[svn-commits] kpfleming: branch 1.4 r44627 - /branches/1.4/CHANGES
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Fri Oct 6 14:07:44 MST 2006
Author: kpfleming
Date: Fri Oct 6 16:07:44 2006
New Revision: 44627
URL: http://svn.digium.com/view/asterisk?rev=44627&view=rev
Log:
various cleanups
Modified:
branches/1.4/CHANGES
Modified: branches/1.4/CHANGES
URL: http://svn.digium.com/view/asterisk/branches/1.4/CHANGES?rev=44627&r1=44626&r2=44627&view=diff
==============================================================================
--- branches/1.4/CHANGES (original)
+++ branches/1.4/CHANGES Fri Oct 6 16:07:44 2006
@@ -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 @@
* 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 @@
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 @@
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 @@
9. GLOBAL()
10. IFTIME()
11. KEYPADHASH()
- 12. ODBC interface;
+ 12. ODBC()
13. QUOTE()
14. RAND()
15. REALTIME()
@@ -159,7 +156,7 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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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