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1 ------------------------------------------------------------------------------
2 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
3 ------------------------------------------------------------------------------
5 Dialplan Functions
6 ------------------
7 * Added a new dialplan function, AST_CONFIG(), which allows you to access
8 variables from an Asterisk configuration file.
9 * The JACK_HOOK function now has a c() option to supply a custom client name.
10 * Added two new dialplan functions from libspeex for audio gain control and
11 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
12 rx directions of a channel from the dialplan.
13 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
14 based on other parameters. The default is still to search based on the
15 forwarding station ID. However, there are new options that allow you to search
16 based on the message desk terminal ID, or the message desk number.
17 * TIMEOUT() has been modified to be accurate down to the millisecond.
18 * ENUM*() functions now include the following new options:
19 - 'u' returns the full URI and does not strip off the URI-scheme.
20 - 's' triggers ISN specific rewriting
21 - 'i' looks for branches into an Infrastructure ENUM tree
22 - 'd' for a direct DNS lookup without any flipping of digits.
23 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
24 * CHANNEL() now has options for the maximum, minimum, and standard or normal
25 deviation of jitter, rtt, and loss for a call using chan_sip.
27 Zaptel channel driver (chan_zap) Changes
28 ----------------------------------------
29 * Channels can now be configured using named sections in zapata.conf, just
30 like other channel drivers, including the use of templates.
31 * The default for pridialplan has changed from 'national' to 'unknown'.
33 PBX Changes
34 -----------
35 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
36 to something that matches the pattern a hint will be created using the contents
37 and variables evaluated.
38 * Dialplan matching has been extended to allow an extension to return to the
39 PBX core to wait for more digits. This is done by using the new dialplan
40 application called "Incomplete". This will permit a whole new level of
41 extension control, by giving the administrator more control over early
42 matches employing one of the short-circuit pattern match operators. Note
43 that custom applications can trigger this same behavior by returning the
44 special value AST_PBX_INCOMPLETE.
46 Application Changes
47 -------------------
48 * Directory now permits both first and last names to be matched at the same
49 time. In addition, the number of digits to enter of the name can be set in
50 the arguments to Directory; previously, you could enter only 3, regardless
51 of how many names are in your company. For large companies, this should be
52 quite helpful.
53 * Voicemail now permits a mailbox setting to wrap around from first to last
54 messages, if the "messagewrap" option is set to a true value.
55 * Voicemail now permits an external script to be run, for password validation.
56 The script should output "VALID" or "INVALID" on stdout, depending upon the
57 wish to validate or invalidate the password given. Arguments are:
58 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
59 more details
60 * Dial has a new option: F(context^extension^pri), which permits a callee to
61 continue in the dialplan, at the specified label, if the caller hangs up.
62 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
63 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
64 * The Jack application now has a c() option to supply a custom client name.
65 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
66 like the pre-existing whisper mode, except that the spy can also talk to the
67 participant on the bridged channel as well.
68 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
69 to be spoken instead of the channel name or number. For more information on the
70 use of this option, issue the command "core show application ChanSpy" from the
71 Asterisk CLI.
72 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
73 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
74 words, if using the 'd' option, it is not possible to enter a number to append to
75 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
76 change to whisper mode, and pressing 6 will change to barge mode.
77 * ExternalIVR now takes several options that affect the way it performs, as
78 well as having several new commands. Please see doc/externalivr.txt for the
79 complete documentation.
80 * ChanIsAvail has a new option, 'a', which will return all available channels instead
81 of just the first one if you give the function more then one channel to check.
82 * PrivacyManager now takes an option where you can specify a context where the
83 given number will be matched. This way you have more control over who is allowed
84 and it stops the people who blindly enter 10 digits.
86 SIP Changes
87 -----------
88 * The ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using setvar to cause a given
89 audio file to be played upon completion of an attended transfer.
90 * Added DNS manager support to registrations for peers referencing peer entries.
91 DNS manager runs in the background which allows DNS lookups to be run asynchronously
92 as well as periodically updating the IP address. These properties allow for
93 better performance as well as recovery in the event of an IP change.
94 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
95 load/reload of large numbers of peers/users by ~40x (for large lists of peers.
