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