1 Changes since Asterisk 1.2:
3 * over 4,000 commits since 1.2
6 o Change the way CLI commands are structured.
7 o Most commands are now <module> <verb> <args>
10 * SLA (Shared Line Appearance) support
11 * T.38 Passthrough Support for faxing in SIP
12 * Generic channel jitterbuffer (spawned from RTP)
13 * Variable Length DTMF for better DTMF compatibility
14 * Improved chan_iax2 scalability by using multithreading
15 * AEL2 has replaced the original implementation of AEL. The "2" is removed. For more details,
16 read: http://www.voip-info.org/wiki/view/Asterisk+AEL2
17 AEL is no longer considered experimental.
18 * New sounds; English, Spanish, and French prompts, as well as music on hold files, in
19 multiple Asterisk native formats.
20 * IMAP storage of voicemail
21 * Jabber/GoogleTalk integration
22 * New speech recognition API for interfacing to different Voice Recognition software packages
23 * much more customizable and portable build system
24 o also for asterisk-addons
27 * SMDI (Simplified Message Desk Interface) support
28 * Redesign of MusicOnHold configuration settings
30 * Significant chan_skinny updates
31 * Significant chan_misdn updates
32 * Improved SIP transfers
33 * SIP MWI subscription support
34 * Much improved support for SIP video
35 * Control over SIP transfers and subscriptions (enable/disable per device)
36 * ChanSpy whisper mode (Whisper Paging)
37 * Configurable language support for saying dates and times
38 * Significant architecture improvements for memory usage and performance
39 * Media-only IAX2 transfers
40 * Updates to the Radio Repeater app code
41 * Deprecation of AgentCallbackLogin in favor of a dialplan-based solution
42 * uClibc builds supported
43 * Work done for freeBSD portability
44 * Work done for Solaris portability
45 * FreeTDS-based database can be used with Realtime
46 * New internal data structure, stringfields, is implemented in IAX and SIP, reducing memory consumption by about 50%.
47 * Use of thread local storage for reduced memory allocation/freeing and lower stack consumption
48 * Reorganized files into docs/ main/ configs/, including name changes in some cases
49 * Much effort was expended in arranging documentation in source files in doxygen format
50 * Improved IP TOS support for IAX and SIP
51 * Builtin mini HTTP server
52 * Added support for Sigma Designs cards.
53 * Frame header caching to reduce memory allocation/freeing
54 * Passthrough and record/playback support for G.722 wideband audio
55 * using mpg123 to play MP3 files for music-on-hold will be deprecated in 1.4 (start using the "native support")
57 1. AMD() ;; Answering Machine Detection
58 2. ChannelRedirect() ;; asynch goto, redirect chan to context/exten/priority
59 3. ContinueWhile() ;; Addition to the While() suite. Acts like "continue".
60 4. ExitWhile() ;; Addition to the While() suite. Acts like "break".
61 5. ExtenSpy() ;; A close cousin to ChanSpy().
62 6. FollowMe() ;; findme/followme call redirect app
63 7. Log() ;; Send a message to the log, based on severity level.
64 8. MacroExclusive() ;; No more than one invocation of this macro allowed at any one time.
65 9. MorseCode() ;; turns strings into dits and dahs. A playground for ham radio licensees!
66 10. OSPAuth() ;; OSP authentication
67 11. QueueLog() ;; allows you to write your own events into the queue log
68 12. SLAStation() ;; Shared Line Appearance
69 13. SLATrunk() ;; Shared Line Appearance
70 14. SpeechCreate() ;; Voice Recognition Engine interface...
71 15. SpeechActivateGrammar()
74 18. SpeechDeactivateGrammar()
75 19. SpeechProcessingSound()
77 21. SpeechLoadGrammar()
78 22. SpeechUnloadGrammar()
79 23. StopMixMonitor() ;; to stop the MixMonitor App.
80 24. TryExec() ;; execute dialplan app without fatal consequences
82 1. CheckGroup -- do a comparison to ${GROUP()}
83 2. Curl -- use the function CURL() instead
84 3. Cut -- use the function CUT() instead
85 4. DateTime -- use sayunixtime() app instead.
86 5. DBget -- deprecated in 1.2, now removed.
87 6. DBput -- deprecated in 1.2, now removed.
88 7. Enumlookup -- use the function ENUMLOOKUP() instead
89 8. Eval -- use the function EVAL() instead
90 9. GetGroupCount -- use the function GROUP_COUNT() instead
91 10. GetGroupMatchCount -- use the function GROUP_MATCH_COUNT() instead
92 11. Intercom -- use the chan_oss module instead
93 12. Math -- use the function MATH() instead
94 13. MD5 -- use the function MD5() instead
95 14. SetCIDname -- use the function CALLERID(name) instead
96 15. SetCIDnum -- use the function CALLERID(number) instead
97 16. SetGroup -- use Set(GROUP=group) instead
98 17. SetRDNIS -- use the function CALLERID(rdnis) instead
99 18. Sql_postgres -- was deprecated in 1.2, now removed
100 19. Txtcidname -- use the function TXTCIDNAME instead
101 * New Dialplan Functions:
123 * Apps that have changes to their interface:
124 1. Authenticate() -- optional maxdigits argument added.
125 2. ChanSpy() -- new options:
126 o w -- Enable 'whisper' mode, so the spying channel can talk to...
127 o W -- Enable 'private whisper' mode, so the spying channel can...
128 3. DBdel() -- now marked as DEPRECATED in favor of the DB_DELETE func
130 o New Option: O([x]) for Zaptel operator mode
131 o New Option: K/k parking via dtmf tones
132 5. Dictate() -- optional filename argument added.
133 6. Directory() -- new option: e - In addition to the name, also read the extension number...
134 7. ForkCDR() -- new options:
135 o 'a' -- update answer time on new cdr
136 o 'A' -- Lock the orig CDR answer time against changes.
137 o 'D' -- Copy the disposition from the orig to the new CDR.
138 o 'd' -- clear the dstcannel field in the new CDR.
139 o 'e' -- set the end time of the original CDR.
140 o 'R' -- do NOT reset the new CDR.
141 o 's' -- Add/change var in orig CDR.
142 o 'T' -- Force ast_cdr_end, answer to obey LOCKED flag for the orig. CDR.
143 -- ast_cdr_setvar will be forced also (used by the CDR() func in write mode)
144 8. Meetme() -- new options:
145 o 'I' -- announce user join/leave without review
146 o 'l' -- set listen only mode (Listen only, no talking)
147 o 'o' -- set talker optimization - treats talkers who aren't speaking as...
148 o '1' -- do not play message when first person enters
149 9. MeetmeAdmin() -- new options:
150 o 'r' -- Reset one user's volume settings
151 o 'R' -- Reset all users volume settings
152 o 's' -- Lower entire conference speaking volume
153 o 'S' -- Raise entire conference speaking volume
154 o 't' -- Lower one user's talk volume
155 o 'T' -- Lower all users talk volume
156 o 'u' -- Lower one user's listen volume
157 o 'U' -- Lower all users listen volume
158 o 'v' -- Lower entire conference listening volume
159 o 'V' -- Raise entire conference listening volume
160 10. OSPFinish() : now also can return ERROR result.
161 11. OSPLookup() : Sets more variables, also now returns ERROR result.
162 12. Page() -- New option: r - record the page into a file (see 'r' for app_meetme)
163 13. Pickup() -- multiple extensions, PICKUPMARK; read the description!
167 15. Random() -- is now deprecated in 1.4
168 16. Read() -- replace 'skip' and 'noanswer' options with 's', 'n', add 'i' option.
169 17. Record() -- New option: 'x' : ignore all terminator keys (DTMF) and keep recording until hangup
170 18. UserEvent() -- slight change in behavior. Read the description.
171 19. VoiceMailMain() -- new a(#) option, goes to folder # directly.
172 20. WaitForSilence() -- new optional 3rd arg, time delay before returning.
173 * Functions that have changes to their interfaces:
174 1. CDR -- new options: u and s
175 2. LANGUAGE -- Deprecated. Use CHANNEL(language) instead.
176 3. MUSICCLASS -- Deprecated. Use CHANNEL(musicclass) instead.
177 * Configuration File Changes:
179 1. amd.conf -- Answering Machine Detection parameters
180 2. followme.conf -- parameters for the findme/followme call forwarding
181 3. func_odbc.conf -- define sql access functions here
182 4. gtalk.conf -- how to handle gtalk protocol calls
183 5. h323.conf -- h323 configuration
184 6. http.conf -- config for the builtin mini-http server in asterisk
185 7. jabber.conf -- jabber interface
186 8. jingle.conf -- jingle protocol interface config
187 10. res_snmp.conf -- to enable snmp in asterisk, and define full/sub agent status
188 11. say.conf -- define per-language rules for numbers, dates, etc.
189 12. skinny.conf -- for those special skinny phones you want to use...
190 13. sla.conf -- Shared Line Appearance config
191 14. smdi.conf -- SMDI messaging config
192 15. udptl.conf -- T38's udptl transport config
193 16. users.conf -- user config
194 2. Changes to Existing Config files:
196 o Jitterbuffer support added to several channels. Usually adds these variables to a config file:
202 o MusicOnHold upgrade introduces two new variables:
206 o maxlogintries variable added
207 o autologoffunavail variable added
208 o endcall variable added
209 o agentgoodbye variable added
210 o createlink variable REMOVED
212 o mohinterpret variable added
213 o Jitterbuffer variables added
215 o endbeforehexten variable added
216 o sections for csv and radius added, with variables usegmtime, loguniqueid,
217 loguserfield, and radiuscfg variables.
219 o table variable added
221 o Many upgrades. See the info at http://www.voip-info.org/wiki/view/Asterisk+AEL2
223 o autofallthru now set to "yes" by default
224 o userscontext variable added
225 o added info/examples on paging and hints.
227 o parkedplay variable added (who to beep at)
229 o atxfernoanswertimeout variable added
230 o parkcall variable added (one step parking)
231 o improved documentation for dynamic feature declarations!
233 o adsi variable added
234 o mohinterpret variable added
235 o mohsuggest variable added
236 o jitterbuffer updates
237 o iaxthreadcount variable added
238 o iaxmaxthreadcount variable added
239 o the way to specify TOS has changed.
240 o mailboxdetail variable has been REMOVED.
242 o [bg] entry added (Bulgaria).
243 o [il] entry added (Israel)
244 o [in] entry added (India)
245 o [jp] entry added (Japan)
246 o [my] entry added (Malaysia)
247 o [th] entry added (Thailand)
249 o webenabled variable added
250 o httptimeout variable added
251 o timestampevents variable added
253 o Jitterbuffer support added
255 o l1watcher_timeout variable added
256 o pp_l2_check variable added
257 o echocancelwhenbridged variable added
258 o echotraining variable added
259 o max_incoming variable added
260 o max_outgoing variable added
262 o a comment for preloading res_speech.so is added
263 o mention of global symbols is removed
264 o obsolesced entries for chan_modem_* and app_intercom have been removed
266 o the default is now to do native moh from /var/lib/asterisk/moh
268 o authpolicy variable added
270 o debug variable added
271 o device variable added
272 o mixer variable added
273 o boost variable added
274 o callerid variable added
275 o autohangup variable added
276 o queuesize variable added
277 o frags variable added
278 o JitterBuffer support
279 o sections to define alternate sound cards
281 o autofill variable added
282 o monitor-type variable added
283 o musiconhold is now musicclass, with a difference in interpretation
284 o autofill variable added
285 o autopause variable added
286 o setinterfacevar variable added
287 o ringinuse variable added
289 o pooling variable added
291 o duplex variable added
292 o tailmessagetime variable added
293 o tailsquashedtime variable added
294 o tailmessages variable added
296 o rtcpinterval varaible added
298 o allowoverlap variable added
299 o allowtransfer variable added
300 o tos variable REMOVED
301 o tos_sip variable added
302 o tos_audio variable added
303 o tos_video variable added
304 o minexpiry variable added
305 o t1min variable added
306 o musicclass variable REMOVED
307 o mohinterpret variable added
308 o maxcallbitratesuggest variable added
309 o allowsubscribe variable added
310 o videosupport variable added
311 o maxcallbitrate variable added
312 o g726nonstandard variable added
313 o dumphistory variable added
314 o allowsubscribe variable added
315 o t38pt_udptl variable added
316 o canreinvite variable can also now be set to 'nonat'
317 o rtsavesysname variable added
318 o JitterBuffer support added
319 o t38pt_usertpsource variable added
321 o port variable renamed to bindport
322 o JitterBuffer support added
323 o model variable REMOVED
324 o mohinterpret variable added
325 o mohsuggest variable added
326 o speeddial variable added
327 o addon variable added
329 o userscontext variable added
330 o smdiport variable added
331 o attachfmt variable added
332 o volgain variable added
333 o tempgreetwarn variable added
335 o pritimer variable has improved documentation
336 o New signalling method: fgccama
337 o New signalling method: fgccamamf
338 o outsignalling variable added
339 o distinctiveringaftercid variable added
340 o cidsignalling now also accepts v23_jp, and smdi
341 o usesmdi variable added
342 o smdiport variable added
343 o mohinterpret variable added
344 o mohsuggest variable added
345 o JitterBuffer support added
346 * Removed Codecs/Channels:
347 1. codec_g723 was removed because the actual codec implementation it was designed to use is not distributable
348 2. chan_modem_* and related modules are gone because the kernel support for those interfaces is old, buggy and unsupported
350 1. aelparse -- compile .ael files outside of asterisk
351 * New manager events:
352 1. OriginateResponse event comes to replace OriginateSuccess and OriginateFailure
353 * iLBC source code no longer included (see UPGRADE.txt for details)
354 * New CLI command "pri show version" that shows the current version of libpri
355 that the library was built against (requires a version of libpri since this API