1 ------------------------------------------------------------------------------
2 --- Functionality changes since Asterisk 1.4-beta was branched ----------------
3 -------------------------------------------------------------------------------
5 AMI - The manager (TCP/TLS/HTTP)
6 --------------------------------
7 * Manager has undergone a lot of changes, all of them documented
9 * Manager version has changed to 1.1
10 * Added a new action 'CoreShowChannels' to list currently defined channels
11 and some information about them.
12 * Added a new action 'SIPshowregistry' to list SIP registrations.
13 * Added TLS support for the manager interface and HTTP server
14 * Added the URI redirect option for the built-in HTTP server
15 * The output of CallerID in Manager events is now more consistent.
16 CallerIDNum is used for number and CallerIDName for name.
17 * Enable https support for builtin web server.
18 See configs/http.conf.sample for details.
19 * Added a new action, GetConfigJSON, which can return the contents of an
20 Asterisk configuration file in JSON format. This is intended to help
21 improve the performance of AJAX applications using the manager interface
23 * SIP and IAX manager events now use "ChannelType" in all cases where we
24 indicate channel driver. Previously, we used a mixture of "Channel"
25 and "ChannelDriver" headers.
26 * Added a "Bridge" action which allows you to bridge any two channels that
27 are currently active on the system.
28 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
29 the voicemail users setup.
30 * Added 'DBDel' and 'DBDelTree' manager commands.
31 * cdr_manager now reports events via the "cdr" level, separating it from
32 the very verbose "call" level.
33 * Manager users are now stored in memory. If you change the manager account
34 list (delete or add accounts) you need to reload manager.
35 * Added Masquerade manager event for when a masquerade happens between
37 * Added "manager reload" command for the CLI
38 * Lots of commands that only provided information are now allowed under the
39 Reporting privilege, instead of only under Call or System.
40 * The IAX* commands now require either System or Reporting privilege, to
41 mirror the privileges of the SIP* commands.
45 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
46 state in the dialplan, as well as creating custom device states that are
47 controllable from the dialplan.
48 * Extend CALLERID() function with "pres" and "ton" parameters to
49 fetch string representation of calling number presentation indicator
50 and numeric representation of type of calling number value.
51 * MailboxExists converted to dialplan function
52 * A new option to Dial() for telling IP phones not to count the call
53 as "missed" when dial times out and cancels.
54 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
55 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
56 held for any given channel. Also, locks are automatically freed when a
58 * Added HINT() dialplan function that allows retrieving hint information.
59 Hints are mappings between extensions and devices for the sake of
60 determining the state of an extension. This function can retrieve the list
61 of devices or the name associated with a hint.
62 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
64 * Added SYSINFO() dialplan function which allows retrieval of system information
65 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
66 the existence of a dialplan target.
67 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
68 upper and lower case, respectively.
72 * New CLI command "core show hint" (usage: core show hint <exten>)
73 * New CLI command "core show settings"
74 * Added 'core show channels count' CLI command.
75 * Added the ability to set the core debug and verbose values on a per-file basis.
76 * Added 'queue pause member' and 'queue unpause member' CLI commands
77 * Ability to set process limits ("ulimit") without restarting Asterisk
78 * Enhanced "agi debug" to print the channel name as a prefix to the debug
79 output to make debugging on busy systems much easier.
80 * New CLI commands "dialplan set extenpatternmatching true/false"
81 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
85 * Improved NAT and STUN support.
86 chan_sip now can use port numbers in bindaddr, externip and externhost
87 options, as well as contact a STUN server to detect its external address
88 for the SIP socket. See sip.conf.sample, 'NAT' section.
89 * The default SIP useragent= identifier now includes the Asterisk version
90 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
91 If set, and the incoming request carries authentication info,
92 the username to match in the users list is taken from the Digest header
93 rather than from the From: field. This feature is considered experimental.
94 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
95 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
96 * The "localmask" setting was removed in version 1.2 and the reminder about it
97 being removed is now also removed.
