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