1 -- Merge and edit Nick's FXO dial support
2 -- Reengineer SIP registration (outbound)
3 -- Support call pickup on SIP and compatibly with ZAP
4 -- Support 302 Redirect on SIP
5 -- Management interface improvements
7 -- Improve call forwarding using new "Local" channel driver.
9 -- Substantial SIP enhancements including retransmissions
10 -- Enforce case sensitivity on extension/context names
11 -- Add monitor support (Thanks, Mahmut)
12 -- Add experimental "trunk" option to IAX2 for high density VoIP
13 -- Add experimental "debug channel" command
14 -- Add 'C' flag to dial command to reset call detail record (handy for calling cards)
15 -- Add NAT and dynamic support to MGCP
16 -- Allow selection of in-band, out-of-band, or INFO based DTMF
17 -- Add contributed "*80" support to blacklist numbers (Thanks James!)
18 -- Add "NAT" option to sip user, peer, friend
19 -- Add experimental "IAX2" protocol
20 -- Change special variable "EXTEN-n" to "EXTEN:n" to follow Bash syntax
21 -- Add "Enhanced" AGI with audio pass-through (voice recognition anyone?)
22 -- Choose best priority from codec from allow/disallow
23 -- Reject SIP calls to self
24 -- Allow SIP registration to provide an alternative contact
25 -- Make HOLD on SIP make use of asterisk MOH
26 -- Add supervised transfer (tested with Pingtel only)
27 -- Allow maxexpirey and defaultexpirey to be runtime configurable for SIP
28 -- Preliminary codec 13 support (RFC3389)
29 -- Add app_authenticate for general purpose authentication
30 -- Optimize RTP and smoother
31 -- Create special variable "EXTEN-n" where it is extension stripped by n MSD
32 -- Fix uninitialized frame pointer in channel.c
33 -- Add global variables support under [globals] of extensions.conf
34 -- Add macro support (show application Macro)
35 -- Allow [123-5] etc in extensions
36 -- Allow format of App(arg1,arg2,...) instead of just App,arg1|arg2 in dialplan
37 -- Add message waiting indicator to SIP
38 -- Fix double free bug in channel.c
40 -- Add fastfoward, rewind, seek, and truncate functions to streams
41 -- Support registration
43 -- Permit applications to return a digit indicating new extension
44 -- Change "SHUTDOWN" to "STOP" in commands
45 -- SIP "Hold" fixes and VXML URI support
46 -- New chan_zap with 160 sample chunk size
47 -- Add DTMF, MF, and Fax tone detector to dsp routines
48 -- Allow overlap dialing (inbound) on PRI
49 -- Enable tone detection with PRI
50 -- Add special information tone detection
51 -- Add Asterisk DB support
53 -- Re-record all system prompts
54 -- Change "timelen" to samples for better accuracy
55 -- Move to editline, eliminating readline dependency
56 -- Add peer "poke" support to SIP and IAX
57 -- Add experimental call progress detection
58 -- Add SIP authentication (digest)
60 -- Reroute faxes to "fax" extension
61 -- Create ISDN/modem group concept
62 -- Centralize indication
63 -- Add initial MGCP support
64 -- SIP debugging cleanup
66 -- SIP commands (show channels, etc)
67 -- Add optional busy detection
68 -- Add Visual Message Waiting Indicator (MDMF and SDMF)
69 -- Add ambiguous extension matching
71 -- Major SIP enhancements from SIPit
72 -- Rewrite of ZAP CLASS features using subchannels
73 -- Enhanced call parking
74 -- Add extended outgoing spool support (pbx_spool)
76 -- Outbound origination API
77 -- Call management improvements
78 -- Add Do Not Disturb (*78, *79)
81 -- Add transfer capability on the console
82 -- Add SpeeX codec translator
84 -- Add setcallerid functionality (AGI, application)
85 -- Add special variables ${CALLERID}, ${EXTEN}, ${CONTEXT}, ${PRIORITY}
86 -- Don't echo cancel on pure TDM connections by default
87 -- Implement Async GOTO
88 -- Differentiate softhangups
91 -- Fix for Big Endian machines
93 -- Various SIP fixes and enhancements
