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1 ------------------------------------------------------------------------------
2 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
3 ------------------------------------------------------------------------------
5 Device State Handling
6 ---------------------
7 * The event infrastructure in Asterisk got another big update to help support
8 distributed events. It currently supports distributed device state and
9 distributed Voicemail MWI (Message Waiting Indication). A new module has
10 been merged, res_ais, which facilitates communicating events between servers.
11 It uses the SAForum AIS (Service Availability Forum Application Interface
12 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
13 a cluster of Asterisk servers, and to share events between them. For more
14 information on setting this up, see doc/distributed_devstate.txt.
16 Dialplan Functions
17 ------------------
18 * Added a new dialplan function, AST_CONFIG(), which allows you to access
19 variables from an Asterisk configuration file.
20 * The JACK_HOOK function now has a c() option to supply a custom client name.
21 * Added two new dialplan functions from libspeex for audio gain control and
22 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
23 rx directions of a channel from the dialplan.
24 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
25 based on other parameters. The default is still to search based on the
26 forwarding station ID. However, there are new options that allow you to search
27 based on the message desk terminal ID, or the message desk number.
28 * TIMEOUT() has been modified to be accurate down to the millisecond.
29 * ENUM*() functions now include the following new options:
30 - 'u' returns the full URI and does not strip off the URI-scheme.
31 - 's' triggers ISN specific rewriting
32 - 'i' looks for branches into an Infrastructure ENUM tree
33 - 'd' for a direct DNS lookup without any flipping of digits.
34 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
35 * CHANNEL() now has options for the maximum, minimum, and standard or normal
36 deviation of jitter, rtt, and loss for a call using chan_sip.
38 DAHDI channel driver (chan_dahdi) Changes
39 ----------------------------------------
40 * Channels can now be configured using named sections in chan_dahdi.conf, just
41 like other channel drivers, including the use of templates.
42 * The default for pridialplan has changed from 'national' to 'unknown'.
44 PBX Changes
45 -----------
46 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
47 to something that matches the pattern a hint will be created using the contents
48 and variables evaluated.
49 * Dialplan matching has been extended to allow an extension to return to the
50 PBX core to wait for more digits. This is done by using the new dialplan
51 application called "Incomplete". This will permit a whole new level of
52 extension control, by giving the administrator more control over early
53 matches employing one of the short-circuit pattern match operators. Note
54 that custom applications can trigger this same behavior by returning the
55 special value AST_PBX_INCOMPLETE.
57 Application Changes
58 -------------------
59 * Directory now permits both first and last names to be matched at the same
60 time. In addition, the number of digits to enter of the name can be set in
61 the arguments to Directory; previously, you could enter only 3, regardless
62 of how many names are in your company. For large companies, this should be
63 quite helpful.
64 * Voicemail now permits a mailbox setting to wrap around from first to last
65 messages, if the "messagewrap" option is set to a true value.
66 * Voicemail now permits an external script to be run, for password validation.
67 The script should output "VALID" or "INVALID" on stdout, depending upon the
68 wish to validate or invalidate the password given. Arguments are:
69 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
70 more details
71 * Dial has a new option: F(context^extension^pri), which permits a callee to
72 continue in the dialplan, at the specified label, if the caller hangs up.
73 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
74 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
75 * The Jack application now has a c() option to supply a custom client name.
76 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
77 like the pre-existing whisper mode, except that the spy can also talk to the
78 participant on the bridged channel as well.
79 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
80 to be spoken instead of the channel name or number. For more information on the
81 use of this option, issue the command "core show application ChanSpy" from the
82 Asterisk CLI.
83 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
84 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
85 words, if using the 'd' option, it is not possible to enter a number to append to
86 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
87 change to whisper mode, and pressing 6 will change to barge mode.
88 * ExternalIVR now takes several options that affect the way it performs, as
89 well as having several new commands. Please see doc/externalivr.txt for the
90 complete documentation.
91 * ChanIsAvail has a new option, 'a', which will return all available channels instead
92 of just the first one if you give the function more then one channel to check.
93 * PrivacyManager now takes an option where you can specify a context where the
94 given number will be matched. This way you have more control over who is allowed
95 and it stops the people who blindly enter 10 digits.