98 * A new option "busylevel" for setting a level of calls where asterisk reports
99 a device as busy, to separate it from call-limit. This value is also added
100 to the SIP_PEER dialplan function.
101 * A new realtime family called "sipregs" is now supported to store SIP registration
102 data. If this family is defined, "sippeers" will be used for configuration and
103 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
104 registration data, as before.
105 * The SIPPEER function have new options for port address, call and pickup groups
106 * Added support for T.140 realtime text in SIP/RTP
107 * The "checkmwi" option has been removed from sip.conf, as it is no longer
108 required due to the restructuring of how MWI is handled. See the descriptions
109 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
110 for more information.
111 * Added rtpdest option to CHANNEL() dialplan function.
112 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
113 * SIP now adds a header to the CANCEL if the call was answered by another phone
114 in the same dial command, or if the new c option in dial() is used.
115 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
116 states it is not needed. For phones, however, that do require it the "registertrying" option
117 has been added so it can be enabled.
118 * A new option called "callcounter" (global/peer/user level) enables call counters needed
119 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
120 used to enable this functionality).
121 * New settings for timer T1 and timer B on a global level or per device. This makes it
122 possible to force timeout faster on non-responsive SIP servers. These settings are
123 considered advanced, so don't use them unless you have a problem.
124 * Added a dial string option to be able to set the To: header in an INVITE to any
126 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
127 the qualify frequency.
131 * Added the trunkmaxsize configuration option to chan_iax2.
132 * Added the srvlookup option to iax.conf
133 * Added support for OSP. The token is set and retrieved through the CHANNEL()
136 XMPP Google Talk/Jingle changes
137 -------------------------------
138 * Added the bindaddr option to gtalk.conf.
142 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
143 * Proper codec support in chan_skinny.
144 * Added settings for IP and Ethernet QoS requests
148 * Added separate settings for media QoS in mgcp.conf
150 Console Channel Driver changes
152 * Added experimental support for video send & receive to chan_oss.
153 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
156 Phone channel changes (chan_phone)
157 ----------------------------------
158 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
160 H.323 channel Changes
161 ---------------------
162 * H323 remote hold notification support added (by NOTIFY message
163 and/or H.450 supplementary service)
165 Local channel changes
166 ---------------------
167 * The device state functionality in the Local channel driver has been updated
168 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
169 to just UNKNOWN if the extension exists.
170 * Added jitterbuffer support for chan_local. This allows you to use the
171 generic jitterbuffer on incoming calls going to Asterisk applications.
172 For example, this would allow you to use a jitterbuffer for an incoming
173 SIP call to Voicemail by putting a Local channel in the middle. This
174 feature is enabled by using the 'j' option in the Dial string to the Local
175 channel in conjunction with the existing 'n' option for local channels.
177 Zaptel channel driver (chan_zap) Changes
178 ----------------------------------------
179 * SS7 support in chan_zap (via libss7 library)
180 * In India, some carriers transmit CID via dtmf. Some code has been added
181 that will handle some situations. The cidstart=polarity_IN choice has been added for
182 those carriers that transmit CID via dtmf after a polarity change.
183 * CID matching information is now shown when doing 'dialplan show'.
184 * Added zap show version CLI command to chan_zap.
185 * Added setvar support to zapata.conf channel entries.
186 * Added two new options: mwimonitor and mwimonitornotify. These options allow
187 you to enable MWI monitoring on FXO lines. When the MWI state changes,
188 the script specified in the mwimonitornotify option is executed. An internal
189 event indicating the new state of the mailbox is also generated, so that
190 the normal MWI facilities in Asterisk work as usual.
191 * Added signalling type 'auto', which attempts to use the same signalling type
192 for a channel as configured in Zaptel. This is primarily designed for analog
193 ports, but will also work for digital ports that are configured for FXS or FXO
194 signalling types. This mode is also the default now, so if your zapata.conf
195 does not specify signalling for a channel (which is unlikely as the sample
196 configuration file has always recommended specifying it for every channel) then
197 the 'auto' mode will be used for that channel if possible.