94 -- Add "zapateller application and arbitrary tone pairs
95 -- Don't always start at "s"
96 -- Separate linear mode for pseudo and real
97 -- Add initial RTP and SIP support (no jitter buffer yet, unknown stability)
98 -- Add 'h' extension, executed on hangup
99 -- Add duration timer to message info
100 -- Add web based voicemail checking ("make webvmail")
101 -- Add ast_queue_frame function and eliminate frame pipes in most drivers
102 -- Centralize host access (and possibly future ACL's)
103 -- Add Caller*ID on PhoneJack (Thanks Nathan)
104 -- Add "safe_asterisk" wrapper script to auto-restart Asterisk
105 -- Indicate ringback on chan_phone
106 -- Add answer confirmation (press '#' to confirm answer)
107 -- Add distinctive ring support (e.g. Dial,Zap/4r2)
108 -- Add ANSI/vt100 color support
109 -- Make parking configurable through parking.conf
110 -- Fix the empty voicemail problem
112 -- Add ADSI Compiler (app_adsiprog)
113 -- Extensive DISA re-work to improve tone generation
114 -- Reset all idle channels every 10 minutes on a PRI
115 -- Reset channels which are hungup with "channel in use"
116 -- Implement VNAK support in chan_iax
117 -- Fix chan_oss to support proper hangups and autoanswer
118 -- Make shutdown properly hangup channels
119 -- Add idling capability to chan_zap for idle-net
120 -- Add "MeetMe" conferencing app (app_meetme)
121 -- Add timing information to include
123 -- Add ISDN RAS capability
124 -- Add stutter dialtone to Chan Zap
125 -- Add "#include" capability to config files.
126 -- Add call-forward variable to Chan Zap (*72, *73)
127 -- Optimize IAX flow when transfer isn't possible
128 -- Allow transmission of ANI over IAX
130 -- Make ast_readstring parameter be the max # of digits, not the max size with \0
131 -- Make up any missing messages on the fly
132 -- Add support for specific DTMF interruption to saying numbers
133 -- Add new "u" and "b" options to condense busy/unavail handling
134 -- Add support for RSA authentication on IAX calls
135 -- Add support for ADSI compatible CPE
136 -- Outgoing call queue
137 -- Remote dialplan fixes for Quicknet
138 -- Added AGI commands supporting TDD functions (RECEIVE CHAR & TDD MODE)
139 -- Added TDD support (send/receive text in chan_zap)
140 -- Fix all strncpy references
141 -- Implement CSV CDR backend
142 -- Implement Call Detail Records
144 -- Implement IAX quelching
145 -- Allow Caller*ID to be overridden and suggested
146 -- Configure defaults to use IAXTEL
147 -- Allow remote dialplan polling via IAX
148 -- Eliminate ast_longest_extension
149 -- Implement dialplan request/reply
150 -- Let peers have allow/disallow for codecs
151 -- Change allow/deny to permit/deny in IAX
152 -- Allow dialplan entries to match Caller*ID as well
153 -- Added AGI (Asterisk Gateway Interface) scripting interface (app_agi)
154 -- Added chan_zap for zapata telephony kernel interface, removed chan_tor
155 -- Add convenience functions
156 -- Fix race condition in channel hangup
157 -- Fix memory leaks in both asterisk and iax frame allocations
158 -- Add "iax show stats" command and -DTRACE_FRAMES (for frame tracing)
159 -- Add DISA application (Thanks to Jim Dixon)
160 -- Add IAX transfer support
161 -- Add URL and HTML transmission
162 -- Add application for sending images
163 -- Add RedHat RPM spec file and build capability
164 -- Fix GSM WAV file format bug
165 -- Move ignorepat to main dialplan
166 -- Add ability to specificy TOS bits in IAX
167 -- Allow username:password in IAX strings
168 -- Updates to PhoneJack interface
169 -- Allow "servermail" in voicemail.conf to override e-mail in "from" line
170 -- Add 'skip' option to app_playback
171 -- Reject IAX calls on unknown extensions
174 -- Keep track of version information