96 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
97 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
98 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
99 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
100 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
101 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
102 * The Dial() application no longer copies the language used by the caller to the callee's
103 channel. If you desire for the caller's channel's language to be used for file playback
104 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
106 SIP Changes
107 -----------
108 * The ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using setvar to cause a given
109 audio file to be played upon completion of an attended transfer.
110 * Added DNS manager support to registrations for peers referencing peer entries.
111 DNS manager runs in the background which allows DNS lookups to be run asynchronously
112 as well as periodically updating the IP address. These properties allow for
113 better performance as well as recovery in the event of an IP change.
114 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
115 load/reload of large numbers of peers/users by ~40x (for large lists of peers.
116 Initially, we saw 4x improvement in call setup/destruction, but at the time
117 of merging, this gain has disappeared; further research will be done to try
118 and restore this performance improvement. Astobj2 refcounting is now used
119 for users, peers, and dialogs. Users are encouraged to assist in regression
120 testing and problem reporting!
121 * Added ability to specify registration expiry time on a per registration basis in
122 the register line.
123 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
124 lost packets.
125 * Added t38pt_usertpsource option. See sip.conf.sample for details.
126 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
127 * 'sip show peers' and 'sip show users' display their entries sorted in
128 alphabetical order, as opposed to the order they were in, in the config
129 file or database.
130 * Videosupport now supports an additional option, "always", which always sets
131 up video RTP ports, even on clients that don't support it. This helps with
132 callfiles and certain transfers to ensure that if two video phones are
133 connected, they will always share video feeds.
135 IAX Changes
136 -----------
137 * Existing DNS manager lookups extended to check for SRV records.
139 CLI Changes
140 -----------
141 * New CLI command, "config reload <file.conf>" which reloads any module that
142 references that particular configuration file. Also added "config list"
143 which shows which configuration files are in use.
144 * New CLI commands, "pri show version" and "ss7 show version" that will
145 display which version of libpri and libss7 are being used, respectively.
146 A new API call was added so trunk will now have to be compiled against
147 a versions of libpri and libss7 that have them or it will not know that
148 these libraries exist.
150 DNS manager changes
151 -------------------
152 * Addresses managed by DNS manager now can check to see if there is a DNS
153 SRV record for a given domain and will use that hostname/port if present.
155 AMI - The manager (TCP/TLS/HTTP)
156 --------------------------------
157 * The Status command now takes an optional list of variables to display
158 along with channel status.
160 ODBC Changes
161 ------------
162 * res_odbc no longer has a limit of 1023 total possible unshared connections,
163 as some people were running into this limit. This limit has been increased
164 to 4.2 billion.
166 Queue changes
167 -------------
168 * The TRANSFER queue log entry now includes the the caller's original
169 position in the transferred-from queue.
170 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
171 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
172 as well as an explanation about timeout options in general
174 ------------------------------------------------------------------------------
175 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
176 ------------------------------------------------------------------------------
178 AMI - The manager (TCP/TLS/HTTP)
179 --------------------------------
180 * Manager has undergone a lot of changes, all of them documented
181 in doc/manager_1_1.txt
182 * Manager version has changed to 1.1
183 * Added a new action 'CoreShowChannels' to list currently defined channels
184 and some information about them.
185 * Added a new action 'SIPshowregistry' to list SIP registrations.
186 * Added TLS support for the manager interface and HTTP server
187 * Added the URI redirect option for the built-in HTTP server
188 * The output of CallerID in Manager events is now more consistent.
189 CallerIDNum is used for number and CallerIDName for name.
190 * Enable https support for builtin web server.
191 See configs/http.conf.sample for details.
192 * Added a new action, GetConfigJSON, which can return the contents of an
193 Asterisk configuration file in JSON format. This is intended to help
194 improve the performance of AJAX applications using the manager interface
195 over HTTP.
196 * SIP and IAX manager events now use "ChannelType" in all cases where we
197 indicate channel driver. Previously, we used a mixture of "Channel"
198 and "ChannelDriver" headers.
199 * Added a "Bridge" action which allows you to bridge any two channels that
200 are currently active on the system.
201 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
202 the voicemail users setup.
203 * Added 'DBDel' and 'DBDelTree' manager commands.
204 * cdr_manager now reports events via the "cdr" level, separating it from
205 the very verbose "call" level.
206 * Manager users are now stored in memory. If you change the manager account
207 list (delete or add accounts) you need to reload manager.
