Asterisk 0.7.0
 -- Removed MP3 format and codec
 -- Can now load and unload SIP,IAX,IAX2,H323 channels without core
 -- Fixed various compiler warnings and clean up source tree
 -- Preliminary AES Support
 -- Fix SIP REINVITE
 -- Outbound SIP registration behind NAT using externip
 -- More CLI documentation and clean up
 -- Pin numbers on MeeMe
 -- Dynamic MeetMe conferences are more consistent with static conferences
 -- Added channel variables ${HANGUPCAUSE}, ${SIPDOMAIN}, ${TIMESTAMP}, ${ACCONTCODE}
 -- ODBC support for logging CDRs
 -- Indications for Norway and New Zeland
 -- Major redesign of app_voicemail
 -- Syslog support
 -- Reload logfiles with CLI command 'logger reload' and rotate logs with "logger rotate'
 -- Configurable DEBUG, NOTICE, WARNING, ERROR and ast_verbose messages now appear on remote console 
 -- Properly reaping any zombie processes
 -- Added applications SayUnixTime, SetCDRUserField, HasNewVoicemail, ZapScan, Random, ResetCDR, NoCDR
 -- Make PRI Hangup Cause available to the dialplan
 -- Verify included contexts in extensions.conf
 -- Add DESTDIR support for building RPMs and packages
 -- Do route lookups on OpenBSD
 -- Add support for building on FreeBSD and OS X
 -- Add support for PostgreSQL in Voicemail
 -- Translate SIP hangup cause to PRI hangup cause where needed
 -- Better support for MOH in IAX2
 -- Fix SIP problem where channels were not removed on BYE
 -- Display codecs by name
 -- Remove MySQL and put PGSql instead for licensing reasons
 -- Better capability matching in SIP
 -- Full IBR4 compliance for chan_zap
 -- More flexible CDR handling
 -- Distinguish between BUSY and FAILURE on outbound calls
 -- Add initial support for SCCP via chan_skinny
 -- Better support for Future Group B signaling
Asterisk 0.5.0
 -- Retain IAX2 and SIP registrations past shutdown/crash and restart
 -- True data mode bridging when possible
 -- H.323 build improvements
 -- Agent Callback-login support
 -- RFC2833 Improvements
 -- Add thread debugging
 -- Add optional pedantic SIP checking for Pingtel
 -- Allow extension names, include context, switch to use global vars.
 -- Allow variables in extensions.conf to reference previously defined ones
 -- Merge voicemail enhancements (app_voicemail2)
 -- Add multiple queueing strategies
 -- Merge support for 'T'
 -- Allow pending agent calling (Agent/:1)
 -- Add groupings to agents.conf
 -- Add video support to IAX2
 -- Zaptel optimize playback
 -- Add video support to SIP
 -- Make RTP ports configurable
 -- Add RDNIS support to SIP and IAX2
 -- Add transfer app (implement in SIP and IAX2)
 -- Make voicemail segmentable by context (app_voicemail2)
 -- Major restructuring of voicemail (app_voicemail2)
 -- Add initial ENUM support
 -- Add malloc debugging support
 -- Add preliminary Voicetronix support
 -- Add iLBC codec
Asterisk 0.4.0
 -- Merge and edit Nick's FXO dial support
 -- Reengineer SIP registration (outbound)
 -- Support call pickup on SIP and compatibly with ZAP
 -- Support 302 Redirect on SIP
 -- Management interface improvements
 -- Add "hint" support
 -- Improve call forwarding using new "Local" channel driver.
 -- Add "Local" channel
 -- Substantial SIP enhancements including retransmissions
 -- Enforce case sensitivity on extension/context names
 -- Add monitor support (Thanks, Mahmut)
 -- Add experimental "trunk" option to IAX2 for high density VoIP
 -- Add experimental "debug channel" command
 -- Add 'C' flag to dial command to reset call detail record (handy for calling cards)
 -- Add NAT and dynamic support to MGCP
 -- Allow selection of in-band, out-of-band, or INFO based DTMF
 -- Add contributed "*80" support to blacklist numbers (Thanks James!)
 -- Add "NAT" option to sip user, peer, friend
 -- Add experimental "IAX2" protocol
 -- Change special variable "EXTEN-n" to "EXTEN:n" to follow Bash syntax
 -- Add "Enhanced" AGI with audio pass-through (voice recognition anyone?)
 -- Choose best priority from codec from allow/disallow
 -- Reject SIP calls to self
 -- Allow SIP registration to provide an alternative contact
 -- Make HOLD on SIP make use of asterisk MOH
 -- Add supervised transfer (tested with Pingtel only)
 -- Allow maxexpirey and defaultexpirey to be runtime configurable for SIP
 -- Preliminary codec 13 support (RFC3389)
 -- Add app_authenticate for general purpose authentication
 -- Optimize RTP and smoother
 -- Create special variable "EXTEN-n" where it is extension stripped by n MSD
 -- Fix uninitialized frame pointer in channel.c
 -- Add global variables support under [globals] of extensions.conf
 -- Add macro support (show application Macro)
 -- Allow [123-5] etc in extensions
 -- Allow format of App(arg1,arg2,...) instead of just App,arg1|arg2 in dialplan
 -- Add message waiting indicator to SIP
 -- Fix double free bug in channel.c
Asterisk 0.3.0
 -- Add fastfoward, rewind, seek, and truncate functions to streams
 -- Support registration
 -- Add G729 format
 -- Permit applications to return a digit indicating new extension
 -- Change "SHUTDOWN" to "STOP" in commands
 -- SIP "Hold" fixes and VXML URI support
 -- New chan_zap with 160 sample chunk size
 -- Add DTMF, MF, and Fax tone detector to dsp routines
 -- Allow overlap dialing (inbound) on PRI
 -- Enable tone detection with PRI
 -- Add special information tone detection
 -- Add Asterisk DB support
 -- Add pulse dialing
 -- Re-record all system prompts
 -- Change "timelen" to samples for better accuracy
 -- Move to editline, eliminating readline dependency
 -- Add peer "poke" support to SIP and IAX
 -- Add experimental call progress detection
 -- Add SIP authentication (digest)
 -- Add RDNIS
 -- Reroute faxes to "fax" extension
 -- Create ISDN/modem group concept
 -- Centralize indication
 -- Add initial MGCP support
 -- SIP debugging cleanup
 -- SIP reload
 -- SIP commands (show channels, etc)
 -- Add optional busy detection
 -- Add Visual Message Waiting Indicator (MDMF and SDMF)
 -- Add ambiguous extension matching
 -- Add *69
 -- Major SIP enhancements from SIPit
 -- Rewrite of ZAP CLASS features using subchannels
 -- Enhanced call parking
 -- Add extended outgoing spool support (pbx_spool)
Asterisk 0.2.0
 -- Outbound origination API
 -- Call management improvements
 -- Add Do Not Disturb (*78, *79)
 -- Add agents
 -- Document variables
 -- Add transfer capability on the console
 -- Add SpeeX codec translator
 -- Add call queues
 -- Add setcallerid functionality (AGI, application)
 -- Add special variables ${CALLERID}, ${EXTEN}, ${CONTEXT}, ${PRIORITY}
 -- Don't echo cancel on pure TDM connections by default
 -- Implement Async GOTO
 -- Differentiate softhangups
 -- Add date/time
Asterisk 0.1.12
 -- Fix for Big Endian machines
 -- MySQL CDR Engine
 -- Various SIP fixes and enhancements
 -- Add "zapateller application and arbitrary tone pairs
 -- Don't always start at "s"
 -- Separate linear mode for pseudo and real
 -- Add initial RTP and SIP support (no jitter buffer yet, unknown stability)
 -- Add 'h' extension, executed on hangup
 -- Add duration timer to message info
 -- Add web based voicemail checking ("make webvmail")
 -- Add ast_queue_frame function and eliminate frame pipes in most drivers
 -- Centralize host access (and possibly future ACL's)
 -- Add Caller*ID on PhoneJack (Thanks Nathan)
 -- Add "safe_asterisk" wrapper script to auto-restart Asterisk
 -- Indicate ringback on chan_phone
 -- Add answer confirmation (press '#' to confirm answer)
 -- Add distinctive ring support (e.g. Dial,Zap/4r2)
 -- Add ANSI/vt100 color support
 -- Make parking configurable through parking.conf
 -- Fix the empty voicemail problem
 -- Add Music On Hold
 -- Add ADSI Compiler (app_adsiprog)
 -- Extensive DISA re-work to improve tone generation
 -- Reset all idle channels every 10 minutes on a PRI
 -- Reset channels which are hungup with "channel in use"
 -- Implement VNAK support in chan_iax
 -- Fix chan_oss to support proper hangups and autoanswer
 -- Make shutdown properly hangup channels
 -- Add idling capability to chan_zap for idle-net
 -- Add "MeetMe" conferencing app (app_meetme)
 -- Add timing information to include
Asterisk 0.1.11
 -- Add ISDN RAS capability
 -- Add stutter dialtone to Chan Zap
 -- Add "#include" capability to config files.
 -- Add call-forward variable to Chan Zap (*72, *73)
 -- Optimize IAX flow when transfer isn't possible
 -- Allow transmission of ANI over IAX
Asterisk 0.1.10
 -- Make ast_readstring parameter be the max # of digits, not the max size with \0
 -- Make up any missing messages on the fly
 -- Add support for specific DTMF interruption to saying numbers
 -- Add new "u" and "b" options to condense busy/unavail handling
 -- Add support for RSA authentication on IAX calls
 -- Add support for ADSI compatible CPE
 -- Outgoing call queue
 -- Remote dialplan fixes for Quicknet
 -- Added AGI commands supporting TDD functions (RECEIVE CHAR & TDD MODE)
 -- Added TDD support (send/receive text in chan_zap)
 -- Fix all strncpy references
 -- Implement CSV CDR backend
 -- Implement Call Detail Records
Asterisk 0.1.9
 -- Implement IAX quelching
 -- Allow Caller*ID to be overridden and suggested
 -- Configure defaults to use IAXTEL
 -- Allow remote dialplan polling via IAX
 -- Eliminate ast_longest_extension
 -- Implement dialplan request/reply
 -- Let peers have allow/disallow for codecs
 -- Change allow/deny to permit/deny in IAX
 -- Allow dialplan entries to match Caller*ID as well
 -- Added AGI (Asterisk Gateway Interface) scripting interface (app_agi)
 -- Added chan_zap for zapata telephony kernel interface, removed chan_tor
 -- Add convenience functions
 -- Fix race condition in channel hangup
 -- Fix memory leaks in both asterisk and iax frame allocations
 -- Add "iax show stats" command and -DTRACE_FRAMES (for frame tracing)
 -- Add DISA application (Thanks to Jim Dixon)
 -- Add IAX transfer support
 -- Add URL and HTML transmission
 -- Add application for sending images
 -- Add RedHat RPM spec file and build capability
 -- Fix GSM WAV file format bug
 -- Move ignorepat to main dialplan
 -- Add ability to specificy TOS bits in IAX
 -- Allow username:password in IAX strings
 -- Updates to PhoneJack interface
 -- Allow "servermail" in voicemail.conf to override e-mail in "from" line
 -- Add 'skip' option to app_playback
 -- Reject IAX calls on unknown extensions
 -- Fix version stuff
Asterisk 0.1.8
 -- Keep track of version information
 -- Add -f to cause Asterisk not to fork
 -- Keep important information in voicemail .txt file
 -- Adtran Voice over Frame Relay updates
 -- Implement option setting/querying of channel drivers
 -- IAX performance improvements and protocol fixes
 -- Substantial enhancement of console channel driver