96 Initially, we saw 4x improvement in call setup/destruction, but at the time
97 of merging, this gain has disappeared; further research will be done to try
98 and restore this performance improvement. Astobj2 refcounting is now used
99 for users, peers, and dialogs. Users are encouraged to assist in regression
100 testing and problem reporting!
101 * Added ability to specify registration expiry time on a per registration basis in
102 the register line.
103 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
104 lost packets.
105 * Added t38pt_usertpsource option. See sip.conf.sample for details.
106 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
108 IAX Changes
109 -----------
110 * Existing DNS manager lookups extended to check for SRV records.
112 CLI Changes
113 -----------
114 * New CLI command, "config reload <file.conf>" which reloads any module that
115 references that particular configuration file. Also added "config list"
116 which shows which configuration files are in use.
117 * New CLI commands, "pri show version" and "ss7 show version" that will
118 display which version of libpri and libss7 are being used, respectively.
119 A new API call was added so trunk will now have to be compiled against
120 a versions of libpri and libss7 that have them or it will not know that
121 these libraries exist.
123 DNS manager changes
124 -------------------
125 * Addresses managed by DNS manager now can check to see if there is a DNS
126 SRV record for a given domain and will use that hostname/port if present.
128 AMI - The manager (TCP/TLS/HTTP)
129 --------------------------------
130 * The Status command now takes an optional list of variables to display
131 along with channel status.
133 ODBC Changes
134 ------------
135 * res_odbc no longer has a limit of 1023 total possible unshared connections,
136 as some people were running into this limit. This limit has been increased
137 to 4.2 billion.
139 ------------------------------------------------------------------------------
140 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
141 ------------------------------------------------------------------------------
143 AMI - The manager (TCP/TLS/HTTP)
144 --------------------------------
145 * Manager has undergone a lot of changes, all of them documented
146 in doc/manager_1_1.txt
147 * Manager version has changed to 1.1
148 * Added a new action 'CoreShowChannels' to list currently defined channels
149 and some information about them.
150 * Added a new action 'SIPshowregistry' to list SIP registrations.
151 * Added TLS support for the manager interface and HTTP server
152 * Added the URI redirect option for the built-in HTTP server
153 * The output of CallerID in Manager events is now more consistent.
154 CallerIDNum is used for number and CallerIDName for name.
155 * Enable https support for builtin web server.
156 See configs/http.conf.sample for details.
157 * Added a new action, GetConfigJSON, which can return the contents of an
158 Asterisk configuration file in JSON format. This is intended to help
159 improve the performance of AJAX applications using the manager interface
160 over HTTP.
161 * SIP and IAX manager events now use "ChannelType" in all cases where we
162 indicate channel driver. Previously, we used a mixture of "Channel"
163 and "ChannelDriver" headers.
164 * Added a "Bridge" action which allows you to bridge any two channels that
165 are currently active on the system.
166 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
167 the voicemail users setup.
168 * Added 'DBDel' and 'DBDelTree' manager commands.
169 * cdr_manager now reports events via the "cdr" level, separating it from
170 the very verbose "call" level.
171 * Manager users are now stored in memory. If you change the manager account
172 list (delete or add accounts) you need to reload manager.
173 * Added Masquerade manager event for when a masquerade happens between
174 two channels.
175 * Added "manager reload" command for the CLI
176 * Lots of commands that only provided information are now allowed under the
177 Reporting privilege, instead of only under Call or System.
178 * The IAX* commands now require either System or Reporting privilege, to
179 mirror the privileges of the SIP* commands.
180 * Added ability to retrieve list of categories in a config file.
181 * Added ability to retrieve the content of a particular category.
182 * Added ability to empty a context.
183 * Created new action to create a new file.
184 * Updated delete action to allow deletion by line number with respect to category.
185 * Added new action insert to add new variable to category at specified line.
186 * Updated action newcat to allow new category to be inserted in file above another
187 existing category.
188 * Added new event "JitterBufStats" in the IAX2 channel
189 * Originate now requires the Originate privilege and, if you want to call out
190 to a subshell, it requires the System privilege, as well. This was done to
191 enhance manager security.
