module> <verb> <args>
+    * chan_h323 update
+    * multi-parking
+    * RTP packetization
+    * SLA (Shared Line Appearance) support various apps (meetme, etc).
+    * T.38 Passthrough Support for faxing
+    * Generic channel jitterbuffer (spawned from RTP)
+    * VLDTMF for better DTMF compatibility
+    * Improved chan_iax2 scalability
+    * AEL2 has replaced the original implementation of AEL. The "2" is removed. For more details, 
+      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
+          o also for asterisk-addons
+    * Radius CDR logging
+    * SNMP support
+    * STUN support in SIP
+    * SMDI (Simplified Message Desk Interface) support
+    * Manager over http
+    * Significant chan_skinny updates
+    * Significant chan_misdn updates
+    * 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
+    * Updates to the Radio Repeater app code
+    * deprecation of agentcallbacklogin
+    * uClibc builds supported
+    * work done for cygwin portability
+    * work done for freeBSD portability
+    * a lot of 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.
+    * 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
+    * 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").
+    * New Apps:
+         1. AMD() ;; Answering Machine Detection
+         2. ChannelRedirect() ;; asynch goto, redirect chan to context/exten/priority
+         3. ContinueWhile() ;; Addition to the While() suite. Acts like "continue".
+         4. ExitWhile() ;; Addition to the While() suite. Acts like "break".
+         5. ExtenSpy() ;; A close cousin to ChanSpy().
+         6. FollowMe() ;; findme/followme call redirect app
+         7. Log() ;; Send a message to the log, based on severity level.
+         8. MacroExclusive() ;; No more than one invocation of this macro allowed at any one time.
+         9. MorseCode() ;; turns strings into dits and dahs. A playground for ham radio licensees!
+        10. OSPAuth() ;; OSP authentication
+        11. QueueLog() ;; allows you to write your own events into the queue log
+        12. SLAStation() ;; Shared Line Appearance
+        13. SLATrunk() ;; Shared Line Appearance
+        14. SpeechCreate() ;; Voice Recognition Engine interface...
+        15. SpeechActivateGrammar()
+        16. SpeechStart()
+        17. SpeechBackground
+        18. SpeechDeactivateGrammar()
+        19. SpeechProcessingSound()
+        20. SpeechDestroy()
+        21. SpeechLoadGrammar()
+        22. SpeechUnloadGrammar()
+        23. StopMixMonitor() ;; to stop the MixMonitor App.
+        24. TryExec() ;; execute dialplan app without fatal consequences
+    * Apps removed:
+         1. CheckGroup -- do a comparison to ${GROUP()}
+         2. Curl -- use the function CURL() instead
+         3. Cut -- use the function CUT() instead
+         4. DateTime -- use sayunixtime() app instead.
+         5. DBget -- deprecated in 1.2, now removed.
+         6. DBput -- deprecated in 1.2, now removed.
+         7. Enumlookup -- use the function ENUMLOOKUP() instead
+         8. Eval -- use the function EVAL() instead
+         9. GetGroupCount -- use the function GROUP_COUNT() instead
+        10. GetGroupMatchCount -- use the function GROUP_MATCH_COUNT() instead
+        11. Intercom -- use the chan_oss module instead
+        12. Math -- use the function MATH() instead
+        13. MD5 -- use the function MD5() instead
+        14. SetCIDname -- use the function CALLERID(name) instead
+        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 ??
+        19. Txtcidname -- use the function TXTCIDNAME instead
+    * New Funcs:
+         1. ARRAY()
+         2. BASE_64_DECODE()
+         3. BASE_64_ENCODE()
+         4. CHANNEL()
+         5. CURL()
+         6. CUT()
+         7. DB_DELETE()
+         8. FILTER()
+         9. GLOBAL()
+        10. IFTIME()
+        11. KEYPADHASH()
+        12. ODBC interface;
+        13. QUOTE()
+        14. RAND()
+        15. REALTIME()
+        16. SHA1()
+        17. SORT()
+        18. SPRINTF()
+        19. SQL_ESC()
+        20. STAT()
+        21. STRPTIME()
+    * Apps that have changes to their interface:
+         1. Authenticate() -- optional maxdigits argument added.
