diff options
author | murf <murf@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-10-05 15:22:37 +0000 |
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committer | murf <murf@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-10-05 15:22:37 +0000 |
commit | 5d3a32773c350fc0d3e6fcce54a7ddb685160814 (patch) | |
tree | 56c5a4d504e2d83e770678f552c0cc52389f3f06 /CHANGES | |
parent | ade967e271500422f0590c3d232cfc505b522696 (diff) |
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
Diffstat (limited to 'CHANGES')
-rw-r--r-- | CHANGES | 362 |
1 files changed, 340 insertions, 22 deletions
@@ -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: |