192 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
193 * New command: Atxfer. See doc/manager_1_1.txt for more details or
194 manager show command Atxfer from the CLI
196 Dialplan functions
197 ------------------
198 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
199 state in the dialplan, as well as creating custom device states that are
200 controllable from the dialplan.
201 * Extend CALLERID() function with "pres" and "ton" parameters to
202 fetch string representation of calling number presentation indicator
203 and numeric representation of type of calling number value.
204 * MailboxExists converted to dialplan function
205 * A new option to Dial() for telling IP phones not to count the call
206 as "missed" when dial times out and cancels.
207 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
208 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
209 held for any given channel. Also, locks are automatically freed when a
210 channel is hung up.
211 * Added HINT() dialplan function that allows retrieving hint information.
212 Hints are mappings between extensions and devices for the sake of
213 determining the state of an extension. This function can retrieve the list
214 of devices or the name associated with a hint.
215 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
216 of any extension.
217 * Added SYSINFO() dialplan function which allows retrieval of system information
218 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
219 the existence of a dialplan target.
220 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
221 upper and lower case, respectively.
222 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
223 ID for the call (not the Asterisk call ID or unique ID), provided that the
224 channel driver supports this. For SIP, you get the SIP call-ID for the
225 bridged channel which you can store in the CDR with a custom field.
227 CLI Changes
228 -----------
229 * New CLI command "core show hint" (usage: core show hint <exten>)
230 * New CLI command "core show settings"
231 * Added 'core show channels count' CLI command.
232 * Added the ability to set the core debug and verbose values on a per-file basis.
233 * Added 'queue pause member' and 'queue unpause member' CLI commands
234 * Ability to set process limits ("ulimit") without restarting Asterisk
235 * Enhanced "agi debug" to print the channel name as a prefix to the debug
236 output to make debugging on busy systems much easier.
237 * New CLI commands "dialplan set extenpatternmatching true/false"
238 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
239 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
240 listed in the startup_commands section of cli.conf will get executed.
241 * Added a CLI command, "devstate change", which allows you to set custom device
242 states from the func_devstate module that provides the DEVICE_STATE() function
243 and handling of the "Custom:" devices.
244 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
245 sorted into the different possible callbacks, with the number of entries
246 currently scheduled for each. Gives you a feel for how busy the sip channel
247 driver is.
249 SIP changes
250 -----------
251 * Improved NAT and STUN support.
252 chan_sip now can use port numbers in bindaddr, externip and externhost
253 options, as well as contact a STUN server to detect its external address
254 for the SIP socket. See sip.conf.sample, 'NAT' section.
255 * The default SIP useragent= identifier now includes the Asterisk version
256 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
257 If set, and the incoming request carries authentication info,
258 the username to match in the users list is taken from the Digest header
259 rather than from the From: field. This feature is considered experimental.
260 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
261 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
262 * The "localmask" setting was removed in version 1.2 and the reminder about it
263 being removed is now also removed.
264 * A new option "busylevel" for setting a level of calls where asterisk reports
265 a device as busy, to separate it from call-limit. This value is also added
266 to the SIP_PEER dialplan function.
267 * A new realtime family called "sipregs" is now supported to store SIP registration
268 data. If this family is defined, "sippeers" will be used for configuration and
269 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
270 registration data, as before.
271 * The SIPPEER function have new options for port address, call and pickup groups
272 * Added support for T.140 realtime text in SIP/RTP
273 * The "checkmwi" option has been removed from sip.conf, as it is no longer
274 required due to the restructuring of how MWI is handled. See the descriptions
275 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
276 for more information.
277 * Added rtpdest option to CHANNEL() dialplan function.
278 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
279 * SIP now adds a header to the CANCEL if the call was answered by another phone
280 in the same dial command, or if the new c option in dial() is used.
281 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
282 states it is not needed. For phones, however, that do require it the "registertrying" option
283 has been added so it can be enabled.
284 * A new option called "callcounter" (global/peer/user level) enables call counters needed
285 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
286 used to enable this functionality).
287 * New settings for timer T1 and timer B on a global level or per device. This makes it
288 possible to force timeout faster on non-responsive SIP servers. These settings are
289 considered advanced, so don't use them unless you have a problem.
