1 ------------------------------------------------------------------------------
2 --- Functionality changes since Asterisk 1.4-beta was branched ----------------
3 -------------------------------------------------------------------------------
5 AMI - The manager (TCP/TLS/HTTP)
6 --------------------------------
7 * Added a new action 'CoreShowChannels' to list currently defined channels
8 and some information about them.
9 * Added a new action 'SIPshowregistry' to list SIP registrations.
10 * Added TLS support for the manager interface and HTTP server
11 * Added the URI redirect option for the built-in HTTP server
12 * The output of CallerID in Manager events is now more consistent.
13 CallerIDNum is used for number and CallerIDName for name.
14 * Enable https support for builtin web server.
15 See configs/http.conf.sample for details.
16 * Added a new action, GetConfigJSON, which can return the contents of an
17 Asterisk configuration file in JSON format. This is intended to help
18 improve the performance of AJAX applications using the manager interface
20 * SIP and IAX manager events now use "ChannelType" in all cases where we
21 indicate channel driver. Previously, we used a mixture of "Channel"
22 and "ChannelDriver" headers.
23 * Added a "Bridge" action which allows you to bridge any two channels that
24 are currently active on the system.
25 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
26 the voicemail users setup.
27 * Added 'DBDel' and 'DBDelTree' manager commands.
28 * cdr_manager now reports events via the "cdr" level, separating it from
29 the very verbose "call" level.
30 * Manager users are now stored in memory. If you change the manager account
31 list (delete or add accounts) you need to reload manager.
32 * Added Masquerade manager event for when a masquerade happens between
34 * Added "manager reload" command for the CLI
38 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
39 state in the dialplan, as well as creating custom device states that are
40 controllable from the dialplan.
41 * Extend CALLERID() function with "pres" and "ton" parameters to
42 fetch string representation of calling number presentation indicator
43 and numeric representation of type of calling number value.
44 * MailboxExists converted to dialplan function
45 * A new option to Dial() for telling IP phones not to count the call
46 as "missed" when dial times out and cancels.
47 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
48 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
49 held for any given channel. Also, locks are automatically freed when a
51 * Added HINT() dialplan function that allows retrieving hint information.
52 Hints are mappings between extensions and devices for the sake of
53 determining the state of an extension. This function can retrieve the list
54 of devices or the name associated with a hint.
55 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
57 * Added SYSINFO() dialplan function which allows retrieval of system information
61 * New CLI command "core show hint" (usage: core show hint <exten>)
62 * New CLI command "core show settings"
63 * Added 'core show channels count' CLI command.
64 * Added the ability to set the core debug and verbose values on a per-file basis.
65 * Added 'queue pause member' and 'queue unpause member' CLI commands
66 * Ability to set process limits ("ulimit") without restarting Asterisk
67 * Enhanced "agi debug" to print the channel name as a prefix to the debug
68 output to make debugging on busy systems much easier.
69 * New CLI commands "dialplan set extenpatternmatching true/false"
73 * Improved NAT and STUN support.
74 chan_sip now can use port numbers in bindaddr, externip and externhost
75 options, as well as contact a STUN server to detect its external address
76 for the SIP socket. See sip.conf.sample, 'NAT' section.
77 * The default SIP useragent= identifier now includes the Asterisk version
78 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
79 If set, and the incoming request carries authentication info,
80 the username to match in the users list is taken from the Digest header
81 rather than from the From: field. This feature is considered experimental.
82 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
83 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
84 * The "localmask" setting was removed in version 1.2 and the reminder about it
85 being removed is now also removed.
86 * A new option "busylevel" for setting a level of calls where asterisk reports
87 a device as busy, to separate it from call-limit. This value is also added
88 to the SIP_PEER dialplan function.
89 * A new realtime family called "sipregs" is now supported to store SIP registration
90 data. If this family is defined, "sippeers" will be used for configuration and
91 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
92 registration data, as before.
93 * The SIPPEER function have new options for port address, call and pickup groups
94 * Added support for T.140 realtime text in SIP/RTP
95 * The "checkmwi" option has been removed from sip.conf, as it is no longer
96 required due to the restructuring of how MWI is handled. See the descriptions
97 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
99 * Added rtpdest option to CHANNEL() dialplan function.
