[svn-commits] murf: branch 1.4 r44466 - /branches/1.4/CHANGES
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Thu Oct 5 08:22:38 MST 2006
Author: murf
Date: Thu Oct 5 10:22:37 2006
New Revision: 44466
URL: http://svn.digium.com/view/asterisk?rev=44466&view=rev
Log:
I put the accumulated changes from the commit logs and inspection, into CHANGES. Hope everyone approves\!
Modified:
branches/1.4/CHANGES
Modified: branches/1.4/CHANGES
URL: http://svn.digium.com/view/asterisk/branches/1.4/CHANGES?rev=44466&r1=44465&r2=44466&view=diff
==============================================================================
--- branches/1.4/CHANGES (original)
+++ branches/1.4/CHANGES Thu Oct 5 10:22:37 2006
@@ -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:
More information about the svn-commits
mailing list