290 * Added a dial string option to be able to set the To: header in an INVITE to any
291 SIP uri.
292 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
293 the qualify frequency.
294 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
295 were not properly torn down due to network or endpoint failures during an established
296 SIP session.
297 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
298 configs/sip.conf.sample for more information on how it is used.
299 * Added a new configuration option "authfailureevents" that enables manager events when
300 a peer can't authenticate properly.
301 * Added DNS manager support to registrations for peers not referencing a peer entry.
303 IAX2 changes
304 ------------
305 * Added the trunkmaxsize configuration option to chan_iax2.
306 * Added the srvlookup option to iax.conf
307 * Added support for OSP. The token is set and retrieved through the CHANNEL()
308 dialplan function.
310 XMPP Google Talk/Jingle changes
311 -------------------------------
312 * Added the bindaddr option to gtalk.conf.
314 Skinny changes
315 -------------
316 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
317 * Proper codec support in chan_skinny.
318 * Added settings for IP and Ethernet QoS requests
320 MGCP changes
321 ------------
322 * Added separate settings for media QoS in mgcp.conf
324 Console Channel Driver changes
325 ------------------------------
326 * Added experimental support for video send & receive to chan_oss.
327 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
328 a video source.
330 Phone channel changes (chan_phone)
331 ----------------------------------
332 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
334 H.323 channel Changes
335 ---------------------
336 * H323 remote hold notification support added (by NOTIFY message
337 and/or H.450 supplementary service)
339 Local channel changes
340 ---------------------
341 * The device state functionality in the Local channel driver has been updated
342 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
343 to just UNKNOWN if the extension exists.
344 * Added jitterbuffer support for chan_local. This allows you to use the
345 generic jitterbuffer on incoming calls going to Asterisk applications.
346 For example, this would allow you to use a jitterbuffer for an incoming
347 SIP call to Voicemail by putting a Local channel in the middle. This
348 feature is enabled by using the 'j' option in the Dial string to the Local
349 channel in conjunction with the existing 'n' option for local channels.
350 * A 'b' option has been added which causes chan_local to return the actual channel
351 that is behind it when queried. This is useful for transfer scenarios as the
352 actual channel will be transferred, not the Local channel.
354 Zaptel channel driver (chan_zap) Changes
355 ----------------------------------------
356 * SS7 support in chan_zap (via libss7 library)
357 * In India, some carriers transmit CID via dtmf. Some code has been added
358 that will handle some situations. The cidstart=polarity_IN choice has been added for
359 those carriers that transmit CID via dtmf after a polarity change.
360 * CID matching information is now shown when doing 'dialplan show'.
361 * Added zap show version CLI command to chan_zap.
362 * Added setvar support to zapata.conf channel entries.
363 * Added two new options: mwimonitor and mwimonitornotify. These options allow
364 you to enable MWI monitoring on FXO lines. When the MWI state changes,
365 the script specified in the mwimonitornotify option is executed. An internal
366 event indicating the new state of the mailbox is also generated, so that
367 the normal MWI facilities in Asterisk work as usual.
368 * Added signalling type 'auto', which attempts to use the same signalling type
369 for a channel as configured in Zaptel. This is primarily designed for analog
370 ports, but will also work for digital ports that are configured for FXS or FXO
371 signalling types. This mode is also the default now, so if your zapata.conf
372 does not specify signalling for a channel (which is unlikely as the sample
373 configuration file has always recommended specifying it for every channel) then
374 the 'auto' mode will be used for that channel if possible.
375 * Added a 'zap set dnd' command to allow CLI control of the Do-Not-Disturb
376 state for a channel; also ensured that the DNDState Manager event is
377 emitted no matter how the DND state is set or cleared.
379 New Channel Drivers
380 -------------------
381 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
382 configs/unistim.conf.sample for details. This new channel driver allows
383 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
384 * Added a new channel driver, chan_console, which uses portaudio as a cross
385 platform audio interface. It was written as a channel driver that would
386 work with Mac CoreAudio, but portaudio supports a number of other audio
387 interfaces, as well. Note that this channel driver requires v19 or higher
388 of portaudio; older versions have a different API.
