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

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Steve Murphy
2006-10-05 15:22:37 +00:00
parent 743097a6c1
commit 80e8dd637b
1 changed files with 340 additions and 22 deletions
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@@ -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: