-- Use Q.931 standard cause codes for asterisk cause codes -- Bug fixes from the bug tracker Asterisk 1.0-RC2 -- Additional CDR backends -- Allow muted to reconnect -- Call parking improvements (including SIP parking support) -- Added licensed hold music from FreePlayMusic -- GR-303 and Zap improvements -- More bug fixes from the bug tracker -- Improved FreeBSD/OpenBSD/MacOS X support Asterisk 1.0-RC1 -- Innumerable bug fixes and features from the bug tracker -- Added Open Settlement Protocol (OSP) support -- Added Non-facility Associated Signalling (NFAS) Support -- Added alarm Monitoring support -- Added new MeetMe options -- Added GR-303 Support -- Added trunk groups -- ADPCM Standardization -- Numerous bug fixes -- Add IAX2 Firmware Support -- Add G.726 support -- Add ices/icecast support -- Numerous bug fixes Asterisk 0.7.2 -- Countless small bug fixes from bug tracker -- DSP Fixes -- Fix unloading of Zaptel -- Pass Caller*ID/ANI properly on call forwarding -- Add indication for Italy Asterisk 0.7.1 -- Fixed timed include context's and GotoIfTime -- Fixed chan_h323 it now gets remote ip properly instead of 127.0.0.1 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