390 DUNDi changes
391 -------------
392 * Added the ability to specify arguments to the Dial application when using
393 the DUNDi switch in the dialplan.
394 * Added the ability to set weights for responses dynamically. This can be
395 done using a global variable or a dialplan function. Using the SHELL()
396 function would allow you to have an external script set the weight for
397 each response.
398 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
399 functions will allow you to initiate a DUNDi query from the dialplan,
400 find out how many results there are, and access each one.
402 ENUM changes
403 ------------
404 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
405 functions will allow you to initiate an ENUM lookup from the dialplan,
406 and Asterisk will cache the results. ENUMRESULT can be used to access
407 the results without doing multiple DNS queries.
409 Voicemail Changes
410 -----------------
411 * Added the ability to customize which sound files are used for some of the
412 prompts within the Voicemail application by changing them in voicemail.conf
413 * Added the ability for the "voicemail show users" CLI command to show users
414 configured by the dynamic realtime configuration method.
415 * MWI (Message Waiting Indication) handling has been significantly
416 restructured internally to Asterisk. It is now totally event based
417 instead of polling based. The voicemail application will notify other
418 modules that have subscribed to MWI events when something in the mailbox
419 changes.
420 This also means that if any other entity outside of Asterisk is changing
421 the contents of mailboxes, then the voicemail application still needs to
422 poll for changes. Examples of situations that would require this option
423 are web interfaces to voicemail or an email client in the case of using
424 IMAP storage. So, two new options have been added to voicemail.conf
425 to account for this: "pollmailboxes" and "pollfreq". See the sample
426 configuration file for details.
427 * Added "tw" language support
428 * Added support for storage of greetings using an IMAP server
429 * Added ability to customize forward, reverse, stop, and pause keys for message playback
430 * SMDI is now enabled in voicemail using the smdienable option.
431 * A "lockmode" option has been added to asterisk.conf to configure the file
432 locking method used for voicemail, and potentially other things in the
433 future. The default is the old behavior, lockfile. However, there is a
434 new method, "flock", that uses a different method for situations where the
435 lockfile will not work, such as on SMB/CIFS mounts.
436 * Added the ability to backup deleted messages, to ease recovery in the case
437 that a user accidentally deletes a message, and discovers that they need it.
438 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
439 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
440 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
441 voicemail boxes. The SMDI interface can also poll for MWI changes when some
442 outside entity is modifying the state of the mailbox (such as IMAP storage or
443 a web interface of some kind).
444 * Added the support for marking messages as "urgent." There are two methods to accomplish
445 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
446 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
447 the message as urgent after he has recorded a voicemail by following the voice instructions.
448 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
449 messages
451 Queue changes
452 -------------
453 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
454 used across multiple queues.
455 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
456 setqueueentryvar options for each queue, see queues.conf.sample for details.
457 * Added keepstats option to queues.conf which will keep queue
458 statistics during a reload.
459 * setinterfacevar option in queues.conf also now sets a variable
460 called MEMBERNAME which contains the member's name.
461 * Added 'Strategy' field to manager event QueueParams which represents
462 the queue strategy in use.
463 * Added option to run macro when a queue member is connected to a caller,
464 see queues.conf.sample for details.
465 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
466 does not count paused queue members as unavailable.
467 * Added min-announce-frequency option to queues.conf which allows you to control the
468 minimum amount of time between queue announcements for use when the caller's queue
469 position changes frequently.
470 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
471 queue log.
472 * Added ability for non-realtime queues to have realtime members
473 * Added the "linear" strategy to queues.
474 * Added the "wrandom" strategy to queues.
475 * Added new channel variable QUEUE_MIN_PENALTY
476 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
477 rules in queuerules.conf. See configs/queuerules.conf.sample for details
478 * Added a new parameter for member definition, called state_interface. This may be
479 used so that a member may be called via one interface but have a different interface's
480 device state reported.
481 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
482 specified by the periodic-announce option, then one will be chosen randomly when it is time
483 to play a periodic announcment
484 * New configuration options: announce-position now takes two more values in addition to "yes" and
485 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
486 announce-position-limit. By setting announce-position to "limit" callers will only have their
487 position announced if their position is less than what is specified by announce-position-limit.