208 * Added Masquerade manager event for when a masquerade happens between
209 two channels.
210 * Added "manager reload" command for the CLI
211 * Lots of commands that only provided information are now allowed under the
212 Reporting privilege, instead of only under Call or System.
213 * The IAX* commands now require either System or Reporting privilege, to
214 mirror the privileges of the SIP* commands.
215 * Added ability to retrieve list of categories in a config file.
216 * Added ability to retrieve the content of a particular category.
217 * Added ability to empty a context.
218 * Created new action to create a new file.
219 * Updated delete action to allow deletion by line number with respect to category.
220 * Added new action insert to add new variable to category at specified line.
221 * Updated action newcat to allow new category to be inserted in file above another
222 existing category.
223 * Added new event "JitterBufStats" in the IAX2 channel
224 * Originate now requires the Originate privilege and, if you want to call out
225 to a subshell, it requires the System privilege, as well. This was done to
226 enhance manager security.
227 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
228 * New command: Atxfer. See doc/manager_1_1.txt for more details or
229 manager show command Atxfer from the CLI
231 Dialplan functions
232 ------------------
233 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
234 state in the dialplan, as well as creating custom device states that are
235 controllable from the dialplan.
236 * Extend CALLERID() function with "pres" and "ton" parameters to
237 fetch string representation of calling number presentation indicator
238 and numeric representation of type of calling number value.
239 * MailboxExists converted to dialplan function
240 * A new option to Dial() for telling IP phones not to count the call
241 as "missed" when dial times out and cancels.
242 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
243 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
244 held for any given channel. Also, locks are automatically freed when a
245 channel is hung up.
246 * Added HINT() dialplan function that allows retrieving hint information.
247 Hints are mappings between extensions and devices for the sake of
248 determining the state of an extension. This function can retrieve the list
249 of devices or the name associated with a hint.
250 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
251 of any extension.
252 * Added SYSINFO() dialplan function which allows retrieval of system information
253 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
254 the existence of a dialplan target.
255 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
256 upper and lower case, respectively.
257 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
258 ID for the call (not the Asterisk call ID or unique ID), provided that the
259 channel driver supports this. For SIP, you get the SIP call-ID for the
260 bridged channel which you can store in the CDR with a custom field.
262 CLI Changes
263 -----------
264 * New CLI command "core show hint" (usage: core show hint <exten>)
265 * New CLI command "core show settings"
266 * Added 'core show channels count' CLI command.
267 * Added the ability to set the core debug and verbose values on a per-file basis.
268 * Added 'queue pause member' and 'queue unpause member' CLI commands
269 * Ability to set process limits ("ulimit") without restarting Asterisk
270 * Enhanced "agi debug" to print the channel name as a prefix to the debug
271 output to make debugging on busy systems much easier.
272 * New CLI commands "dialplan set extenpatternmatching true/false"
273 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
274 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
275 listed in the startup_commands section of cli.conf will get executed.
276 * Added a CLI command, "devstate change", which allows you to set custom device
277 states from the func_devstate module that provides the DEVICE_STATE() function
278 and handling of the "Custom:" devices.
279 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
280 sorted into the different possible callbacks, with the number of entries
281 currently scheduled for each. Gives you a feel for how busy the sip channel
282 driver is.
284 SIP changes
285 -----------
286 * Improved NAT and STUN support.
287 chan_sip now can use port numbers in bindaddr, externip and externhost
288 options, as well as contact a STUN server to detect its external address
289 for the SIP socket. See sip.conf.sample, 'NAT' section.
290 * The default SIP useragent= identifier now includes the Asterisk version
291 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
292 If set, and the incoming request carries authentication info,
293 the username to match in the users list is taken from the Digest header
294 rather than from the From: field. This feature is considered experimental.
295 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
296 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
297 * The "localmask" setting was removed in version 1.2 and the reminder about it
298 being removed is now also removed.
299 * A new option "busylevel" for setting a level of calls where asterisk reports
300 a device as busy, to separate it from call-limit. This value is also added
301 to the SIP_PEER dialplan function.
302 * A new realtime family called "sipregs" is now supported to store SIP registration
303 data. If this family is defined, "sippeers" will be used for configuration and
304 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
305 registration data, as before.
