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