198 * Added a 'zap set dnd' command to allow CLI control of the Do-Not-Disturb
199 state for a channel; also ensured that the DNDState Manager event is
200 emitted no matter how the DND state is set or cleared.
204 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
205 configs/unistim.conf.sample for details. This new channel driver allows
206 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
207 * Added a new channel driver, chan_console, which uses portaudio as a cross
208 platform audio interface. It was written as a channel driver that would
209 work with Mac CoreAudio, but portaudio supports a number of other audio
210 interfaces, as well. Note that this channel driver requires v19 or higher
211 of portaudio; older versions have a different API.
215 * Added the ability to specify arguments to the Dial application when using
216 the DUNDi switch in the dialplan.
217 * Added the ability to set weights for responses dynamically. This can be
218 done using a global variable or a dialplan function. Using the SHELL()
219 function would allow you to have an external script set the weight for
221 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
222 functions will allow you to initiate a DUNDi query from the dialplan,
223 find out how many results there are, and access each one.
227 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
228 functions will allow you to initiate an ENUM lookup from the dialplan,
229 and Asterisk will cache the results. ENUMRESULT can be used to access
230 the results without doing multiple DNS queries.
234 * Added the ability to customize which sound files are used for some of the
235 prompts within the Voicemail application by changing them in voicemail.conf
236 * Added the ability for the "voicemail show users" CLI command to show users
237 configured by the dynamic realtime configuration method.
238 * MWI (Message Waiting Indication) handling has been significantly
239 restructured internally to Asterisk. It is now totally event based
240 instead of polling based. The voicemail application will notify other
241 modules that have subscribed to MWI events when something in the mailbox
243 This also means that if any other entity outside of Asterisk is changing
244 the contents of mailboxes, then the voicemail application still needs to
245 poll for changes. Examples of situations that would require this option
246 are web interfaces to voicemail or an email client in the case of using
247 IMAP storage. So, two new options have been added to voicemail.conf
248 to account for this: "pollmailboxes" and "pollfreq". See the sample
249 configuration file for details.
250 * Added "tw" language support
251 * Added support for storage of greetings using an IMAP server
252 * Added ability to customize forward, reverse, stop, and pause keys for message playback
253 * SMDI is now enabled in voicemail using the smdienable option.
254 * A "lockmode" option has been added to asterisk.conf to configure the file
255 locking method used for voicemail, and potentially other things in the
256 future. The default is the old behavior, lockfile. However, there is a
257 new method, "flock", that uses a different method for situations where the
258 lockfile will not work, such as on SMB/CIFS mounts.
259 * Added the ability to backup deleted messages, to ease recovery in the case
260 that a user accidentally deletes a message, and discovers that they need it.
264 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
265 used across multiple queues.
266 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
267 setqueueentryvar options for each queue, see queues.conf.sample for details.
268 * Added keepstats option to queues.conf which will keep queue
269 statistics during a reload.
270 * setinterfacevar option in queues.conf also now sets a variable
271 called MEMBERNAME which contains the member's name.
272 * Added 'Strategy' field to manager event QueueParams which represents
273 the queue strategy in use.
274 * Added option to run macro when a queue member is connected to a caller,
275 see queues.conf.sample for details.
276 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
277 does not count paused queue members as unavailable.
278 * Added min-announce-frequency option to queues.conf which allows you to control the
279 minimum amount of time between queue announcements for use when the caller's queue
280 position changes frequently.
281 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
283 * Added ability for non-realtime queues to have realtime members
284 * Added the "linear" strategy to queues.
285 * Added the "wrandom" strategy to queues.
286 * Added new channel variable QUEUE_MIN_PENALTY
287 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
288 rules in queuerules.conf. See configs/queuerules.conf.sample for details
289 * Added a new parameter for member definition, called state_interface. This may be
290 used so that a member may be called via one interface but have a different interface's
291 device state reported.
295 * The 'o' option to provide an optimization has been removed and its functionality
296 has been enabled by default.
