1 ======================================================================
3 === This file documents the new and/or enhanced functionality added in
4 === the Asterisk versions listed below. This file does NOT include
5 === changes in behavior that would not be backwards compatible with
6 === previous versions; for that information see the UPGRADE.txt file
7 === and the other UPGRADE files for older releases.
9 ======================================================================
11 ------------------------------------------------------------------------------
12 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
13 ------------------------------------------------------------------------------
17 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
18 Snom phones use this for call pickup of extensions that the phone is
20 * Added support for subscribing to a voice mailbox on a remote server and
21 making the new/old message count available to local devices.
22 * Added support for setting the domain in the URI for caller of an
23 outbound call by using the SIPFROMDOMAIN channel variable.
24 * Added a new configuration option "remotesecret" for authentication to
25 remote services. For backwards compatibility, "secret" still has the
26 same function as before, but now you can configure both a remote secret and a
27 local secret for mutual authentication.
28 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
29 option is enabled, a SIP channel will go to the fax extension (if it exists)
30 after T38 is negotiated. This option is disabled by default.
34 * The configuration file now holds seperate sections for devices and lines.
35 Please have a look at configs/skinny.conf.sample and change your skinny.conf
40 * The UK option waitfordialtone has been added for use with BT analog
45 * Added a new dialplan function, CURLOPT, which permits setting various
46 options that may be useful with the CURL dialplan function, such as
47 cookies, proxies, connection timeouts, passwords, etc.
48 * Permit the syntax and synopsis fields of the corresponding dialplan
49 functions to be individually set from func_odbc.conf.
50 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
51 * func_odbc now may specify an insert query to execute, when the write query
52 affects 0 rows (usually indicating that no such row exists).
53 * Added a new dialplan function, LISTFILTER, which permits removing elements
54 from a set list, by name. Uses the same general syntax as the existing CUT
55 and FIELDQTY dialplan functions, which also manage lists.
56 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
57 obtaining realtime data from the dialplan.
58 * Added LOCAL_PEEK, which I have no idea how to use, but Leif Madsen wanted it.
59 Russell says it's, like, a scope resolution function for LOCAL variables.
60 Totally. Hopefully, that means more to you than it does to me.
64 * Scheduled meetme conferences may now have their end times extended by
66 * app_authenticate now gives the ability to select a prompt other than
68 * app_directory now pays attention to the searchcontexts setting in
69 voicemail.conf and will look through all contexts, if no context is
70 specified in the initial argument.
74 * res_jabber: autoprune has been disabled by default, to avoid misconfiguration
75 that would end up being interpreted as a bug once Asterisk started removing
76 the contacts from a user list.
77 * extensions.conf now allows you to use keyword "same" to define an extension
78 without actually specifying an extension. It uses exactly the same pattern
79 as previously used on the last "exten" line. For example:
80 exten => 123,1,NoOp(something)
81 same => n,SomethingElse()
82 * musiconhold.conf classes of type 'files' can now use relative directory paths,
83 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
84 * All deprecated CLI commands are removed from the sourcecode. They are now handled
85 by the new clialiases module. See cli_aliases.conf.sample file.
87 Asterisk Manager Interface
88 --------------------------
89 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
90 a non-empty value) in your request. If you do this, any pending AMI events will
91 *not* be included in the response to your request as they would normally, but
92 will be left in the event queue for the next request you make to retrieve. For
93 some applications, this will allow you to guarantee that you will only see
94 events in responses to 'WaitEvent' actions, and can better know when to expect them.
95 To know whether the Asterisk server supports this header or not, your client can
96 inspect the first response back from the server to see if it includes this header:
98 Pragma: SuppressEvents
100 If this is included, the server supports event suppression.
102 ------------------------------------------------------------------------------
103 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
104 ------------------------------------------------------------------------------
106 Device State Handling
107 ---------------------
108 * The event infrastructure in Asterisk got another big update to help support
109 distributed events. It currently supports distributed device state and
110 distributed Voicemail MWI (Message Waiting Indication). A new module has
111 been merged, res_ais, which facilitates communicating events between servers.