+         2. ChanSpy() -- new options:
+                o w -- Enable 'whisper' mode, so the spying channel can talk to...
+                o W -- Enable 'private whisper' mode, so the spying channel can...
+         3. DBdel() -- now marked as DEPRECATED in favor of the DB_DELETE func
+         4. Dial()
+                o New Option: O([x]) for Zaptel operator mode
+                o New Option: K/k parking via dtmf tones
+         5. Dictate() -- optional filename argument added.
+         6. Directory() -- new option: e - In addition to the name, also read the extension number...
+         7. Meetme() -- new options:
+                o 'I' -- announce user join/leave without review
+                o 'l' -- set listen only mode (Listen only, no talking)
+                o 'o' -- set talker optimization - treats talkers who aren't speaking as...
+                o '1' -- do not play message when first person enters
+         8. MeetmeAdmin() -- new options:
+                o 'r' -- Reset one user's volume settings
+                o 'R' -- Reset all users volume settings
+                o 's' -- Lower entire conference speaking volume
+                o 'S' -- Raise entire conference speaking volume
+                o 't' -- Lower one user's talk volume
+                o 'T' -- Lower all users talk volume
+                o 'u' -- Lower one user's listen volume
+                o 'U' -- Lower all users listen volume
+                o 'v' -- Lower entire conference listening volume
+                o 'V' -- Raise entire conference listening volume
+         9. OSPFinish() : now also can return ERROR result.
+        10. OSPLookup() : Sets more variables, also now returns ERROR result.
+        11. Page() -- New option: r - record the page into a file (see 'r' for app_meetme)
+        12. Pickup() -- multiple extensions, PICKUPMARK; read the description!
+        13. Queue()
+                o New Argument: AGI
+                o New option: i
+        14. Random() -- is now deprecated in 1.4
+        15. Read() -- replace 'skip' and 'noanswer' options with 's', 'n', add 'i' option.
+        16. Record() -- New option: 'x' : ignore all terminator keys (DTMF) and keep recording until hangup
+        17. UserEvent() -- slight change in behavior. Read the description.
+        18. VoiceMailMain() -- new a(#) option, goes to folder # directly.
+        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.
+         3. MUSICCLASS -- Deprecated. Use CHANNEL(musicclass) instead.
+    * Config File Changes:
+         1. NEW config files:
+               1. amd.conf -- Answering Machine Detection parameters
+               2. followme.conf -- parameters for the findme/followme call forwarding
+               3. func_odbc.conf -- define sql access functions here
+               4. gtalk.conf -- how to handle gtalk protocol calls
+               5. h323.conf -- h323 configuration
+               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...
+              13. sla.conf -- Shared Line Appearance config
+              14. smdi.conf -- SMDI messaging config
+              15. udptl.conf -- T38's udptl transport config
+              16. users.conf -- user config
+         2. Changes to Existing Config files:
+               1. In General:
+                      o Jitterbuffer support added to several channels. Usually adds these variables to a config file:
+                           1. jbenable
+                           2. jbmaxsize
+                           3. jbresyncthreshold
+                           4. jbimpl
+                           5. jblog
+                      o MusicOnHold upgrade introduces two new variables:
+                           1. mohinterpret
+                           2. mohsuggest
+               2. agents.conf
+                      o maxlogintries variable added
+                      o autologoffunavail variable added
+                      o endcall variable added
+                      o agentgoodbye variable added
+                      o createlink variable REMOVED
+               3. alsa.conf
+                      o mohinterpret variable added
+                      o Jitterbuffer variables added
+               4. cdr.conf
+                      o endbeforehexten variable added
+                      o sections for csv and radius added, with variables usegmtime, loguniqueid, 
+                        loguserfield, and radiuscfg variables.