306 * The SIPPEER function have new options for port address, call and pickup groups
307 * Added support for T.140 realtime text in SIP/RTP
308 * The "checkmwi" option has been removed from sip.conf, as it is no longer
309 required due to the restructuring of how MWI is handled. See the descriptions
310 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
311 for more information.
312 * Added rtpdest option to CHANNEL() dialplan function.
313 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
314 * SIP now adds a header to the CANCEL if the call was answered by another phone
315 in the same dial command, or if the new c option in dial() is used.
316 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
317 states it is not needed. For phones, however, that do require it the "registertrying" option
318 has been added so it can be enabled.
319 * A new option called "callcounter" (global/peer/user level) enables call counters needed
320 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
321 used to enable this functionality).
322 * New settings for timer T1 and timer B on a global level or per device. This makes it
323 possible to force timeout faster on non-responsive SIP servers. These settings are
324 considered advanced, so don't use them unless you have a problem.
325 * Added a dial string option to be able to set the To: header in an INVITE to any
326 SIP uri.
327 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
328 the qualify frequency.
329 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
330 were not properly torn down due to network or endpoint failures during an established
331 SIP session.
332 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
333 configs/sip.conf.sample for more information on how it is used.
334 * Added a new configuration option "authfailureevents" that enables manager events when
335 a peer can't authenticate properly.
336 * Added DNS manager support to registrations for peers not referencing a peer entry.
338 IAX2 changes
339 ------------
340 * Added the trunkmaxsize configuration option to chan_iax2.
341 * Added the srvlookup option to iax.conf
342 * Added support for OSP. The token is set and retrieved through the CHANNEL()
343 dialplan function.
345 XMPP Google Talk/Jingle changes
346 -------------------------------
347 * Added the bindaddr option to gtalk.conf.
349 Skinny changes
350 -------------
351 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
352 * Proper codec support in chan_skinny.
353 * Added settings for IP and Ethernet QoS requests
355 MGCP changes
356 ------------
357 * Added separate settings for media QoS in mgcp.conf
359 Console Channel Driver changes
360 ------------------------------
361 * Added experimental support for video send & receive to chan_oss.
362 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
363 a video source.
365 Phone channel changes (chan_phone)
366 ----------------------------------
367 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
369 H.323 channel Changes
370 ---------------------
371 * H323 remote hold notification support added (by NOTIFY message
372 and/or H.450 supplementary service)
374 Local channel changes
375 ---------------------
376 * The device state functionality in the Local channel driver has been updated
377 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
378 to just UNKNOWN if the extension exists.
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 * A 'b' option has been added which causes chan_local to return the actual channel
386 that is behind it when queried. This is useful for transfer scenarios as the
387 actual channel will be transferred, not the Local channel.
389 Agent channel changes
390 ----------------------
391 * The ackcall and endcall options are now supplemented with options acceptdtmf
392 and enddtmf. These allow for the DTMF keypress to be configurable. The options
393 default to their old hard-coded values ('#' and '*' respectively) so this should
394 not break any existing agent installations.
396 DAHDI channel driver (chan_dahdi) Changes
397 ----------------------------------------
398 * SS7 support (via libss7 library)
399 * In India, some carriers transmit CID via dtmf. Some code has been added
400 that will handle some situations. The cidstart=polarity_IN choice has been added for
401 those carriers that transmit CID via dtmf after a polarity change.
402 * CID matching information is now shown when doing 'dialplan show'.
403 * Added dahdi show version CLI command.
404 * Added setvar support to chan_dahdi.conf channel entries.
405 * Added two new options: mwimonitor and mwimonitornotify. These options allow
406 you to enable MWI monitoring on FXO lines. When the MWI state changes,
407 the script specified in the mwimonitornotify option is executed. An internal
408 event indicating the new state of the mailbox is also generated, so that
409 the normal MWI facilities in Asterisk work as usual.
410 * Added signalling type 'auto', which attempts to use the same signalling type
411 for a channel as configured in DAHDI. This is primarily designed for analog
412 ports, but will also work for digital ports that are configured for FXS or FXO
413 signalling types. This mode is also the default now, so if your chan_dahdi.conf
414 does not specify signalling for a channel (which is unlikely as the sample
415 configuration file has always recommended specifying it for every channel) then
416 the 'auto' mode will be used for that channel if possible.
