1 -- Major revamp of PBX core including 'n' and 's' priorities and labels
2 -- Deprecate pbx_wilcalu and app_qcall in favor of pbx_spool
3 -- Remove old chan_iax and chan_vofr
4 -- Major Caller*ID Restructuring
6 -- Added AGI over TCP support
7 -- Add ability to purge callers from queue if no agents are logged in
8 -- Fix inband PRI indication detection
9 -- Fix for MGCP - always request digits if no RTP stream
10 -- Fixed seg fault for ast_control_streamfile
11 -- Make pick-up extension configurable via features.conf
12 -- Numerous other bug fixes
14 -- Use Q.931 standard cause codes for asterisk cause codes
15 -- Bug fixes from the bug tracker
17 -- Additional CDR backends
18 -- Allow muted to reconnect
19 -- Call parking improvements (including SIP parking support)
20 -- Added licensed hold music from FreePlayMusic
21 -- GR-303 and Zap improvements
22 -- More bug fixes from the bug tracker
23 -- Improved FreeBSD/OpenBSD/MacOS X support
25 -- Innumerable bug fixes and features from the bug tracker
26 -- Added Open Settlement Protocol (OSP) support
27 -- Added Non-facility Associated Signalling (NFAS) Support
28 -- Added alarm Monitoring support
29 -- Added new MeetMe options
30 -- Added GR-303 Support
32 -- ADPCM Standardization
34 -- Add IAX2 Firmware Support
36 -- Add ices/icecast support
39 -- Countless small bug fixes from bug tracker
41 -- Fix unloading of Zaptel
42 -- Pass Caller*ID/ANI properly on call forwarding
43 -- Add indication for Italy
45 -- Fixed timed include context's and GotoIfTime
46 -- Fixed chan_h323 it now gets remote ip properly instead of 127.0.0.1
48 -- Removed MP3 format and codec
49 -- Can now load and unload SIP,IAX,IAX2,H323 channels without core
50 -- Fixed various compiler warnings and clean up source tree
51 -- Preliminary AES Support
53 -- Outbound SIP registration behind NAT using externip
54 -- More CLI documentation and clean up
55 -- Pin numbers on MeeMe
56 -- Dynamic MeetMe conferences are more consistent with static conferences
57 -- Added channel variables ${HANGUPCAUSE}, ${SIPDOMAIN}, ${TIMESTAMP}, ${ACCONTCODE}
58 -- ODBC support for logging CDRs
59 -- Indications for Norway and New Zeland
60 -- Major redesign of app_voicemail
62 -- Reload logfiles with CLI command 'logger reload' and rotate logs with "logger rotate'
63 -- Configurable DEBUG, NOTICE, WARNING, ERROR and ast_verbose messages now appear on remote console
64 -- Properly reaping any zombie processes
65 -- Added applications SayUnixTime, SetCDRUserField, HasNewVoicemail, ZapScan, Random, ResetCDR, NoCDR
66 -- Make PRI Hangup Cause available to the dialplan
67 -- Verify included contexts in extensions.conf
68 -- Add DESTDIR support for building RPMs and packages
69 -- Do route lookups on OpenBSD
70 -- Add support for building on FreeBSD and OS X
71 -- Add support for PostgreSQL in Voicemail
72 -- Translate SIP hangup cause to PRI hangup cause where needed
73 -- Better support for MOH in IAX2
74 -- Fix SIP problem where channels were not removed on BYE
75 -- Display codecs by name
76 -- Remove MySQL and put PGSql instead for licensing reasons
77 -- Better capability matching in SIP
78 -- Full IBR4 compliance for chan_zap
79 -- More flexible CDR handling
80 -- Distinguish between BUSY and FAILURE on outbound calls
81 -- Add initial support for SCCP via chan_skinny
82 -- Better support for Future Group B signaling
84 -- Retain IAX2 and SIP registrations past shutdown/crash and restart
85 -- True data mode bridging when possible
86 -- H.323 build improvements
87 -- Agent Callback-login support
88 -- RFC2833 Improvements
89 -- Add thread debugging
90 -- Add optional pedantic SIP checking for Pingtel
91 -- Allow extension names, include context, switch to use global vars.