100 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
101 * SIP now adds a header to the CANCEL if the call was answered by another phone
102 in the same dial command, or if the new c option in dial() is used.
103 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
104 states it is not needed. For phones, however, that do require it the "registertrying" option
105 has been added so it can be enabled.
106 * A new option called "callcounter" (global/peer/user level) enables call counters needed
107 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
108 used to enable this functionality).
109 * New settings for timer T1 and timer B on a global level or per device. This makes it
110 possible to force timeout faster on non-responsive SIP servers. These settings are
111 considered advanced, so don't use them unless you have a problem.
112 * Added a dial string option to be able to set the To: header in an INVITE to any
117 * Added the trunkmaxsize configuration option to chan_iax2.
118 * Added the srvlookup option to iax.conf
119 * Added support for OSP. The token is set and retrieved through the CHANNEL()
122 XMPP Google Talk/Jingle changes
123 -------------------------------
124 * Added the bindaddr option to gtalk.conf.
128 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
129 * Proper codec support in chan_skinny.
130 * Added settings for IP and Ethernet QoS requests
134 * Added separate settings for media QoS in mgcp.conf
138 * Added experimental support for video under X windows
142 * Added the ability to specify arguments to the Dial application when using
143 the DUNDi switch in the dialplan.
144 * Added the ability to set weights for responses dynamically. This can be
145 done using a global variable or a dialplan function. Using the SHELL()
146 function would allow you to have an external script set the weight for
148 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
149 functions will allow you to initiate a DUNDi query from the dialplan,
150 find out how many results there are, and access each one.
154 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
155 functions will allow you to initiate an ENUM lookup from the dialplan,
156 and Asterisk will cache the results. ENUMRESULT can be used to access
157 the results without doing multiple DNS queries.
161 * Added the ability to customize which sound files are used for some of the
162 prompts within the Voicemail application by changing them in voicemail.conf
163 * Added the ability for the "voicemail show users" CLI command to show users
164 configured by the dynamic realtime configuration method.
165 * MWI (Message Waiting Indication) handling has been significantly
166 restructured internally to Asterisk. It is now totally event based
167 instead of polling based. The voicemail application will notify other
168 modules that have subscribed to MWI events when something in the mailbox
170 This also means that if any other entity outside of Asterisk is changing
171 the contents of mailboxes, then the voicemail application still needs to
172 poll for changes. Examples of situations that would require this option
173 are web interfaces to voicemail or an email client in the case of using
174 IMAP storage. So, two new options have been added to voicemail.conf
175 to account for this: "pollmailboxes" and "pollfreq". See the sample
176 configuration file for details.
177 * Added "tw" language support
178 * Added support for storage of greetings using an IMAP server
179 * Added ability to customize forward, reverse, stop, and pause keys for message playback
180 * SMDI is now enabled in voicemail using the smdienable option.
181 * A "lockmode" option has been added to asterisk.conf to configure the file
182 locking method used for voicemail, and potentially other things in the
183 future. The default is the old behavior, lockfile. However, there is a
184 new method, "flock", that uses a different method for situations where the
185 lockfile will not work, such as on SMB/CIFS mounts.
189 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
190 used across multiple queues.
191 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
192 setqueueentryvar options for each queue, see queues.conf.sample for details.
193 * Added keepstats option to queues.conf which will keep queue
194 statistics during a reload.
195 * setinterfacevar option in queues.conf also now sets a variable
196 called MEMBERNAME which contains the member's name.
197 * Added 'Strategy' field to manager event QueueParams which represents
198 the queue strategy in use.
199 * Added option to run macro when a queue member is connected to a caller,
200 see queues.conf.sample for details.
201 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
202 does not count paused queue members as unavailable.
203 * Added min-announce-frequency option to queues.conf which allows you to control the
204 minimum amount of time between queue announcements for use when the caller's queue
205 position changes frequently.
206 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
208 * Added ability for non-realtime queues to have realtime members
209 * Added the "linear" strategy to queues.
210 * Added the "wrandom" strategy to queues.
214 * The 'o' option to provide an optimization has been removed and its functionality
215 has been enabled by default.
