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asterisk
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various cleanups

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.4
Kevin P. Fleming 19 years ago
parent 2d3da5b500
commit 0f29a4ddd3
1 changed files with 24 additions and 135 deletions
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@ -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

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