417 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
418 state for a channel; also ensured that the DNDState Manager event is
419 emitted no matter how the DND state is set or cleared.
421 New Channel Drivers
422 -------------------
423 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
424 configs/unistim.conf.sample for details. This new channel driver allows
425 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
426 * Added a new channel driver, chan_console, which uses portaudio as a cross
427 platform audio interface. It was written as a channel driver that would
428 work with Mac CoreAudio, but portaudio supports a number of other audio
429 interfaces, as well. Note that this channel driver requires v19 or higher
430 of portaudio; older versions have a different API.
432 DUNDi changes
433 -------------
434 * Added the ability to specify arguments to the Dial application when using
435 the DUNDi switch in the dialplan.
436 * Added the ability to set weights for responses dynamically. This can be
437 done using a global variable or a dialplan function. Using the SHELL()
438 function would allow you to have an external script set the weight for
439 each response.
440 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
441 functions will allow you to initiate a DUNDi query from the dialplan,
442 find out how many results there are, and access each one.
444 ENUM changes
445 ------------
446 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
447 functions will allow you to initiate an ENUM lookup from the dialplan,
448 and Asterisk will cache the results. ENUMRESULT can be used to access
449 the results without doing multiple DNS queries.
451 Voicemail Changes
452 -----------------
453 * Added the ability to customize which sound files are used for some of the
454 prompts within the Voicemail application by changing them in voicemail.conf
455 * Added the ability for the "voicemail show users" CLI command to show users
456 configured by the dynamic realtime configuration method.
457 * MWI (Message Waiting Indication) handling has been significantly
458 restructured internally to Asterisk. It is now totally event based
459 instead of polling based. The voicemail application will notify other
460 modules that have subscribed to MWI events when something in the mailbox
461 changes.
462 This also means that if any other entity outside of Asterisk is changing
463 the contents of mailboxes, then the voicemail application still needs to
464 poll for changes. Examples of situations that would require this option
465 are web interfaces to voicemail or an email client in the case of using
466 IMAP storage. So, two new options have been added to voicemail.conf
467 to account for this: "pollmailboxes" and "pollfreq". See the sample
468 configuration file for details.
469 * Added "tw" language support
470 * Added support for storage of greetings using an IMAP server
471 * Added ability to customize forward, reverse, stop, and pause keys for message playback
472 * SMDI is now enabled in voicemail using the smdienable option.
473 * A "lockmode" option has been added to asterisk.conf to configure the file
474 locking method used for voicemail, and potentially other things in the
475 future. The default is the old behavior, lockfile. However, there is a
476 new method, "flock", that uses a different method for situations where the
477 lockfile will not work, such as on SMB/CIFS mounts.
478 * Added the ability to backup deleted messages, to ease recovery in the case
479 that a user accidentally deletes a message, and discovers that they need it.
480 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
481 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
482 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
483 voicemail boxes. The SMDI interface can also poll for MWI changes when some
484 outside entity is modifying the state of the mailbox (such as IMAP storage or
485 a web interface of some kind).
486 * Added the support for marking messages as "urgent." There are two methods to accomplish
487 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
488 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
489 the message as urgent after he has recorded a voicemail by following the voice instructions.
490 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
491 messages
493 Queue changes
494 -------------
495 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
496 used across multiple queues.
497 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
498 setqueueentryvar options for each queue, see queues.conf.sample for details.
499 * Added keepstats option to queues.conf which will keep queue
500 statistics during a reload.
501 * setinterfacevar option in queues.conf also now sets a variable
502 called MEMBERNAME which contains the member's name.
503 * Added 'Strategy' field to manager event QueueParams which represents
504 the queue strategy in use.
505 * Added option to run macro when a queue member is connected to a caller,
506 see queues.conf.sample for details.
507 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
508 does not count paused queue members as unavailable.
509 * Added min-announce-frequency option to queues.conf which allows you to control the
510 minimum amount of time between queue announcements for use when the caller's queue
511 position changes frequently.
512 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
513 queue log.
514 * Added ability for non-realtime queues to have realtime members
515 * Added the "linear" strategy to queues.
516 * Added the "wrandom" strategy to queues.
