254 lines
10 KiB
Plaintext
Executable File
254 lines
10 KiB
Plaintext
Executable File
-- Add experimental "debug channel" command
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-- Add 'C' flag to dial command to reset call detail record (handy for calling cards)
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-- Add NAT and dynamic support to MGCP
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-- Allow selection of in-band, out-of-band, or INFO based DTMF
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-- Add contributed "*80" support to blacklist numbers (Thanks James!)
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-- Add "NAT" option to sip user, peer, friend
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-- Add experimental "IAX2" protocol
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-- Add "Enhanced" AGI with audio pass-through (voice recognition anyone?)
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-- Choose best priority from codec from allow/disallow
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-- Reject SIP calls to self
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-- Allow SIP registration to provide an alternative contact
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-- Make HOLD on SIP make use of asterisk MOH
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-- Add supervised transfer (tested with Pingtel only)
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-- Allow maxexpirey and defaultexpirey to be runtime configurable for SIP
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-- Preliminary codec 13 support (RFC3389)
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-- Add app_authenticate for general purpose authentication
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-- Optimize RTP and smoother
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-- Create special variable "EXTEN-n" where it is extension stripped by n MSD
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-- Fix uninitialized frame pointer in channel.c
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-- Add global variables support under [globals] of extensions.conf
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-- Add macro support (show application Macro)
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-- Allow [123-5] etc in extensions
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-- Allow format of App(arg1,arg2,...) instead of just App,arg1|arg2 in dialplan
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-- Add message waiting indicator to SIP
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-- Fix double free bug in channel.c
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Asterisk 0.3.0
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-- Add fastfoward, rewind, seek, and truncate functions to streams
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-- Support registration
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-- Add G729 format
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-- Permit applications to return a digit indicating new extension
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-- Change "SHUTDOWN" to "STOP" in commands
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-- SIP "Hold" fixes and VXML URI support
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-- New chan_zap with 160 sample chunk size
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-- Add DTMF, MF, and Fax tone detector to dsp routines
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-- Allow overlap dialing (inbound) on PRI
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-- Enable tone detection with PRI
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-- Add special information tone detection
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-- Add Asterisk DB support
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-- Add pulse dialing
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-- Re-record all system prompts
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-- Change "timelen" to samples for better accuracy
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-- Move to editline, eliminating readline dependency
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-- Add peer "poke" support to SIP and IAX
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-- Add experimental call progress detection
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-- Add SIP authentication (digest)
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-- Add RDNIS
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-- Reroute faxes to "fax" extension
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-- Create ISDN/modem group concept
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-- Centralize indication
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-- Add initial MGCP support
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-- SIP debugging cleanup
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-- SIP reload
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-- SIP commands (show channels, etc)
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-- Add optional busy detection
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-- Add Visual Message Waiting Indicator (MDMF and SDMF)
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-- Add ambiguous extension matching
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-- Add *69
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-- Major SIP enhancements from SIPit
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-- Rewrite of ZAP CLASS features using subchannels
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-- Enhanced call parking
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-- Add extended outgoing spool support (pbx_spool)
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Asterisk 0.2.0
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-- Outbound origination API
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-- Call management improvements
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-- Add Do Not Disturb (*78, *79)
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-- Add agents
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-- Document variables
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-- Add transfer capability on the console
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-- Add SpeeX codec translator
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-- Add call queues
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-- Add setcallerid functionality (AGI, application)
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-- Add special variables ${CALLERID}, ${EXTEN}, ${CONTEXT}, ${PRIORITY}
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-- Don't echo cancel on pure TDM connections by default
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-- Implement Async GOTO
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-- Differentiate softhangups
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-- Add date/time
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Asterisk 0.1.12
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-- Fix for Big Endian machines
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-- MySQL CDR Engine
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-- Various SIP fixes and enhancements
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-- Add "zapateller application and arbitrary tone pairs
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-- Don't always start at "s"
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-- Separate linear mode for pseudo and real
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-- Add initial RTP and SIP support (no jitter buffer yet, unknown stability)
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-- Add 'h' extension, executed on hangup