297 * When a conference is created, the UNIQUEID of the channel that caused it to be
298 created is stored. Then, every channel that joins the conference will have the
299 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
300 callers that come and go from long standing conferences.
301 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
302 except it does operations on a channel by name, instead of number in a conference.
303 This is a very useful feature in combination with the 'X' option to ChanSpy.
304 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
306 * Added new RealTime functionality to provide support for scheduled conferencing.
307 This includes optional messages to the caller if they attempt to join before
308 the schedule start time, or to allow the caller to join the conference early.
309 Also included is optional support for limiting the number of callers per
311 * Added the S() and L() options to the MeetMe application. These are pretty
312 much identical to the S() and L() options to Dial(). They let you set
313 timeouts for the conference, as well as have warning sounds played to
314 let the caller know how much time is left, and when it is running out.
315 * Added the ability to do "meetme concise" with the "meetme" CLI command.
316 This extends the concise capabilities of this CLI command to include
317 listing all conferences, instead of an addition to the other sub commands
318 for the "meetme" command.
319 * Added the ability to specify the music on hold class used to play into the
320 conference when there is only one member and the M option is used.
322 Other Dialplan Application Changes
323 ----------------------------------
324 * Argument support for Gosub application
325 * From the to-do lists: straighten out the app timeout args:
326 Wait() app now really does 0.3 seconds- was truncating arg to an int.
327 WaitExten() same as Wait().
328 Congestion() - Now takes floating pt. argument.
329 Busy() - now takes floating pt. argument.
330 Read() - timeout now can be floating pt.
331 WaitForRing() now takes floating pt timeout arg.
332 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
333 * Added 's' option to Page application.
334 * Added 'E' and 'V' commands to ExternalIVR.
335 * Added 'o' and 'X' options to Chanspy.
336 * Added a new dialplan application, Bridge, which allows you to bridge the
337 calling channel to any other active channel on the system.
338 * Added the ability to specify a music on hold class to play instead of ringing
339 for the SLATrunk application.
340 * The Read application no longer exits the dialplan on error. Instead, it sets
341 READSTATUS to ERROR, which you can catch and handle separately.
342 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
343 of asking for verification of each name, one at a time.
344 * Privacy() no longer uses privacy.conf, as all options are specifyable as
345 direct options to the app.
346 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
349 Music On Hold Changes
350 ---------------------
351 * A new option, "digit", has been added for music on hold classes in
352 musiconhold.conf. If this is set for a music on hold class, a caller
353 listening to music on hold can press this digit to switch to listening
354 to this music on hold class.
355 * Support for realtime music on hold has been added.
356 * In conjunction with the realtime music on hold, a general section has
357 been added to musiconhold.conf, its sole variable is cachertclasses. If this
358 is set, then music on hold classes found in realtime will be cached in memory.
362 * AEL upgraded to use the Gosub with Arguments instead
363 of Macro application, to hopefully reduce the problems
364 seen with the artificially low stack ceiling that
365 Macro bumps into. Macros can only call other Macros
366 to a depth of 7. Tests run using gosub, show depths
367 limited only by virtual memory. A small test demonstrated
368 recursive call depths of 100,000 without problems.
369 -- in addition to this, all apps that allowed a macro
370 to be called, as in Dial, queues, etc, are now allowing
371 a gosub call in similar fashion.
372 * AEL now generates LOCAL(argname) declarations when it
373 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
374 etc. That makes the arguments local in scope. The user
375 can define their own local variables in macros, now,
376 by saying "local myvar=someval;" or using Set() in this
377 fashion: Set(LOCAL(myvar)=someval); ("local" is now
379 * utils/conf2ael introduced. Will convert an extensions.conf
380 file into extensions.ael. Very crude and unfinished, but
381 will be improved as time goes by. Should be useful for a
382 first pass at conversion.
383 * aelparse will now read extensions.conf to see if a referenced
384 macro or context is there before issueing a warning.