216 * When a conference is created, the UNIQUEID of the channel that caused it to be
217 created is stored. Then, every channel that joins the conference will have the
218 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
219 callers that come and go from long standing conferences.
220 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
221 except it does operations on a channel by name, instead of number in a conference.
222 This is a very useful feature in combination with the 'X' option to ChanSpy.
223 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
225 * Added new RealTime functionality to provide support for scheduled conferencing.
226 This includes optional messages to the caller if they attempt to join before
227 the schedule start time, or to allow the caller to join the conference early.
228 Also included is optional support for limiting the number of callers per
230 * Added the S() and L() options to the MeetMe application. These are pretty
231 much identical to the S() and L() options to Dial(). They let you set
232 timeouts for the conference, as well as have warning sounds played to
233 let the caller know how much time is left, and when it is running out.
234 * Added the ability to do "meetme concise" with the "meetme" CLI command.
235 This extends the concise capabilities of this CLI command to include
236 listing all conferences, instead of an addition to the other sub commands
237 for the "meetme" command.
238 * Added the ability to specify the music on hold class used to play into the
239 conference when there is only one member and the M option is used.
241 Other Dialplan Application Changes
242 ----------------------------------
243 * Argument support for Gosub application
244 * From the to-do lists: straighten out the app timeout args:
245 Wait() app now really does 0.3 seconds- was truncating arg to an int.
246 WaitExten() same as Wait().
247 Congestion() - Now takes floating pt. argument.
248 Busy() - now takes floating pt. argument.
249 Read() - timeout now can be floating pt.
250 WaitForRing() now takes floating pt timeout arg.
251 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
252 * Added 's' option to Page application.
253 * Added 'E' and 'V' commands to ExternalIVR.
254 * Added 'o' and 'X' options to Chanspy.
255 * Added a new dialplan application, Bridge, which allows you to bridge the
256 calling channel to any other active channel on the system.
257 * Added the ability to specify a music on hold class to play instead of ringing
258 for the SLATrunk application.
259 * The Read application no longer exits the dialplan on error. Instead, it sets
260 READSTATUS to ERROR, which you can catch and handle separately.
261 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
262 of asking for verification of each name, one at a time.
263 * Privacy() no longer uses privacy.conf, as all options are specifyable as
264 direct options to the app.
266 Music On Hold Changes
267 ---------------------
268 * A new option, "digit", has been added for music on hold classes in
269 musiconhold.conf. If this is set for a music on hold class, a caller
270 listening to music on hold can press this digit to switch to listening
271 to this music on hold class.
272 * Support for realtime music on hold has been added.
273 * In conjunction with the realtime music on hold, a general section has
274 been added to musiconhold.conf, its sole variable is cachertclasses. If this
275 is set, then music on hold classes found in realtime will be cached in memory.
279 * AEL upgraded to use the Gosub with Arguments instead
280 of Macro application, to hopefully reduce the problems
281 seen with the artificially low stack ceiling that
282 Macro bumps into. Macros can only call other Macros
283 to a depth of 7. Tests run using gosub, show depths
284 limited only by virtual memory. A small test demonstrated
285 recursive call depths of 100,000 without problems.
286 -- in addition to this, all apps that allowed a macro
287 to be called, as in Dial, queues, etc, are now allowing
288 a gosub call in similar fashion.
289 * AEL now generates LOCAL(argname) declarations when it
290 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
291 etc. That makes the arguments local in scope. The user
292 can define their own local variables in macros, now,
293 by saying "local myvar=someval;" or using Set() in this
294 fashion: Set(LOCAL(myvar)=someval); ("local" is now
296 * utils/conf2ael introduced. Will convert an extensions.conf
297 file into extensions.ael. Very crude and unfinished, but
298 will be improved as time goes by. Should be useful for a
299 first pass at conversion.
300 * aelparse will now read extensions.conf to see if a referenced
301 macro or context is there before issueing a warning.
303 Zaptel channel driver (chan_zap) Changes
304 ----------------------------------------
305 * SS7 support in chan_zap (via libss7 library)
306 * In India, some carriers transmit CID via dtmf. Some code has been added
307 that will handle some situations. The cidstart=polarity_IN choice has been added for
308 those carriers that transmit CID via dtmf after a polarity change.