517 * Added new channel variable QUEUE_MIN_PENALTY
518 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
519 rules in queuerules.conf. See configs/queuerules.conf.sample for details
520 * Added a new parameter for member definition, called state_interface. This may be
521 used so that a member may be called via one interface but have a different interface's
522 device state reported.
523 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
524 specified by the periodic-announce option, then one will be chosen randomly when it is time
525 to play a periodic announcment
526 * New configuration options: announce-position now takes two more values in addition to "yes" and
527 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
528 announce-position-limit. By setting announce-position to "limit" callers will only have their
529 position announced if their position is less than what is specified by announce-position-limit.
530 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
531 will be told that their are more than announce-position-limit callers waiting.
532 * Two new queue log events have been added. An ADDMEMBER event will be logged
533 when a realtime queue member is added and a REMOVEMEMBER event will be logged
534 when a realtime queue member is removed. Since there is no calling channel associated
535 with these events, the string "REALTIME" is placed where the channel's unique id
536 is typically placed.
538 MeetMe Changes
539 --------------
540 * The 'o' option to provide an optimization has been removed and its functionality
541 has been enabled by default.
542 * When a conference is created, the UNIQUEID of the channel that caused it to be
543 created is stored. Then, every channel that joins the conference will have the
544 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
545 callers that come and go from long standing conferences.
546 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
547 except it does operations on a channel by name, instead of number in a conference.
548 This is a very useful feature in combination with the 'X' option to ChanSpy.
549 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
550 when kicked out.
551 * Added new RealTime functionality to provide support for scheduled conferencing.
552 This includes optional messages to the caller if they attempt to join before
553 the schedule start time, or to allow the caller to join the conference early.
554 Also included is optional support for limiting the number of callers per
555 RealTime conference.
556 * Added the S() and L() options to the MeetMe application. These are pretty
557 much identical to the S() and L() options to Dial(). They let you set
558 timeouts for the conference, as well as have warning sounds played to
559 let the caller know how much time is left, and when it is running out.
560 * Added the ability to do "meetme concise" with the "meetme" CLI command.
561 This extends the concise capabilities of this CLI command to include
562 listing all conferences, instead of an addition to the other sub commands
563 for the "meetme" command.
564 * Added the ability to specify the music on hold class used to play into the
565 conference when there is only one member and the M option is used.
566 * Added MEETME_INFO dialplan function which provides a way to query
567 various properties of a Meetme conference.
569 Other Dialplan Application Changes
570 ----------------------------------
571 * Argument support for Gosub application
572 * From the to-do lists: straighten out the app timeout args:
573 Wait() app now really does 0.3 seconds- was truncating arg to an int.
574 WaitExten() same as Wait().
575 Congestion() - Now takes floating pt. argument.
576 Busy() - now takes floating pt. argument.
577 Read() - timeout now can be floating pt.
578 WaitForRing() now takes floating pt timeout arg.
579 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
580 * Added 's' option to Page application.
581 * Added 'E' and 'V' commands to ExternalIVR.
582 * Added 'o' and 'X' options to Chanspy.
583 * Added a new dialplan application, Bridge, which allows you to bridge the
584 calling channel to any other active channel on the system.
585 * Added the ability to specify a music on hold class to play instead of ringing
586 for the SLATrunk application.
587 * The Read application no longer exits the dialplan on error. Instead, it sets
588 READSTATUS to ERROR, which you can catch and handle separately.
589 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
590 of asking for verification of each name, one at a time.
591 * Privacy() no longer uses privacy.conf, as all options are specifyable as
592 direct options to the app.
593 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
594 for more details
595 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
596 * The ChannelRedirect application no longer exits the dialplan if the given channel
597 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
598 or NOCHANNEL if the given channel was not found.
599 * The silencethreshold setting that was previously configurable in multiple
600 applications is now settable globally via dsp.conf.
601 * Added ability to communicate over a TCP socket instead of forking a child process for the
602 ExternalIVR application.
604 Music On Hold Changes
605 ---------------------
606 * A new option, "digit", has been added for music on hold classes in
607 musiconhold.conf. If this is set for a music on hold class, a caller
608 listening to music on hold can press this digit to switch to listening
609 to this music on hold class.
610 * Support for realtime music on hold has been added.
611 * In conjunction with the realtime music on hold, a general section has
612 been added to musiconhold.conf, its sole variable is cachertclasses. If this
613 is set, then music on hold classes found in realtime will be cached in memory.