92 -- Allow variables in extensions.conf to reference previously defined ones
93 -- Merge voicemail enhancements (app_voicemail2)
94 -- Add multiple queueing strategies
95 -- Merge support for 'T'
96 -- Allow pending agent calling (Agent/:1)
97 -- Add groupings to agents.conf
98 -- Add video support to IAX2
99 -- Zaptel optimize playback
100 -- Add video support to SIP
101 -- Make RTP ports configurable
102 -- Add RDNIS support to SIP and IAX2
103 -- Add transfer app (implement in SIP and IAX2)
104 -- Make voicemail segmentable by context (app_voicemail2)
105 -- Major restructuring of voicemail (app_voicemail2)
106 -- Add initial ENUM support
107 -- Add malloc debugging support
108 -- Add preliminary Voicetronix support
111 -- Merge and edit Nick's FXO dial support
112 -- Reengineer SIP registration (outbound)
113 -- Support call pickup on SIP and compatibly with ZAP
114 -- Support 302 Redirect on SIP
115 -- Management interface improvements
116 -- Add "hint" support
117 -- Improve call forwarding using new "Local" channel driver.
118 -- Add "Local" channel
119 -- Substantial SIP enhancements including retransmissions
120 -- Enforce case sensitivity on extension/context names
121 -- Add monitor support (Thanks, Mahmut)
122 -- Add experimental "trunk" option to IAX2 for high density VoIP
123 -- Add experimental "debug channel" command
124 -- Add 'C' flag to dial command to reset call detail record (handy for calling cards)
125 -- Add NAT and dynamic support to MGCP
126 -- Allow selection of in-band, out-of-band, or INFO based DTMF
127 -- Add contributed "*80" support to blacklist numbers (Thanks James!)
128 -- Add "NAT" option to sip user, peer, friend
129 -- Add experimental "IAX2" protocol
130 -- Change special variable "EXTEN-n" to "EXTEN:n" to follow Bash syntax
131 -- Add "Enhanced" AGI with audio pass-through (voice recognition anyone?)
132 -- Choose best priority from codec from allow/disallow
133 -- Reject SIP calls to self
134 -- Allow SIP registration to provide an alternative contact
135 -- Make HOLD on SIP make use of asterisk MOH
136 -- Add supervised transfer (tested with Pingtel only)
137 -- Allow maxexpirey and defaultexpirey to be runtime configurable for SIP
138 -- Preliminary codec 13 support (RFC3389)
139 -- Add app_authenticate for general purpose authentication
140 -- Optimize RTP and smoother
141 -- Create special variable "EXTEN-n" where it is extension stripped by n MSD
142 -- Fix uninitialized frame pointer in channel.c
143 -- Add global variables support under [globals] of extensions.conf
144 -- Add macro support (show application Macro)
145 -- Allow [123-5] etc in extensions
146 -- Allow format of App(arg1,arg2,...) instead of just App,arg1|arg2 in dialplan
147 -- Add message waiting indicator to SIP
148 -- Fix double free bug in channel.c
150 -- Add fastfoward, rewind, seek, and truncate functions to streams
151 -- Support registration
153 -- Permit applications to return a digit indicating new extension
154 -- Change "SHUTDOWN" to "STOP" in commands
155 -- SIP "Hold" fixes and VXML URI support
156 -- New chan_zap with 160 sample chunk size
157 -- Add DTMF, MF, and Fax tone detector to dsp routines
158 -- Allow overlap dialing (inbound) on PRI
159 -- Enable tone detection with PRI
160 -- Add special information tone detection
161 -- Add Asterisk DB support
163 -- Re-record all system prompts
164 -- Change "timelen" to samples for better accuracy
165 -- Move to editline, eliminating readline dependency
166 -- Add peer "poke" support to SIP and IAX
167 -- Add experimental call progress detection
168 -- Add SIP authentication (digest)
170 -- Reroute faxes to "fax" extension
171 -- Create ISDN/modem group concept
172 -- Centralize indication
173 -- Add initial MGCP support
174 -- SIP debugging cleanup
176 -- SIP commands (show channels, etc)
177 -- Add optional busy detection
178 -- Add Visual Message Waiting Indicator (MDMF and SDMF)
179 -- Add ambiguous extension matching
181 -- Major SIP enhancements from SIPit
182 -- Rewrite of ZAP CLASS features using subchannels
183 -- Enhanced call parking