488 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
489 will be told that their are more than announce-position-limit callers waiting.
490 * Two new queue log events have been added. An ADDMEMBER event will be logged
491 when a realtime queue member is added and a REMOVEMEMBER event will be logged
492 when a realtime queue member is removed. Since there is no calling channel associated
493 with these events, the string "REALTIME" is placed where the channel's unique id
494 is typically placed.
496 MeetMe Changes
497 --------------
498 * The 'o' option to provide an optimization has been removed and its functionality
499 has been enabled by default.
500 * When a conference is created, the UNIQUEID of the channel that caused it to be
501 created is stored. Then, every channel that joins the conference will have the
502 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
503 callers that come and go from long standing conferences.
504 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
505 except it does operations on a channel by name, instead of number in a conference.
506 This is a very useful feature in combination with the 'X' option to ChanSpy.
507 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
508 when kicked out.
509 * Added new RealTime functionality to provide support for scheduled conferencing.
510 This includes optional messages to the caller if they attempt to join before
511 the schedule start time, or to allow the caller to join the conference early.
512 Also included is optional support for limiting the number of callers per
513 RealTime conference.
514 * Added the S() and L() options to the MeetMe application. These are pretty
515 much identical to the S() and L() options to Dial(). They let you set
516 timeouts for the conference, as well as have warning sounds played to
517 let the caller know how much time is left, and when it is running out.
518 * Added the ability to do "meetme concise" with the "meetme" CLI command.
519 This extends the concise capabilities of this CLI command to include
520 listing all conferences, instead of an addition to the other sub commands
521 for the "meetme" command.
522 * Added the ability to specify the music on hold class used to play into the
523 conference when there is only one member and the M option is used.
524 * Added MEETME_INFO dialplan function which provides a way to query
525 various properties of a Meetme conference.
527 Other Dialplan Application Changes
528 ----------------------------------
529 * Argument support for Gosub application
530 * From the to-do lists: straighten out the app timeout args:
531 Wait() app now really does 0.3 seconds- was truncating arg to an int.
532 WaitExten() same as Wait().
533 Congestion() - Now takes floating pt. argument.
534 Busy() - now takes floating pt. argument.
535 Read() - timeout now can be floating pt.
536 WaitForRing() now takes floating pt timeout arg.
537 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
538 * Added 's' option to Page application.
539 * Added 'E' and 'V' commands to ExternalIVR.
540 * Added 'o' and 'X' options to Chanspy.
541 * Added a new dialplan application, Bridge, which allows you to bridge the
542 calling channel to any other active channel on the system.
543 * Added the ability to specify a music on hold class to play instead of ringing
544 for the SLATrunk application.
545 * The Read application no longer exits the dialplan on error. Instead, it sets
546 READSTATUS to ERROR, which you can catch and handle separately.
547 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
548 of asking for verification of each name, one at a time.
549 * Privacy() no longer uses privacy.conf, as all options are specifyable as
550 direct options to the app.
551 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
552 for more details
553 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
554 * The ChannelRedirect application no longer exits the dialplan if the given channel
555 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
556 or NOCHANNEL if the given channel was not found.
557 * The silencethreshold setting that was previously configurable in multiple
558 applications is now settable globally via dsp.conf.
559 * Added ability to communicate over a TCP socket instead of forking a child process for the
560 ExternalIVR application.
562 Music On Hold Changes
563 ---------------------
564 * A new option, "digit", has been added for music on hold classes in
565 musiconhold.conf. If this is set for a music on hold class, a caller
566 listening to music on hold can press this digit to switch to listening
567 to this music on hold class.
568 * Support for realtime music on hold has been added.
569 * In conjunction with the realtime music on hold, a general section has
570 been added to musiconhold.conf, its sole variable is cachertclasses. If this
571 is set, then music on hold classes found in realtime will be cached in memory.