615 AEL Changes
616 -----------
617 * AEL upgraded to use the Gosub with Arguments instead
618 of Macro application, to hopefully reduce the problems
619 seen with the artificially low stack ceiling that
620 Macro bumps into. Macros can only call other Macros
621 to a depth of 7. Tests run using gosub, show depths
622 limited only by virtual memory. A small test demonstrated
623 recursive call depths of 100,000 without problems.
624 -- in addition to this, all apps that allowed a macro
625 to be called, as in Dial, queues, etc, are now allowing
626 a gosub call in similar fashion.
627 * AEL now generates LOCAL(argname) declarations when it
628 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
629 etc. That makes the arguments local in scope. The user
630 can define their own local variables in macros, now,
631 by saying "local myvar=someval;" or using Set() in this
632 fashion: Set(LOCAL(myvar)=someval); ("local" is now
633 an AEL keyword).
634 * utils/conf2ael introduced. Will convert an extensions.conf
635 file into extensions.ael. Very crude and unfinished, but
636 will be improved as time goes by. Should be useful for a
637 first pass at conversion.
638 * aelparse will now read extensions.conf to see if a referenced
639 macro or context is there before issueing a warning.
640 * AEL parser sets a local channel variable ~~EXTEN~~, to
641 preserve the value of ${EXTEN} thru switch statements.
642 * New operator in $[...] expressions: the ~~ operator serves
643 as a concatenation operator. AT THE MOMENT, it is really only
644 necessary and useful in AEL, especially in if() expressions.
645 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
646 any enclosing double-quotes, and evaluate to the value of a
647 concatenated with the value of b. For example if a is set to
648 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
649 evaluate to xyzabc .
652 Call Features (res_features) Changes
653 ------------------------------------
654 * Added the parkedcalltransfers option to features.conf
655 * The built-in method for doing attended transfers has been updated to
656 include some new options that allow you to have the transferee sent
657 back to the person that did the transfer if the transfer is not successful.
658 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
659 in features.conf.sample.
660 * Added support for configuring named groups of custom call features in
661 features.conf. This means that features can be written a single time, and
662 then mapped into groups of features for different key mappings or easier
663 access control.
664 * Updated the ParkedCall application to allow you to not specify a parking
665 extension. If you don't specify a parking space to pick up, it will grab
666 the first one available.
667 * Added cli command 'features reload' to reload call features from features.conf
668 * Moved into core asterisk binary.
670 Language Support Changes
671 ------------------------
672 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
673 * Added support for the Hungarian language for saying numbers, dates, and times.
675 AGI Changes
676 -----------
677 * Added SPEECH commands for speech recognition. A complete listing can be found
678 using agi show.
679 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
680 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
681 does not behave as expected; the native command needs to be used, instead.
683 Logger changes
684 --------------
685 * Added rotatestrategy option to logger.conf, along with two new options:
686 "timestamp" which will use the time to name the logger files instead of
687 sequence number; and "rotate", which rotates the names of the logfiles,
688 similar to the way syslog rotates files.
689 * Added exec_after_rotate option to logger.conf, which allows a system
690 command to be run after rotation. This is primarily useful with
691 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
692 and to ensure that the oldest log file gets deleted.
693 * Added realtime support for the queue log
695 Call Detail Records
696 -------------------
697 * The cdr_manager module has a [mappings] feature, like cdr_custom,
698 to add fields to the manager event from the CDR variables.
699 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
700 backend database CDR table. Specifically, additional, non-standard
701 columns are supported, merely by setting the corresponding CDR variable in
702 your dialplan. In addition, you may alias any column to another name (for
703 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
704 simply "alias src => ANI" in the configuration file). Records may be
705 posted to more than one backend, simply by specifying multiple categories
706 in the configuration file. And finally, you may filter which CDRs get
707 posted to each backend, by specifying a filter (which the record must
708 match) for the particular category. Filters are additive (meaning all
709 rules must match to post that CDR).
710 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
711 module. Specifically, you may add additional columns into the table and
712 they will be set, if you set the corresponding CDR variable name. Also,
713 if you omit columns in your database table, they will be silently skipped
714 (but a record will still be inserted, based on what columns remain). Note
715 that the other two features from cdr_adaptive_odbc (alias and filter) are
716 not currently supported.
