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asterisk/CHANGES
Kevin P. Fleming bf1486e917 Merged revisions 110869 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) | 2 lines

due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@110880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26 16:42:35 +00:00

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Changes since Asterisk 1.2:
* 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
* RTP packetization
* SLA (Shared Line Appearance) support
* T.38 Passthrough Support for faxing in SIP
* Generic channel jitterbuffer (spawned from RTP)
* Variable Length DTMF for better DTMF compatibility
* Improved chan_iax2 scalability by using multithreading
* 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
AEL is no longer considered experimental.
* New sounds; English, Spanish, and French prompts, as well as music on hold files, in
multiple Asterisk native formats.
* IMAP storage of voicemail
* Jabber/GoogleTalk integration
* 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
* SMDI (Simplified Message Desk Interface) support
* Redesign of MusicOnHold configuration settings
* Manager over HTTP
* Significant chan_skinny updates
* Significant chan_misdn updates
* Improved SIP transfers
* SIP MWI subscription support
* Much improved support for SIP video
* Control over SIP transfers and subscriptions (enable/disable per device)
* ChanSpy whisper mode (Whisper Paging)
* Configurable language support for saying dates and times
* Significant architecture improvements for memory usage and performance
* Media-only IAX2 transfers
* Updates to the Radio Repeater app code
* Deprecation of AgentCallbackLogin in favor of a dialplan-based solution
* uClibc builds supported
* 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%.
* 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
* Added support for Sigma Designs cards.
* Frame header caching to reduce memory allocation/freeing
* Passthrough and record/playback support for G.722 wideband audio
* 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 -- was deprecated in 1.2, now removed
19. Txtcidname -- use the function TXTCIDNAME instead
* New Dialplan Functions:
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()
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.
* Functions that have changes to their interfaces:
1. CDR -- new option: u
2. LANGUAGE -- Deprecated. Use CHANNEL(language) instead.
3. MUSICCLASS -- Deprecated. Use CHANNEL(musicclass) instead.
* Configuration 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
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 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 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 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 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 maxcallbitratesuggest 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'
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 distributable
2. chan_modem_* and related modules are gone because the kernel support for those interfaces is old, buggy and unsupported
* New Utils:
1. aelparse -- compile .ael files outside of asterisk
* New manager events:
1. OriginateResponse event comes to replace OriginateSuccess and OriginateFailure
* iLBC source code no longer included (see UPGRADE.txt for details)
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