573 AEL Changes
574 -----------
575 * AEL upgraded to use the Gosub with Arguments instead
576 of Macro application, to hopefully reduce the problems
577 seen with the artificially low stack ceiling that
578 Macro bumps into. Macros can only call other Macros
579 to a depth of 7. Tests run using gosub, show depths
580 limited only by virtual memory. A small test demonstrated
581 recursive call depths of 100,000 without problems.
582 -- in addition to this, all apps that allowed a macro
583 to be called, as in Dial, queues, etc, are now allowing
584 a gosub call in similar fashion.
585 * AEL now generates LOCAL(argname) declarations when it
586 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
587 etc. That makes the arguments local in scope. The user
588 can define their own local variables in macros, now,
589 by saying "local myvar=someval;" or using Set() in this
590 fashion: Set(LOCAL(myvar)=someval); ("local" is now
591 an AEL keyword).
592 * utils/conf2ael introduced. Will convert an extensions.conf
593 file into extensions.ael. Very crude and unfinished, but
594 will be improved as time goes by. Should be useful for a
595 first pass at conversion.
596 * aelparse will now read extensions.conf to see if a referenced
597 macro or context is there before issueing a warning.
598 * AEL parser sets a local channel variable ~~EXTEN~~, to
599 preserve the value of ${EXTEN} thru switch statements.
600 * New operator in $[...] expressions: the ~~ operator serves
601 as a concatenation operator. AT THE MOMENT, it is really only
602 necessary and useful in AEL, especially in if() expressions.
603 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
604 any enclosing double-quotes, and evaluate to the value of a
605 concatenated with the value of b. For example if a is set to
606 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
607 evaluate to xyzabc .
610 Call Features (res_features) Changes
611 ------------------------------------
612 * Added the parkedcalltransfers option to features.conf
613 * The built-in method for doing attended transfers has been updated to
614 include some new options that allow you to have the transferee sent
615 back to the person that did the transfer if the transfer is not successful.
616 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
617 in features.conf.sample.
618 * Added support for configuring named groups of custom call features in
619 features.conf. This means that features can be written a single time, and
620 then mapped into groups of features for different key mappings or easier
621 access control.
622 * Updated the ParkedCall application to allow you to not specify a parking
623 extension. If you don't specify a parking space to pick up, it will grab
624 the first one available.
625 * Added cli command 'features reload' to reload call features from features.conf
626 * Moved into core asterisk binary.
628 Language Support Changes
629 ------------------------
630 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
631 * Added support for the Hungarian language for saying numbers, dates, and times.
633 AGI Changes
634 -----------
635 * Added SPEECH commands for speech recognition. A complete listing can be found
636 using agi show.
637 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
638 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
639 does not behave as expected; the native command needs to be used, instead.
641 Logger changes
642 --------------
643 * Added rotatestrategy option to logger.conf, along with two new options:
644 "timestamp" which will use the time to name the logger files instead of
645 sequence number; and "rotate", which rotates the names of the logfiles,
646 similar to the way syslog rotates files.
647 * Added exec_after_rotate option to logger.conf, which allows a system
648 command to be run after rotation. This is primarily useful with
649 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
650 and to ensure that the oldest log file gets deleted.
651 * Added realtime support for the queue log
653 Call Detail Records
654 -------------------
655 * The cdr_manager module has a [mappings] feature, like cdr_custom,
656 to add fields to the manager event from the CDR variables.
657 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
658 backend database CDR table. Specifically, additional, non-standard
659 columns are supported, merely by setting the corresponding CDR variable in
660 your dialplan. In addition, you may alias any column to another name (for
661 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
662 simply "alias src => ANI" in the configuration file). Records may be
663 posted to more than one backend, simply by specifying multiple categories
664 in the configuration file. And finally, you may filter which CDRs get
665 posted to each backend, by specifying a filter (which the record must
666 match) for the particular category. Filters are additive (meaning all
667 rules must match to post that CDR).
668 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
669 module. Specifically, you may add additional columns into the table and
670 they will be set, if you set the corresponding CDR variable name. Also,
671 if you omit columns in your database table, they will be silently skipped
672 (but a record will still be inserted, based on what columns remain). Note
673 that the other two features from cdr_adaptive_odbc (alias and filter) are
674 not currently supported.