184 -- Add extended outgoing spool support (pbx_spool)
186 -- Outbound origination API
187 -- Call management improvements
188 -- Add Do Not Disturb (*78, *79)
190 -- Document variables
191 -- Add transfer capability on the console
192 -- Add SpeeX codec translator
194 -- Add setcallerid functionality (AGI, application)
195 -- Add special variables ${CALLERID}, ${EXTEN}, ${CONTEXT}, ${PRIORITY}
196 -- Don't echo cancel on pure TDM connections by default
197 -- Implement Async GOTO
198 -- Differentiate softhangups
201 -- Fix for Big Endian machines
203 -- Various SIP fixes and enhancements
204 -- Add "zapateller application and arbitrary tone pairs
205 -- Don't always start at "s"
206 -- Separate linear mode for pseudo and real
207 -- Add initial RTP and SIP support (no jitter buffer yet, unknown stability)
208 -- Add 'h' extension, executed on hangup
209 -- Add duration timer to message info
210 -- Add web based voicemail checking ("make webvmail")
211 -- Add ast_queue_frame function and eliminate frame pipes in most drivers
212 -- Centralize host access (and possibly future ACL's)
213 -- Add Caller*ID on PhoneJack (Thanks Nathan)
214 -- Add "safe_asterisk" wrapper script to auto-restart Asterisk
215 -- Indicate ringback on chan_phone
216 -- Add answer confirmation (press '#' to confirm answer)
217 -- Add distinctive ring support (e.g. Dial,Zap/4r2)
218 -- Add ANSI/vt100 color support
219 -- Make parking configurable through parking.conf
220 -- Fix the empty voicemail problem
222 -- Add ADSI Compiler (app_adsiprog)
223 -- Extensive DISA re-work to improve tone generation
224 -- Reset all idle channels every 10 minutes on a PRI
225 -- Reset channels which are hungup with "channel in use"
226 -- Implement VNAK support in chan_iax
227 -- Fix chan_oss to support proper hangups and autoanswer
228 -- Make shutdown properly hangup channels
229 -- Add idling capability to chan_zap for idle-net
230 -- Add "MeetMe" conferencing app (app_meetme)
231 -- Add timing information to include
233 -- Add ISDN RAS capability
234 -- Add stutter dialtone to Chan Zap
235 -- Add "#include" capability to config files.
236 -- Add call-forward variable to Chan Zap (*72, *73)
237 -- Optimize IAX flow when transfer isn't possible
238 -- Allow transmission of ANI over IAX
240 -- Make ast_readstring parameter be the max # of digits, not the max size with \0
241 -- Make up any missing messages on the fly
242 -- Add support for specific DTMF interruption to saying numbers
243 -- Add new "u" and "b" options to condense busy/unavail handling
244 -- Add support for RSA authentication on IAX calls
245 -- Add support for ADSI compatible CPE
246 -- Outgoing call queue
247 -- Remote dialplan fixes for Quicknet
248 -- Added AGI commands supporting TDD functions (RECEIVE CHAR & TDD MODE)
249 -- Added TDD support (send/receive text in chan_zap)
250 -- Fix all strncpy references
251 -- Implement CSV CDR backend
252 -- Implement Call Detail Records
254 -- Implement IAX quelching
255 -- Allow Caller*ID to be overridden and suggested
256 -- Configure defaults to use IAXTEL
257 -- Allow remote dialplan polling via IAX
258 -- Eliminate ast_longest_extension
259 -- Implement dialplan request/reply
260 -- Let peers have allow/disallow for codecs
261 -- Change allow/deny to permit/deny in IAX
262 -- Allow dialplan entries to match Caller*ID as well
263 -- Added AGI (Asterisk Gateway Interface) scripting interface (app_agi)
264 -- Added chan_zap for zapata telephony kernel interface, removed chan_tor
265 -- Add convenience functions
266 -- Fix race condition in channel hangup
267 -- Fix memory leaks in both asterisk and iax frame allocations
268 -- Add "iax show stats" command and -DTRACE_FRAMES (for frame tracing)
269 -- Add DISA application (Thanks to Jim Dixon)
270 -- Add IAX transfer support
271 -- Add URL and HTML transmission
272 -- Add application for sending images
273 -- Add RedHat RPM spec file and build capability
274 -- Fix GSM WAV file format bug
275 -- Move ignorepat to main dialplan
276 -- Add ability to specificy TOS bits in IAX
277 -- Allow username:password in IAX strings