 -- Add IAX registration.  Now IAX can dynamically register
 -- Add flash-hook transfer on tormenta channels
 -- Added Three Way Calling on tormenta channels
 -- Start on concept of zombie channel
 -- Add Call Waiting CallerID
 -- Keep track of who registeres contexts, includes, and extensions
 -- Added Call Waiting(tm), *67, *70, and *82 codes
 -- Move parked calls into "parkedcalls" context by default
 -- Allow dialplan to be displayed
 -- Allow "=>" instead of just "=" to make instantiation clearer
 -- Asterisk forks if called with no arguments
 -- Add remote control by running asterisk -vvvc
 -- Adjust verboseness with "set verbose" now
 -- No longer requires libaudiofile
 -- Install beep
 -- Make PBX Config module reload extensions on SIGHUP
 -- Allow modules to be reloaded when SIGHUP is received
 -- Variables now contain line numbers
 -- Make dialer send in band signalling
 -- Add record application
 -- Added PRI signalling to Tormenta driver
 -- Allow use of BYEXTENSION in "Goto"
 -- Allow adjustment of gains on tormenta channels
 -- Added raw PCM file format support
 -- Add U-law translator
 -- Fix DTMF handling in bridge code
 -- Fix access control with IAX
* Asterisk 0.1.7
 -- Update configuration files and add some missing sounds
 -- Added ability to include one context in another
 -- Rewrite of PBX switching
 -- Major mods to dialler application
 -- Added Caller*ID spill reception
 -- Added Dialogic VOX file format support
 -- Added ADPCM Codec
 -- Add Tormenta driver (RBS signalling)
 -- Add Caller*ID spill creation
 -- Rewrite of translation layer entirely
 -- Add ability to run PBX without additional thread
* Asterisk 0.1.6
 -- Make app_dial handle a lack of translators smoothly
 -- Add ISDN4Linux support -- dtmf is weird...
 -- Minor bug fixes
* Asterisk 0.1.5
 -- Fix a small mistake in IAX
 -- Fix the QuickNet driver to work with newer cards
* Asterisk 0.1.4
 -- Update VoFR some more
 -- Fix the QuickNet driver to work with LineJack
 -- Add ability to pass images for IAX.
* Asterisk 0.1.3
 -- Update VoFR for latest sangoma code
 -- Update QuickNet Driver
 -- Add text message handling
 -- Fix transfers to use "default" if not in current context
 -- Add call parking
 -- Improve format/content negotiation
 -- Added support for multiple languages
 -- Bug fixes, as always...
* Asterisk 0.1.2
 -- Updated README file with a "Getting Started" section
 -- Added sample sounds and configuration files.
 -- Added LPC10 very low bandwidth (low quality) compression
 -- Enhanced translation selection mechanism.
 -- Enhanced IAX jitter buffer, improved reliability
 -- Support echo cancelation on PhoneJack
 -- Updated PhoneJack driver to std. Telephony interface
 -- Added app_echo for evaluating VoIP latency
 -- Added app_system to execute arbitrary programs
 -- Updated sample configuration files
 -- Added OSS channel driver (full duplex only)
 -- Added IAX implementation
 -- Fixed some deadlocks.
 -- A whole bunch of bug fixes
* Asterisk 0.1.1
 -- Revised translator, fixed some general race conditions throughout *
 -- Made dialer somewhat more aware of incompatible voice channels
 -- Added Voice Modem driver and A/Open Modem Driver stub
 -- Added MP3 decoder channel
 -- Added Microsoft WAV49 support
 -- Revised License -- Pure GPL, nothing else
 -- Modified Copyright statement since code is still currently owned by author
 -- Added RAW GSM headerless data format
 -- Innumerable bug fixes
* Asterisk 0.1.0
 -- Initial Release