+               5. cdr_tds.conf
+                      o table variable addedextensions.ael
+               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)
+                      o parkedmusicclass
+                      o atxfernoanswertimeout variable added
+                      o parkcall variable added (one step parking)
+                      o improved documentation for dynamic feature declarations!
+               9. iax.conf
+                      o adsi variable added
+                      o mohinterpret variable added
+                      o mohsuggest variable added
+                      o jitterbuffer updates
+                      o iaxthreadcount variable added
+                      o iaxmaxthreadcount variable added
+                      o the way to specify TOS has changed.
+                      o mailboxdetail variable has been REMOVED.
+              10. indications.conf
+                      o [bg] entry added (Bulgaria).
+                      o [il] entry added (Israel)
+                      o [in] entry added (India)
+                      o [jp] entry added (Japan)
+                      o [my] entry added (Malaysia)
+                      o [th] entry added (Thailand)
+              11. manager.conf
+                      o displaysystemname variable added
+                      o webenabled variable added
+                      o httptimeout variable added
+                      o timestampevents variable added
+              12. mgcp.conf
+                      o Jitterbuffer support added
+              13. misdn.conf
+                      o l1watcher_timeout variable added
+                      o pp_l2_check variable added
+                      o echocancelwhenbridged variable added
+                      o echotraining variable added
+                      o max_incoming variable added
+                      o max_outgoing variable added
+              14. modules.conf
+                      o a comment for preloading res_speech.so is added
+                      o mention of global symbols is removed
+                      o obsolesced entries for chan_modem_* and app_intercom have been removed
+              15. musiconhold.conf
+                      o the default is now to do native moh from /var/lib/asterisk/moh
+              16. osp.conf
+                      o authpolicy variable added
+              17. oss.conf
+                      o debug variable added
+                      o device variable added
+                      o mixer variable added
+                      o boost variable added
+                      o callerid variable added
+                      o autohangup variable added
+                      o queuesize variable added
+                      o frags variable added
+                      o JitterBuffer support
+                      o sections to define alternate sound cards
+              18. queues.conf
+                      o autofill variable added
+                      o monitor-type variable added
+                      o musiconhold is now musicclass, with a difference in interpretation
+                      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
+              20. rpt.conf
+                      o duplex variable added
+                      o tailmessagetime variable added
+                      o tailsquashedtime variable added
+                      o tailmessages variable added
+              21. rtp.conf
+                      o rtcpinterval varaible added
+              22. sip.conf
+                      o allowoverlap variable added
+                      o allowtransfer variable added
+                      o tos variable REMOVED
+                      o tos_sip variable added
+                      o tos_audio variable added
+                      o tos_video variable added
+                      o minexpiry variable added
+                      o t1min variable added
+                      o musicclass variable REMOVED
+                      o mohinterpret variable added
+                      o mohmaxcallbitratesuggest variable added
+                      o allowsubscribe variable added
+                      o videosupport variable added
+                      o maxcallbitrate variable added
+                      o g726nonstandard variable added
+                      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 rtsavesysname variable added
+                      o JitterBuffer support added
+              23. skinny.conf
+                      o port variable renamed to bindport
+                      o JitterBuffer support added
+                      o model variable REMOVED
+                      o mohinterpret variable added
+                      o mohsuggest variable added
+                      o speeddial variable added
+                      o addon variable added
+              24. voicemail.conf
+                      o userscontext variable added
+                      o smdiport variable added
+                      o attachfmt variable added
+                      o volgain variable added
+                      o tempgreetwarn variable added
+              25. zapata.conf
+                      o pritimer variable has improved documentation
+                      o New signalling method: fgccama
+                      o New signalling method: fgccamamf
+                      o outsignalling variable added
+                      o distinctiveringaftercid variable added
+                      o cidsignalling now also accepts v23_jp, and smdi
+                      o usesmdi variable added
+                      o smdiport variable added
+                      o mohinterpret variable added
+                      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
+         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: