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1 Changes since Asterisk 1.2:
3 * over 4,000 commits since 1.2
4 * queue member naming
5 * CLI commands rework
6 o Change the way CLI commands are structured.
7 o Most commands are now <module> <verb> <args>
8 * chan_h323 update
9 * multi-parking
10 * RTP packetization
11 * SLA (Shared Line Appearance) support
12 * T.38 Passthrough Support for faxing
13 * Generic channel jitterbuffer (spawned from RTP)
14 * VLDTMF for better DTMF compatibility
15 * Improved chan_iax2 scalability
16 * AEL2 has replaced the original implementation of AEL. The "2" is removed. For more details,
17 read: http://www.voip-info.org/wiki/view/Asterisk+AEL2
18 * New sounds; English, Spanish, and French prompts, as well as music on hold files, in multiple Asterisk native formats.
19 * IMAP storage of voicemail
20 * Jabber/GoogleTalk
21 * New speech recognition API for interfacing to different Voice Recognition software packages
22 * much more customizable and portable build system
23 o also for asterisk-addons
24 * Radius CDR logging
25 * SNMP support
26 * SMDI (Simplified Message Desk Interface) support
27 * Manager over HTTP
28 * Significant chan_skinny updates
29 * Significant chan_misdn updates
30 * Improved SIP transfers
31 * ChanSpy whisper mode (Whisper Paging)
32 * Configurable language support for saying dates and times
33 * Significant architecture improvements for memory usage and performance
34 * Media-only IAX2 transfers
35 * Updates to the Radio Repeater app code
36 * Deprecation of AgentCallbackLogin in favor of a dialplan-based solution
37 * uClibc builds supported
38 * Work done for freeBSD portability
39 * Work done for Solaris portability
40 * FreeTDS-based database can be used with Realtime
41 * New internal data structure, stringfields, is implemented in IAX and SIP, reducing memory consumption by about 50%.
42 * Use of thread local storage for reduced memory allocation/freeing and lower stack consumption
43 * Reorganized files into docs/ main/ configs/, including name changes in some cases
44 * Much effort was expended in arranging documentation in source files in doxygen format
45 * Improved IP TOS support for IAX and SIP
46 * Builtin mini HTTP server
47 * Added support for Sigma Designs cards.
48 * Frame header caching to reduce memory allocation/freeing
49 * using mpg123 to play MP3 files for music-on-hold will be deprecated in 1.4 (start using the "native support")
50 * New Apps:
51 1. AMD() ;; Answering Machine Detection
52 2. ChannelRedirect() ;; asynch goto, redirect chan to context/exten/priority
53 3. ContinueWhile() ;; Addition to the While() suite. Acts like "continue".
54 4. ExitWhile() ;; Addition to the While() suite. Acts like "break".
55 5. ExtenSpy() ;; A close cousin to ChanSpy().
56 6. FollowMe() ;; findme/followme call redirect app
57 7. Log() ;; Send a message to the log, based on severity level.
58 8. MacroExclusive() ;; No more than one invocation of this macro allowed at any one time.
59 9. MorseCode() ;; turns strings into dits and dahs. A playground for ham radio licensees!
60 10. OSPAuth() ;; OSP authentication
61 11. QueueLog() ;; allows you to write your own events into the queue log
62 12. SLAStation() ;; Shared Line Appearance
63 13. SLATrunk() ;; Shared Line Appearance
64 14. SpeechCreate() ;; Voice Recognition Engine interface...
65 15. SpeechActivateGrammar()
66 16. SpeechStart()
67 17. SpeechBackground
68 18. SpeechDeactivateGrammar()
69 19. SpeechProcessingSound()
70 20. SpeechDestroy()
71 21. SpeechLoadGrammar()
72 22. SpeechUnloadGrammar()
73 23. StopMixMonitor() ;; to stop the MixMonitor App.
74 24. TryExec() ;; execute dialplan app without fatal consequences
75 * Apps removed:
76 1. CheckGroup -- do a comparison to ${GROUP()}
77 2. Curl -- use the function CURL() instead
78 3. Cut -- use the function CUT() instead
79 4. DateTime -- use sayunixtime() app instead.
80 5. DBget -- deprecated in 1.2, now removed.
81 6. DBput -- deprecated in 1.2, now removed.
82 7. Enumlookup -- use the function ENUMLOOKUP() instead
83 8. Eval -- use the function EVAL() instead
84 9. GetGroupCount -- use the function GROUP_COUNT() instead
85 10. GetGroupMatchCount -- use the function GROUP_MATCH_COUNT() instead
86 11. Intercom -- use the chan_oss module instead
87 12. Math -- use the function MATH() instead
88 13. MD5 -- use the function MD5() instead
89 14. SetCIDname -- use the function CALLERID(name) instead
90 15. SetCIDnum -- use the function CALLERID(number) instead
91 16. SetGroup -- use Set(GROUP=group) instead
92 17. SetRDNIS -- use the function CALLERID(rdnis) instead
93 18. Sql_postgres -- was deprecated in 1.2, now removed
94 19. Txtcidname -- use the function TXTCIDNAME instead
95 * New Funcs:
96 1. ARRAY()
97 2. BASE_64_DECODE()
98 3. BASE_64_ENCODE()
99 4. CHANNEL()
100 5. CURL()
101 6. CUT()
102 7. DB_DELETE()
103 8. FILTER()
104 9. GLOBAL()
105 10. IFTIME()
106 11. KEYPADHASH()
107 12. ODBC()
108 13. QUOTE()
109 14. RAND()
110 15. REALTIME()
111 16. SHA1()
112 17. SORT()
113 18. SPRINTF()
114 19. SQL_ESC()
115 20. STAT()
116 21. STRPTIME()
117 * Apps that have changes to their interface:
118 1. Authenticate() -- optional maxdigits argument added.
119 2. ChanSpy() -- new options:
120 o w -- Enable 'whisper' mode, so the spying channel can talk to...
121 o W -- Enable 'private whisper' mode, so the spying channel can...
122 3. DBdel() -- now marked as DEPRECATED in favor of the DB_DELETE func
123 4. Dial()
124 o New Option: O([x]) for Zaptel operator mode
125 o New Option: K/k parking via dtmf tones
126 5. Dictate() -- optional filename argument added.
127 6. Directory() -- new option: e - In addition to the name, also read the extension number...
128 7. Meetme() -- new options:
129 o 'I' -- announce user join/leave without review
130 o 'l' -- set listen only mode (Listen only, no talking)
131 o 'o' -- set talker optimization - treats talkers who aren't speaking as...
132 o '1' -- do not play message when first person enters
133 8. MeetmeAdmin() -- new options:
134 o 'r' -- Reset one user's volume settings
135 o 'R' -- Reset all users volume settings
136 o 's' -- Lower entire conference speaking volume
137 o 'S' -- Raise entire conference speaking volume
138 o 't' -- Lower one user's talk volume
139 o 'T' -- Lower all users talk volume
140 o 'u' -- Lower one user's listen volume
141 o 'U' -- Lower all users listen volume
142 o 'v' -- Lower entire conference listening volume
143 o 'V' -- Raise entire conference listening volume
144 9. OSPFinish() : now also can return ERROR result.
145 10. OSPLookup() : Sets more variables, also now returns ERROR result.
146 11. Page() -- New option: r - record the page into a file (see 'r' for app_meetme)
147 12. Pickup() -- multiple extensions, PICKUPMARK; read the description!
148 13. Queue()
149 o New Argument: AGI
150 o New option: i
151 14. Random() -- is now deprecated in 1.4
152 15. Read() -- replace 'skip' and 'noanswer' options with 's', 'n', add 'i' option.
153 16. Record() -- New option: 'x' : ignore all terminator keys (DTMF) and keep recording until hangup
154 17. UserEvent() -- slight change in behavior. Read the description.
155 18. VoiceMailMain() -- new a(#) option, goes to folder # directly.
156 19. WaitForSilence() -- new optional 3rd arg, time delay before returning.
157 * Funcs that have changes to their interfaces:
158 1. CDR -- new option: u
159 2. LANGUAGE -- Deprecated. Use CHANNEL(language) instead.
160 3. MUSICCLASS -- Deprecated. Use CHANNEL(musicclass) instead.
161 * Config File Changes:
162 1. NEW config files:
163 1. amd.conf -- Answering Machine Detection parameters
164 2. followme.conf -- parameters for the findme/followme call forwarding
165 3. func_odbc.conf -- define sql access functions here
166 4. gtalk.conf -- how to handle gtalk protocol calls
167 5. h323.conf -- h323 configuration
168 6. http.conf -- config for the builtin mini-http server in asterisk
169 7. jabber.conf -- jabber interface
170 8. jingle.conf -- jingle protocol interface config
171 10. res_snmp.conf -- to enable snmp in asterisk, and define full/sub agent status
172 11. say.conf -- define per-language rules for numbers, dates, etc.
173 12. skinny.conf -- for those special skinny phones you want to use...
174 13. sla.conf -- Shared Line Appearance config
175 14. smdi.conf -- SMDI messaging config
176 15. udptl.conf -- T38's udptl transport config
177 16. users.conf -- user config
178 2. Changes to Existing Config files:
179 1. In General:
180 o Jitterbuffer support added to several channels. Usually adds these variables to a config file:
181 1. jbenable
182 2. jbmaxsize
183 3. jbresyncthreshold
184 4. jbimpl
185 5. jblog
186 o MusicOnHold upgrade introduces two new variables:
187 1. mohinterpret
188 2. mohsuggest
189 2. agents.conf
190 o maxlogintries variable added
191 o autologoffunavail variable added
192 o endcall variable added
193 o agentgoodbye variable added
194 o createlink variable REMOVED
195 3. alsa.conf
196 o mohinterpret variable added
197 o Jitterbuffer variables added
198 4. cdr.conf
199 o endbeforehexten variable added
200 o sections for csv and radius added, with variables usegmtime, loguniqueid,
201 loguserfield, and radiuscfg variables.
202 5. cdr_tds.conf
203 o table variable added
204 6. extensions.ael
205 o Many upgrades. See the info at http://www.voip-info.org/wiki/view/Asterisk+AEL2
206 7. extensions.conf
207 o autofallthru now set to "yes" by default
208 o userscontext variable added
209 o added info/examples on paging and hints.
210 8. features.conf
211 o parkedplay variable added (who to beep at)
212 o parkedmusicclass
213 o atxfernoanswertimeout variable added
214 o parkcall variable added (one step parking)
215 o improved documentation for dynamic feature declarations!
216 9. iax.conf
217 o adsi variable added
218 o mohinterpret variable added
219 o mohsuggest variable added
220 o jitterbuffer updates
221 o iaxthreadcount variable added
222 o iaxmaxthreadcount variable added
223 o the way to specify TOS has changed.
224 o mailboxdetail variable has been REMOVED.
225 10. indications.conf
226 o [bg] entry added (Bulgaria).
227 o [il] entry added (Israel)
228 o [in] entry added (India)
229 o [jp] entry added (Japan)
230 o [my] entry added (Malaysia)
231 o [th] entry added (Thailand)
232 11. manager.conf
233 o displaysystemname variable added
234 o webenabled variable added
235 o httptimeout variable added
236 o timestampevents variable added
237 12. mgcp.conf
238 o Jitterbuffer support added
239 13. misdn.conf
240 o l1watcher_timeout variable added
241 o pp_l2_check variable added
242 o echocancelwhenbridged variable added
243 o echotraining variable added
244 o max_incoming variable added
245 o max_outgoing variable added
246 14. modules.conf
247 o a comment for preloading res_speech.so is added
248 o mention of global symbols is removed
249 o obsolesced entries for chan_modem_* and app_intercom have been removed
250 15. musiconhold.conf
251 o the default is now to do native moh from /var/lib/asterisk/moh
252 16. osp.conf
253 o authpolicy variable added
254 17. oss.conf
255 o debug variable added
256 o device variable added
257 o mixer variable added
258 o boost variable added
259 o callerid variable added
260 o autohangup variable added
261 o queuesize variable added
262 o frags variable added
263 o JitterBuffer support
264 o sections to define alternate sound cards
265 18. queues.conf
266 o autofill variable added
267 o monitor-type variable added
268 o musiconhold is now musicclass, with a difference in interpretation
269 o autofill variable added
270 o autopause variable added
271 o setinterfacevar variable added
272 o ringinuse variable added
273 19. res_odbc.conf
274 o pooling variable added
275 20. rpt.conf
276 o duplex variable added
277 o tailmessagetime variable added
278 o tailsquashedtime variable added
279 o tailmessages variable added
280 21. rtp.conf
281 o rtcpinterval varaible added
282 22. sip.conf
283 o allowoverlap variable added
284 o allowtransfer variable added
285 o tos variable REMOVED
286 o tos_sip variable added
287 o tos_audio variable added
288 o tos_video variable added
289 o minexpiry variable added
290 o t1min variable added
291 o musicclass variable REMOVED
292 o mohinterpret variable added
293 o maxcallbitratesuggest variable added
294 o allowsubscribe variable added
295 o videosupport variable added
296 o maxcallbitrate variable added
297 o g726nonstandard variable added
298 o dumphistory variable added
299 o allowsubscribe variable added
300 o t38pt_udptl variable added
301 o canreinvite variable can also now be set to 'nonat'
302 o rtsavesysname variable added
303 o JitterBuffer support added
304 23. skinny.conf
305 o port variable renamed to bindport
306 o JitterBuffer support added
307 o model variable REMOVED
308 o mohinterpret variable added
309 o mohsuggest variable added
310 o speeddial variable added
311 o addon variable added
312 24. voicemail.conf
313 o userscontext variable added
314 o smdiport variable added
315 o attachfmt variable added
316 o volgain variable added
317 o tempgreetwarn variable added
318 25. zapata.conf
319 o pritimer variable has improved documentation
320 o New signalling method: fgccama
321 o New signalling method: fgccamamf
322 o outsignalling variable added
323 o distinctiveringaftercid variable added
324 o cidsignalling now also accepts v23_jp, and smdi
325 o usesmdi variable added
326 o smdiport variable added
327 o mohinterpret variable added
328 o mohsuggest variable added
329 o JitterBuffer support added
330 * Removed Codecs/Channels:
331 1. codec_g723 was removed because the actual codec implementation it was designed to use is not distributable
332 2. chan_modem_* stuff is gone because the kernel support for those interfaces is old, buggy and unsupported
333 * New Utils:
334 1. aelparse -- compile .ael files outside of asterisk