175 -- Add -f to cause Asterisk not to fork
176 -- Keep important information in voicemail .txt file
177 -- Adtran Voice over Frame Relay updates
178 -- Implement option setting/querying of channel drivers
179 -- IAX performance improvements and protocol fixes
180 -- Substantial enhancement of console channel driver
181 -- Add IAX registration. Now IAX can dynamically register
182 -- Add flash-hook transfer on tormenta channels
183 -- Added Three Way Calling on tormenta channels
184 -- Start on concept of zombie channel
185 -- Add Call Waiting CallerID
186 -- Keep track of who registeres contexts, includes, and extensions
187 -- Added Call Waiting(tm), *67, *70, and *82 codes
188 -- Move parked calls into "parkedcalls" context by default
189 -- Allow dialplan to be displayed
190 -- Allow "=>" instead of just "=" to make instantiation clearer
191 -- Asterisk forks if called with no arguments
192 -- Add remote control by running asterisk -vvvc
193 -- Adjust verboseness with "set verbose" now
194 -- No longer requires libaudiofile
196 -- Make PBX Config module reload extensions on SIGHUP
197 -- Allow modules to be reloaded when SIGHUP is received
198 -- Variables now contain line numbers
199 -- Make dialer send in band signalling
200 -- Add record application
201 -- Added PRI signalling to Tormenta driver
202 -- Allow use of BYEXTENSION in "Goto"
203 -- Allow adjustment of gains on tormenta channels
204 -- Added raw PCM file format support
205 -- Add U-law translator
206 -- Fix DTMF handling in bridge code
207 -- Fix access control with IAX
209 -- Update configuration files and add some missing sounds
210 -- Added ability to include one context in another
211 -- Rewrite of PBX switching
212 -- Major mods to dialler application
213 -- Added Caller*ID spill reception
214 -- Added Dialogic VOX file format support
216 -- Add Tormenta driver (RBS signalling)
217 -- Add Caller*ID spill creation
218 -- Rewrite of translation layer entirely
219 -- Add ability to run PBX without additional thread
221 -- Make app_dial handle a lack of translators smoothly
222 -- Add ISDN4Linux support -- dtmf is weird...
225 -- Fix a small mistake in IAX
226 -- Fix the QuickNet driver to work with newer cards
228 -- Update VoFR some more
229 -- Fix the QuickNet driver to work with LineJack
230 -- Add ability to pass images for IAX.
232 -- Update VoFR for latest sangoma code
233 -- Update QuickNet Driver
234 -- Add text message handling
235 -- Fix transfers to use "default" if not in current context
237 -- Improve format/content negotiation
238 -- Added support for multiple languages
239 -- Bug fixes, as always...
241 -- Updated README file with a "Getting Started" section
242 -- Added sample sounds and configuration files.
243 -- Added LPC10 very low bandwidth (low quality) compression
244 -- Enhanced translation selection mechanism.
245 -- Enhanced IAX jitter buffer, improved reliability
246 -- Support echo cancelation on PhoneJack
247 -- Updated PhoneJack driver to std. Telephony interface
248 -- Added app_echo for evaluating VoIP latency
249 -- Added app_system to execute arbitrary programs
250 -- Updated sample configuration files
251 -- Added OSS channel driver (full duplex only)
252 -- Added IAX implementation
253 -- Fixed some deadlocks.
254 -- A whole bunch of bug fixes
256 -- Revised translator, fixed some general race conditions throughout *
257 -- Made dialer somewhat more aware of incompatible voice channels
258 -- Added Voice Modem driver and A/Open Modem Driver stub
259 -- Added MP3 decoder channel
260 -- Added Microsoft WAV49 support
261 -- Revised License -- Pure GPL, nothing else
262 -- Modified Copyright statement since code is still currently owned by author
263 -- Added RAW GSM headerless data format
264 -- Innumerable bug fixes