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-- Add duration timer to message info
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-- Add web based voicemail checking ("make webvmail")
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-- Add ast_queue_frame function and eliminate frame pipes in most drivers
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-- Centralize host access (and possibly future ACL's)
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-- Add Caller*ID on PhoneJack (Thanks Nathan)
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-- Add "safe_asterisk" wrapper script to auto-restart Asterisk
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-- Indicate ringback on chan_phone
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-- Add answer confirmation (press '#' to confirm answer)
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-- Add distinctive ring support (e.g. Dial,Zap/4r2)
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-- Add ANSI/vt100 color support
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-- Make parking configurable through parking.conf
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-- Fix the empty voicemail problem
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-- Add Music On Hold
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-- Add ADSI Compiler (app_adsiprog)
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-- Extensive DISA re-work to improve tone generation
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-- Reset all idle channels every 10 minutes on a PRI
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-- Reset channels which are hungup with "channel in use"
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-- Implement VNAK support in chan_iax
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-- Fix chan_oss to support proper hangups and autoanswer
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-- Make shutdown properly hangup channels
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-- Add idling capability to chan_zap for idle-net
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-- Add "MeetMe" conferencing app (app_meetme)
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-- Add timing information to include
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Asterisk 0.1.11
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-- Add ISDN RAS capability
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-- Add stutter dialtone to Chan Zap
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-- Add "#include" capability to config files.
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-- Add call-forward variable to Chan Zap (*72, *73)
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-- Optimize IAX flow when transfer isn't possible
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-- Allow transmission of ANI over IAX
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Asterisk 0.1.10
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-- Make ast_readstring parameter be the max # of digits, not the max size with \0
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-- Make up any missing messages on the fly
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-- Add support for specific DTMF interruption to saying numbers
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-- Add new "u" and "b" options to condense busy/unavail handling
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-- Add support for RSA authentication on IAX calls
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-- Add support for ADSI compatible CPE
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-- Outgoing call queue
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-- Remote dialplan fixes for Quicknet
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-- Added AGI commands supporting TDD functions (RECEIVE CHAR & TDD MODE)
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-- Added TDD support (send/receive text in chan_zap)
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-- Fix all strncpy references
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-- Implement CSV CDR backend
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-- Implement Call Detail Records
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Asterisk 0.1.9
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-- Implement IAX quelching
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-- Allow Caller*ID to be overridden and suggested
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-- Configure defaults to use IAXTEL
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-- Allow remote dialplan polling via IAX
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-- Eliminate ast_longest_extension
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-- Implement dialplan request/reply
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-- Let peers have allow/disallow for codecs
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-- Change allow/deny to permit/deny in IAX
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-- Allow dialplan entries to match Caller*ID as well
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-- Added AGI (Asterisk Gateway Interface) scripting interface (app_agi)
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-- Added chan_zap for zapata telephony kernel interface, removed chan_tor
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-- Add convenience functions
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-- Fix race condition in channel hangup
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-- Fix memory leaks in both asterisk and iax frame allocations
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-- Add "iax show stats" command and -DTRACE_FRAMES (for frame tracing)
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-- Add DISA application (Thanks to Jim Dixon)
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-- Add IAX transfer support
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-- Add URL and HTML transmission
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-- Add application for sending images
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-- Add RedHat RPM spec file and build capability
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-- Fix GSM WAV file format bug
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-- Move ignorepat to main dialplan
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-- Add ability to specificy TOS bits in IAX
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-- Allow username:password in IAX strings
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-- Updates to PhoneJack interface
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-- Allow "servermail" in voicemail.conf to override e-mail in "from" line
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-- Add 'skip' option to app_playback
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-- Reject IAX calls on unknown extensions
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-- Fix version stuff
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Asterisk 0.1.8
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-- Keep track of version information
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-- Add -f to cause Asterisk not to fork
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-- Keep important information in voicemail .txt file
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-- Adtran Voice over Frame Relay updates
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-- Implement option setting/querying of channel drivers
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-- IAX performance improvements and protocol fixes
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-- Substantial enhancement of console channel driver