717 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
718 has been disabled using the NoCDR application.
720 Miscellaneous New Modules
721 -------------------------
722 * Added a new CDR module, cdr_sqlite3_custom.
723 * Added a new realtime configuration module, res_config_sqlite
724 * Added a new codec translation module, codec_resample, which re-samples
725 signed linear audio between 8 kHz and 16 kHz to help support wideband
726 codecs.
727 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
728 based on configuration templates that use Asterisk dialplan function and
729 variable substitution. It should be possible to create phone profiles and
730 templates that work for the majority of phones provisioned over http. It
731 is currently only intended to provision a single user account per phone.
732 An example profile and set of templates for Polycom phones is provided.
733 NOTE: Polycom firmware is not included, but should be placed in
734 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
735 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
736 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
737 provided; there is a JACK() application, and a JACK_HOOK() function. Both
738 interfaces create an input and output JACK port. The application makes
739 these ports the endpoint of the call. The audio coming from the channel
740 goes out the output port and whatever comes back in on the input port is
741 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
742 audiohook on the channel. This lets you run the audio coming from a
743 channel through JACK, and whatever comes back in is what gets forwarded
744 on as the channel's audio. This is very useful for building custom
745 vocoders or doing recording or analysis of the channel's audio in another
746 application.
747 * Added a new module, res_config_curl, which permits using a HTTP POST url
748 to retrieve, create, update, and delete realtime information from a remote
749 web server. Note that this module requires func_curl.so to be loaded for
750 backend functionality.
751 * Added a new module, res_config_ldap, which permits the use of an LDAP
752 server for realtime data access.
753 * Added support for writing and running your dialplan in lua using the pbx_lua
754 module. See configs/extensions.lua.sample for examples of how to do this.
756 Miscellaneous
757 -------------
758 * Ability to use libcap to set high ToS bits when non-root
759 on Linux. If configure is unable to find libcap then you
760 can use --with-cap to specify the path.
761 * Added maxfiles option to options section of asterisk.conf which allows you to specify
762 what Asterisk should set as the maximum number of open files when it loads.
763 * Added the jittertargetextra configuration option.
764 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
765 configuration files for the IP channel drivers. The new option is "cos".
766 This information is also documented in doc/qos.tex, or the IP Quality of Service
767 section of asterisk.pdf.
768 * When originating a call using AMI or pbx_spool that fails the reason for failure
769 will now be available in the failed extension using the REASON dialplan variable.
770 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
771 It allows you to configure a prefix for auto-monitor recordings.
772 * A new extension pattern matching algorithm, based on a trie, is introduced
773 here, that could noticeably speed up mid-sized to large dialplans.
774 It is NOT used by default, as duplicating the behaviour of the old pattern
775 matcher is still under development. A config file option, in extensions.conf,
776 in the [general] section, called "extenpatternmatchingnew", is by default
777 set to false; setting that to true will force the use of the new algorithm.
778 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
779 be used to switch the algorithms at run time.
780 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
781 specifying which socket to use to connect to the running Asterisk daemon
782 (-s)
783 * Performance enhancements to the sched facility, which is used in
784 the channel drivers, etc. Added hashtabs and doubly-linked lists
785 to speed up deletion; start at the beginning or end of list to
786 speed up insertion.
787 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
788 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
789 Added regression tests to the tests/ dir, also.
790 * Added a refcount trace feature to astobj2 for those trying to balance
791 object creation, deletion; work, play; space and time. See the
792 notes in astobj2.h. Also, see utils/refcounter as well, as a
793 quick way to find unbalanced refcounts in what could be a sea
794 of objects that were balanced.
795 * Added logging to 'make update' command. See update.log
796 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
797 do not come from the remote party.
798 * Added the 'n' option to the SpeechBackground application to tell it to not
799 answer the channel if it has not already been answered.
800 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
801 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
802 dialplan debugging.
803 * iLBC source code no longer included (see UPGRADE.txt for details)
804 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
805 deadlock is detected, a backtrace of the stack which led to the lock calls
806 will be output to the CLI.
807 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
808 the "core show locks" CLI command will give lock information output as well
809 as a backtrace of the stack which led to the lock calls.
810 * users.conf now sports an optional alternateexts property, which permits
811 allocation of additional extensions which will reach the specified user.