675 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
676 has been disabled using the NoCDR application.
678 Miscellaneous New Modules
679 -------------------------
680 * Added a new CDR module, cdr_sqlite3_custom.
681 * Added a new realtime configuration module, res_config_sqlite
682 * Added a new codec translation module, codec_resample, which re-samples
683 signed linear audio between 8 kHz and 16 kHz to help support wideband
684 codecs.
685 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
686 based on configuration templates that use Asterisk dialplan function and
687 variable substitution. It should be possible to create phone profiles and
688 templates that work for the majority of phones provisioned over http. It
689 is currently only intended to provision a single user account per phone.
690 An example profile and set of templates for Polycom phones is provided.
691 NOTE: Polycom firmware is not included, but should be placed in
692 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
693 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
694 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
695 provided; there is a JACK() application, and a JACK_HOOK() function. Both
696 interfaces create an input and output JACK port. The application makes
697 these ports the endpoint of the call. The audio coming from the channel
698 goes out the output port and whatever comes back in on the input port is
699 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
700 audiohook on the channel. This lets you run the audio coming from a
701 channel through JACK, and whatever comes back in is what gets forwarded
702 on as the channel's audio. This is very useful for building custom
703 vocoders or doing recording or analysis of the channel's audio in another
704 application.
705 * Added a new module, res_config_curl, which permits using a HTTP POST url
706 to retrieve, create, update, and delete realtime information from a remote
707 web server. Note that this module requires func_curl.so to be loaded for
708 backend functionality.
709 * Added a new module, res_config_ldap, which permits the use of an LDAP
710 server for realtime data access.
711 * Added support for writing and running your dialplan in lua using the pbx_lua
712 module. See configs/extensions.lua.sample for examples of how to do this.
714 Miscellaneous
715 -------------
716 * Ability to use libcap to set high ToS bits when non-root
717 on Linux. If configure is unable to find libcap then you
718 can use --with-cap to specify the path.
719 * Added maxfiles option to options section of asterisk.conf which allows you to specify
720 what Asterisk should set as the maximum number of open files when it loads.
721 * Added the jittertargetextra configuration option.
722 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
723 configuration files for the IP channel drivers. The new option is "cos".
724 This information is also documented in doc/qos.tex, or the IP Quality of Service
725 section of asterisk.pdf.
726 * When originating a call using AMI or pbx_spool that fails the reason for failure
727 will now be available in the failed extension using the REASON dialplan variable.
728 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
729 It allows you to configure a prefix for auto-monitor recordings.
730 * A new extension pattern matching algorithm, based on a trie, is introduced
731 here, that could noticeably speed up mid-sized to large dialplans.
732 It is NOT used by default, as duplicating the behaviour of the old pattern
733 matcher is still under development. A config file option, in extensions.conf,
734 in the [general] section, called "extenpatternmatchingnew", is by default
735 set to false; setting that to true will force the use of the new algorithm.
736 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
737 be used to switch the algorithms at run time.
738 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
739 specifying which socket to use to connect to the running Asterisk daemon
740 (-s)
741 * Performance enhancements to the sched facility, which is used in
742 the channel drivers, etc. Added hashtabs and doubly-linked lists
743 to speed up deletion; start at the beginning or end of list to
744 speed up insertion.
745 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
746 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
747 Added regression tests to the tests/ dir, also.
748 * Added a refcount trace feature to astobj2 for those trying to balance
749 object creation, deletion; work, play; space and time. See the
750 notes in astobj2.h. Also, see utils/refcounter as well, as a
751 quick way to find unbalanced refcounts in what could be a sea
752 of objects that were balanced.
753 * Added logging to 'make update' command. See update.log
754 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
755 do not come from the remote party.
756 * Added the 'n' option to the SpeechBackground application to tell it to not
757 answer the channel if it has not already been answered.
758 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
759 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
760 dialplan debugging.
761 * iLBC source code no longer included (see UPGRADE.txt for details)
762 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
763 deadlock is detected, a backtrace of the stack which led to the lock calls
764 will be output to the CLI.
765 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
766 the "core show locks" CLI command will give lock information output as well
767 as a backtrace of the stack which led to the lock calls.