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-- Add IAX registration. Now IAX can dynamically register
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-- Add flash-hook transfer on tormenta channels
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-- Added Three Way Calling on tormenta channels
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-- Start on concept of zombie channel
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-- Add Call Waiting CallerID
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-- Keep track of who registeres contexts, includes, and extensions
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-- Added Call Waiting(tm), *67, *70, and *82 codes
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-- Move parked calls into "parkedcalls" context by default
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-- Allow dialplan to be displayed
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-- Allow "=>" instead of just "=" to make instantiation clearer
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-- Asterisk forks if called with no arguments
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-- Add remote control by running asterisk -vvvc
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-- Adjust verboseness with "set verbose" now
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-- No longer requires libaudiofile
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-- Install beep
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-- Make PBX Config module reload extensions on SIGHUP
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-- Allow modules to be reloaded when SIGHUP is received
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-- Variables now contain line numbers
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-- Make dialer send in band signalling
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-- Add record application
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-- Added PRI signalling to Tormenta driver
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-- Allow use of BYEXTENSION in "Goto"
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-- Allow adjustment of gains on tormenta channels
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-- Added raw PCM file format support
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-- Add U-law translator
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-- Fix DTMF handling in bridge code
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-- Fix access control with IAX
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* Asterisk 0.1.7
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-- Update configuration files and add some missing sounds
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-- Added ability to include one context in another
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-- Rewrite of PBX switching
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-- Major mods to dialler application
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-- Added Caller*ID spill reception
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-- Added Dialogic VOX file format support
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-- Added ADPCM Codec
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-- Add Tormenta driver (RBS signalling)
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-- Add Caller*ID spill creation
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-- Rewrite of translation layer entirely
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-- Add ability to run PBX without additional thread
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* Asterisk 0.1.6
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-- Make app_dial handle a lack of translators smoothly
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-- Add ISDN4Linux support -- dtmf is weird...
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-- Minor bug fixes
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* Asterisk 0.1.5
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-- Fix a small mistake in IAX
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-- Fix the QuickNet driver to work with newer cards
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* Asterisk 0.1.4
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-- Update VoFR some more
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-- Fix the QuickNet driver to work with LineJack
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-- Add ability to pass images for IAX.
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* Asterisk 0.1.3
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-- Update VoFR for latest sangoma code
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-- Update QuickNet Driver
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-- Add text message handling
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-- Fix transfers to use "default" if not in current context
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-- Add call parking
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-- Improve format/content negotiation
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-- Added support for multiple languages
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-- Bug fixes, as always...
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* Asterisk 0.1.2
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-- Updated README file with a "Getting Started" section
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-- Added sample sounds and configuration files.
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-- Added LPC10 very low bandwidth (low quality) compression
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-- Enhanced translation selection mechanism.
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-- Enhanced IAX jitter buffer, improved reliability
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-- Support echo cancelation on PhoneJack
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-- Updated PhoneJack driver to std. Telephony interface
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-- Added app_echo for evaluating VoIP latency
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-- Added app_system to execute arbitrary programs
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-- Updated sample configuration files
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-- Added OSS channel driver (full duplex only)
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-- Added IAX implementation
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-- Fixed some deadlocks.
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-- A whole bunch of bug fixes
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* Asterisk 0.1.1
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-- Revised translator, fixed some general race conditions throughout *
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-- Made dialer somewhat more aware of incompatible voice channels
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-- Added Voice Modem driver and A/Open Modem Driver stub
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-- Added MP3 decoder channel
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-- Added Microsoft WAV49 support
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-- Revised License -- Pure GPL, nothing else
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-- Modified Copyright statement since code is still currently owned by author
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-- Added RAW GSM headerless data format
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-- Innumerable bug fixes
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* Asterisk 0.1.0
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-- Initial Release
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