386 Call Features (res_features) Changes
387 ------------------------------------
388 * Added the parkedcalltransfers option to features.conf
389 * The built-in method for doing attended transfers has been updated to
390 include some new options that allow you to have the transferee sent
391 back to the person that did the transfer if the transfer is not successful.
392 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
393 in features.conf.sample.
394 * Added support for configuring named groups of custom call features in
395 features.conf. This means that features can be written a single time, and
396 then mapped into groups of features for different key mappings or easier
398 * Updated the ParkedCall application to allow you to not specify a parking
399 extension. If you don't specify a parking space to pick up, it will grab
400 the first one available.
402 Language Support Changes
403 ------------------------
404 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
405 * Added support for the Hungarian language for saying numbers, dates, and times.
409 * Added SPEECH commands for speech recognition. A complete listing can be found
414 * Added rotatestrategy option to logger.conf, along with two new options:
415 "timestamp" which will use the time to name the logger files instead of
416 sequence number; and "rotate", which rotates the names of the logfiles,
417 similar to the way syslog rotates files.
418 * Added exec_after_rotate option to logger.conf, which allows a system
419 command to be run after rotation. This is primarily useful with
420 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
421 and to ensure that the oldest log file gets deleted.
422 * Added realtime support for the queue log
424 Miscellaneous New Modules
425 -------------------------
426 * Added a new CDR module, cdr_sqlite3_custom.
427 * Added a new realtime configuration module, res_config_sqlite
428 * Added a new codec translation module, codec_resample, which re-samples
429 signed linear audio between 8 kHz and 16 kHz to help support wideband
431 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
432 based on configuration templates that use Asterisk dialplan function and
433 variable substitution. It should be possible to create phone profiles and
434 templates that work for the majority of phones provisioned over http. It
435 is currently only intended to provision a single user account per phone.
436 An example profile and set of templates for Polycom phones is provided.
437 NOTE: Polycom firmware is not included, but should be placed in
438 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
439 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
440 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
441 provided; there is a JACK() application, and a JACK_HOOK() function. Both
442 interfaces create an input and output JACK port. The application makes
443 these ports the endpoint of the call. The audio coming from the channel
444 goes out the output port and whatever comes back in on the input port is
445 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
446 audiohook on the channel. This lets you run the audio coming from a
447 channel through JACK, and whatever comes back in is what gets forwarded
448 on as the channel's audio. This is very useful for building custom
449 vocoders or doing recording or analysis of the channel's audio in another
454 * Ability to use libcap to set high ToS bits when non-root
455 on Linux. If configure is unable to find libcap then you
456 can use --with-cap to specify the path.
457 * Added maxfiles option to options section of asterisk.conf which allows you to specify
458 what Asterisk should set as the maximum number of open files when it loads.
459 * Added the jittertargetextra configuration option.
460 * The cdr_manager module has a [mappings] feature, like cdr_custom,
461 to add fields to the manager event from the CDR variables.
462 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
463 configuration files for the IP channel drivers. The new option is "cos".
464 This information is also documented in doc/qos.tex, or the IP Quality of Service
465 section of asterisk.pdf.
466 * When originating a call using AMI or pbx_spool that fails the reason for failure
467 will now be available in the failed extension using the REASON dialplan variable.
468 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
469 It allows you to configure a prefix for auto-monitor recordings.
470 * Added support for writing and running your dialplan in lua. See
471 configs/extensions.lua.sample for examples of how to do this.
472 * A new extension pattern matching algorithm, based on a trie, is introduced
473 here, that could noticeably speed up mid-sized to large dialplans.
474 It is NOT used by default, as duplicating the behaviour of the old pattern
475 matcher is still under development. A config file option, in extensions.conf,
476 in the [general] section, called "extenpatternmatchingnew", is by default
477 set to false; setting that to true will force the use of the new algorithm.
478 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
479 be used to switch the algorithms at run time.
480 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
481 specifying which socket to use to connect to the running Asterisk daemon
483 * Added logging to 'make update' command. See update.log
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