From 0f29a4ddd3cc891bf78ce9c27e4efce35d00007a Mon Sep 17 00:00:00 2001 From: "Kevin P. Fleming" Date: Fri, 6 Oct 2006 21:07:44 +0000 Subject: [PATCH] various cleanups git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44627 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- CHANGES | 159 +++++++++----------------------------------------------- 1 file changed, 24 insertions(+), 135 deletions(-) diff --git a/CHANGES b/CHANGES index da3b23bfec..ddcc7baceb 100644 --- a/CHANGES +++ b/CHANGES @@ -1,4 +1,4 @@ -Changes since Asterisk 1.2.0-beta1: +Changes since Asterisk 1.2: * over 4,000 commits since 1.2 * queue member naming @@ -8,7 +8,7 @@ Changes since Asterisk 1.2.0-beta1: * chan_h323 update * multi-parking * RTP packetization - * SLA (Shared Line Appearance) support various apps (meetme, etc). + * SLA (Shared Line Appearance) support * T.38 Passthrough Support for faxing * Generic channel jitterbuffer (spawned from RTP) * VLDTMF for better DTMF compatibility @@ -17,39 +17,36 @@ Changes since Asterisk 1.2.0-beta1: 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 + * Jabber/Jingle/GoogleTalk + * 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 - * STUN support in SIP * SMDI (Simplified Message Desk Interface) support - * Manager over http + * Manager over HTTP * Significant chan_skinny updates * Significant chan_misdn updates - * improved SIP transfers - * ChanSpy whisper mode (whisper Paging) + * 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 + * Media-only IAX2 transfers * Updates to the Radio Repeater app code - * deprecation of agentcallbacklogin + * Deprecation of AgentCallbackLogin in favor of a dialplan-based solution * uClibc builds supported - * work done for cygwin portability - * work done for freeBSD portability - * a lot of work done for Solaris portability + * 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%. - * 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. + * 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 + * 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"). + * Frame header caching to reduce memory allocation/freeing + * 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 @@ -93,7 +90,7 @@ Changes since Asterisk 1.2.0-beta1: 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 ?? + 18. Sql_postgres -- was deprecated in 1.2, now removed 19. Txtcidname -- use the function TXTCIDNAME instead * New Funcs: 1. ARRAY() @@ -107,7 +104,7 @@ Changes since Asterisk 1.2.0-beta1: 9. GLOBAL() 10. IFTIME() 11. KEYPADHASH() - 12. ODBC interface; + 12. ODBC() 13. QUOTE() 14. RAND() 15. REALTIME() @@ -159,7 +156,7 @@ Changes since Asterisk 1.2.0-beta1: 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. + 2. LANGUAGE -- Deprecated. Use CHANNEL(language) instead. 3. MUSICCLASS -- Deprecated. Use CHANNEL(musicclass) instead. * Config File Changes: 1. NEW config files: @@ -171,7 +168,6 @@ Changes since Asterisk 1.2.0-beta1: 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... @@ -204,14 +200,12 @@ Changes since Asterisk 1.2.0-beta1: o sections for csv and radius added, with variables usegmtime, loguniqueid, loguserfield, and radiuscfg variables. 5. cdr_tds.conf - o table variable addedextensions.ael + 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 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) @@ -275,7 +269,6 @@ Changes since Asterisk 1.2.0-beta1: 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 @@ -297,7 +290,7 @@ Changes since Asterisk 1.2.0-beta1: o t1min variable added o musicclass variable REMOVED o mohinterpret variable added - o mohmaxcallbitratesuggest variable added + o maxcallbitratesuggest variable added o allowsubscribe variable added o videosupport variable added o maxcallbitrate variable added @@ -305,7 +298,7 @@ Changes since Asterisk 1.2.0-beta1: 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 canreinvite variable can also now be set to 'nonat' o rtsavesysname variable added o JitterBuffer support added 23. skinny.conf @@ -335,111 +328,7 @@ Changes since Asterisk 1.2.0-beta1: 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 + 1. codec_g723 was removed because the actual codec implementation it was designed to use is not distributable 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: - -This list currently only containts changes made from the end of November until -March 26, 2005. - - * Add new applications: - -- AgentMonitorOutgoing - -- Curl - -- ExecIf - -- ExecIfTime - -- IAX2Provision - -- MacroExit - -- MacroIf - -- PauseQueueMember - -- ReadFile - -- SetRDNIS - -- SIPAddHeader - -- SIPGetHeader - -- StartMusicOnHold - -- StopMusicOnHold - -- UnpauseQueueMember - -- WaitForSilence - -- While / EndWhile - * app Answer - -- added delay option - * app ChanIsAvail - -- added 's' option - * app Dial - -- add option to specify the class for musiconhold with m option - * app EnumLookup - -- added "reload enum" for configuration - * app Goto - -- added relative priorities - * app GotoIf - -- added relative priorities - * app MeetMe - -- added 'i' option - -- added 'r' option - -- added 'T' option - -- added 'P' option - -- added 'c' option - -- added adminpin to meetme.conf - -- added reload command - * app PrivacyManager - -- add config file privacy.conf - * app queue - -- queues.conf - -- added persistentmembers option to queues.conf - -- changed music option to musiconhold - -- added weight option - -- added note about why agent groups probably shouldn't be used - -- added timeoutrestart option - * app Read - -- added attempts parameter - -- added timeout parameter - * app Record - -- added 'q' option - * app SendDTMF - -- add timeout option - * app SMS - -- document alternative syntax for queueing messages - * app Voicemail - -- add info about VM_CATEGORY - -- voicemail.conf - -- added usedirectory option - -- added VM_CIDNUM and VM_CIDNAME in message config - * chan IAX2 - -- new jitterbuffer - -- added setvar option - -- added regex to iax2 show peers/users - -- allow multiple bindaddr lines in iax.conf - -- added reload command - -- added forcejitterbuffer option - -- added note about specifying bindport before bindaddr - -- added trunktimestamps option - * chan Agent - -- added agent logoff CLI command - * chan OSS - -- added Flash CLI command - * chan SIP - -- added setvar option - -- added compactheaders option - -- added usereqphone option - -- added registertimeout option - -- added externhost option - -- added sip notify CLI command - -- added sip_notify.conf - -- added allowguest option - * chan Zap - -- added hanguponplarityswitch option - -- added sendcalleridafter option - -- added priresetinterval option - -- added TON/NPI config options (the ones right above the resetinterval option) - -- added answeronpolarityswitch option - -- added "never" for resetinterval - * extensions - -- allow '*' when including files (#include "sip-*.conf") - -- added eswitch - * General - -- added #exec syntax for including output from a command - -- added show features CLI command - -- added configuration templates for category inheritance