278 -- Updates to PhoneJack interface
279 -- Allow "servermail" in voicemail.conf to override e-mail in "from" line
280 -- Add 'skip' option to app_playback
281 -- Reject IAX calls on unknown extensions
284 -- Keep track of version information
285 -- Add -f to cause Asterisk not to fork
286 -- Keep important information in voicemail .txt file
287 -- Adtran Voice over Frame Relay updates
288 -- Implement option setting/querying of channel drivers
289 -- IAX performance improvements and protocol fixes
290 -- Substantial enhancement of console channel driver
291 -- Add IAX registration. Now IAX can dynamically register
292 -- Add flash-hook transfer on tormenta channels
293 -- Added Three Way Calling on tormenta channels
294 -- Start on concept of zombie channel
295 -- Add Call Waiting CallerID
296 -- Keep track of who registeres contexts, includes, and extensions
297 -- Added Call Waiting(tm), *67, *70, and *82 codes
298 -- Move parked calls into "parkedcalls" context by default
299 -- Allow dialplan to be displayed
300 -- Allow "=>" instead of just "=" to make instantiation clearer
301 -- Asterisk forks if called with no arguments
302 -- Add remote control by running asterisk -vvvc
303 -- Adjust verboseness with "set verbose" now
304 -- No longer requires libaudiofile
306 -- Make PBX Config module reload extensions on SIGHUP
307 -- Allow modules to be reloaded when SIGHUP is received
308 -- Variables now contain line numbers
309 -- Make dialer send in band signalling
310 -- Add record application
311 -- Added PRI signalling to Tormenta driver
312 -- Allow use of BYEXTENSION in "Goto"
313 -- Allow adjustment of gains on tormenta channels
314 -- Added raw PCM file format support
315 -- Add U-law translator
316 -- Fix DTMF handling in bridge code
317 -- Fix access control with IAX
319 -- Update configuration files and add some missing sounds
320 -- Added ability to include one context in another
321 -- Rewrite of PBX switching
322 -- Major mods to dialler application
323 -- Added Caller*ID spill reception
324 -- Added Dialogic VOX file format support
326 -- Add Tormenta driver (RBS signalling)
327 -- Add Caller*ID spill creation
328 -- Rewrite of translation layer entirely
329 -- Add ability to run PBX without additional thread
331 -- Make app_dial handle a lack of translators smoothly
332 -- Add ISDN4Linux support -- dtmf is weird...
335 -- Fix a small mistake in IAX
336 -- Fix the QuickNet driver to work with newer cards
338 -- Update VoFR some more
339 -- Fix the QuickNet driver to work with LineJack
340 -- Add ability to pass images for IAX.
342 -- Update VoFR for latest sangoma code
343 -- Update QuickNet Driver
344 -- Add text message handling
345 -- Fix transfers to use "default" if not in current context
347 -- Improve format/content negotiation
348 -- Added support for multiple languages
349 -- Bug fixes, as always...
351 -- Updated README file with a "Getting Started" section
352 -- Added sample sounds and configuration files.
353 -- Added LPC10 very low bandwidth (low quality) compression
354 -- Enhanced translation selection mechanism.
355 -- Enhanced IAX jitter buffer, improved reliability
356 -- Support echo cancelation on PhoneJack
357 -- Updated PhoneJack driver to std. Telephony interface
358 -- Added app_echo for evaluating VoIP latency
359 -- Added app_system to execute arbitrary programs
360 -- Updated sample configuration files
361 -- Added OSS channel driver (full duplex only)
362 -- Added IAX implementation
363 -- Fixed some deadlocks.
364 -- A whole bunch of bug fixes
366 -- Revised translator, fixed some general race conditions throughout *
367 -- Made dialer somewhat more aware of incompatible voice channels
368 -- Added Voice Modem driver and A/Open Modem Driver stub
369 -- Added MP3 decoder channel
370 -- Added Microsoft WAV49 support
371 -- Revised License -- Pure GPL, nothing else
372 -- Modified Copyright statement since code is still currently owned by author
373 -- Added RAW GSM headerless data format
374 -- Innumerable bug fixes
projects / asterisk/asterisk.git / blob