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