309 * CID matching information is now shown when doing 'dialplan show'.
310 * Added zap show version CLI command to chan_zap.
311 * Added setvar support to zapata.conf channel entries.
312 * Added two new options: mwimonitor and mwimonitornotify. These options allow
313 you to enable MWI monitoring on FXO lines. When the MWI state changes,
314 the script specified in the mwimonitornotify option is executed. An internal
315 event indicating the new state of the mailbox is also generated, so that
316 the normal MWI facilities in Asterisk work as usual.
320 * H323 remote hold notification support added (by NOTIFY message
321 and/or H.450 supplementary service)
323 Call Features (res_features) Changes
324 ------------------------------------
325 * Added the parkedcalltransfers option to features.conf
326 * The built-in method for doing attended transfers has been updated to
327 include some new options that allow you to have the transferee sent
328 back to the person that did the transfer if the transfer is not successful.
329 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
330 in features.conf.sample.
331 * Added support for configuring named groups of custom call features in
332 features.conf. This means that features can be written a single time, and
333 then mapped into groups of features for different key mappings or easier
335 * Updated the ParkedCall application to allow you to not specify a parking
336 extension. If you don't specify a parking space to pick up, it will grab
337 the first one available.
339 Language Support Changes
340 ------------------------
341 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
342 * Added support for the Hungarian language for saying numbers, dates, and times.
346 * Added SPEECH commands for speech recognition. A complete listing can be found
351 * Ability to use libcap to set high ToS bits when non-root
352 on Linux. If configure is unable to find libcap then you
353 can use --with-cap to specify the path.
354 * Added rotatestrategy option to logger.conf, along with two new options:
355 "timestamp" which will use the time to name the logger files instead of
356 sequence number; and "rotate", which rotates the names of the logfiles,
357 similar to the way syslog rotates files.
358 * Added exec_after_rotate option to logger.conf, which allows a system
359 command to be run after rotation. This is primarily useful with
360 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
361 and to ensure that the oldest log file gets deleted.
362 * Added maxfiles option to options section of asterisk.conf which allows you to specify
363 what Asterisk should set as the maximum number of open files when it loads.
364 * Added the jittertargetextra configuration option.
365 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
366 * Added a new CDR module, cdr_sqlite3_custom.
367 * The cdr_manager module has a [mappings] feature, like cdr_custom,
368 to add fields to the manager event from the CDR variables.
369 * Added a new realtime configuration module, res_config_sqlite
370 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
371 configuration files for the IP channel drivers. The new option is "cos".
372 This information is also documented in doc/qos.tex, or the IP Quality of Service
373 section of asterisk.pdf.
374 * The device state functionality in the Local channel driver has been updated
375 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
376 to just UNKNOWN if the extension exists.
377 * When originating a call using AMI or pbx_spool that fails the reason for failure
378 will now be available in the failed extension using the REASON dialplan variable.
379 * Added jitterbuffer support for chan_local. This allows you to use the
380 generic jitterbuffer on incoming calls going to Asterisk applications.
381 For example, this would allow you to use a jitterbuffer for an incoming
382 SIP call to Voicemail by putting a Local channel in the middle. This
383 feature is enabled by using the 'j' option in the Dial string to the Local
384 channel in conjunction with the existing 'n' option for local channels.
385 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
386 It allows you to configure a prefix for auto-monitor recordings.
387 * Added support for writing and running your dialplan in lua. See
388 configs/extensions.lua.sample for examples of how to do this.
389 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
390 configs/unistim.conf.sample for details. This new channel driver allows
391 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
392 * A new extension pattern matching algorithm, based on a trie, is introduced
393 here, that could noticeably speed up mid-sized to large dialplans.
394 It is NOT used by default, as duplicating the behaviour of the old pattern
395 matcher is still under development. A config file option, in extensions.conf,
396 in the [general] section, called "extenpatternmatchingnew", is by default
397 set to false; setting that to true will force the use of the new algorithm.
398 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
399 be used to switch the algorithms at run time.
400 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
401 specifying which socket to use to connect to the running Asterisk daemon
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