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