112 It uses the SAForum AIS (Service Availability Forum Application Interface
113 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
114 a cluster of Asterisk servers, and to share events between them. For more
115 information on setting this up, see doc/distributed_devstate.txt.
119 * Added a new dialplan function, AST_CONFIG(), which allows you to access
120 variables from an Asterisk configuration file.
121 * The JACK_HOOK function now has a c() option to supply a custom client name.
122 * Added two new dialplan functions from libspeex for audio gain control and
123 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
124 rx directions of a channel from the dialplan.
125 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
126 based on other parameters. The default is still to search based on the
127 forwarding station ID. However, there are new options that allow you to search
128 based on the message desk terminal ID, or the message desk number.
129 * TIMEOUT() has been modified to be accurate down to the millisecond.
130 * ENUM*() functions now include the following new options:
131 - 'u' returns the full URI and does not strip off the URI-scheme.
132 - 's' triggers ISN specific rewriting
133 - 'i' looks for branches into an Infrastructure ENUM tree
134 - 'd' for a direct DNS lookup without any flipping of digits.
135 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
136 * CHANNEL() now has options for the maximum, minimum, and standard or normal
137 deviation of jitter, rtt, and loss for a call using chan_sip.
139 DAHDI channel driver (chan_dahdi) Changes
140 ----------------------------------------
141 * Channels can now be configured using named sections in chan_dahdi.conf, just
142 like other channel drivers, including the use of templates.
143 * The default for pridialplan has changed from 'national' to 'unknown'.
147 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
148 to something that matches the pattern a hint will be created using the contents
149 and variables evaluated.
150 * Dialplan matching has been extended to allow an extension to return to the
151 PBX core to wait for more digits. This is done by using the new dialplan
152 application called "Incomplete". This will permit a whole new level of
153 extension control, by giving the administrator more control over early
154 matches employing one of the short-circuit pattern match operators. Note
155 that custom applications can trigger this same behavior by returning the
156 special value AST_PBX_INCOMPLETE.
160 * Directory now permits both first and last names to be matched at the same
161 time. In addition, the number of digits to enter of the name can be set in
162 the arguments to Directory; previously, you could enter only 3, regardless
163 of how many names are in your company. For large companies, this should be
165 * Voicemail now permits a mailbox setting to wrap around from first to last
166 messages, if the "messagewrap" option is set to a true value.
167 * Voicemail now permits an external script to be run, for password validation.
168 The script should output "VALID" or "INVALID" on stdout, depending upon the
169 wish to validate or invalidate the password given. Arguments are:
170 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
172 * Dial has a new option: F(context^extension^pri), which permits a callee to
173 continue in the dialplan, at the specified label, if the caller hangs up.
174 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
175 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
176 * The Jack application now has a c() option to supply a custom client name.
177 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
178 like the pre-existing whisper mode, except that the spy can also talk to the
179 participant on the bridged channel as well.
180 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
181 to be spoken instead of the channel name or number. For more information on the
182 use of this option, issue the command "core show application ChanSpy" from the
184 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
185 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
186 words, if using the 'd' option, it is not possible to enter a number to append to
187 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
188 change to whisper mode, and pressing 6 will change to barge mode.
189 * ExternalIVR now takes several options that affect the way it performs, as
190 well as having several new commands. Please see doc/externalivr.txt for the
191 complete documentation.
192 * Added ability to communicate over a TCP socket instead of forking a child process for the
193 ExternalIVR application.
194 * ChanIsAvail has a new option, 'a', which will return all available channels instead
195 of just the first one if you give the function more then one channel to check.
196 * PrivacyManager now takes an option where you can specify a context where the
197 given number will be matched. This way you have more control over who is allowed
198 and it stops the people who blindly enter 10 digits.
199 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
200 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
201 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
202 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
203 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
204 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
205 * The Dial() application no longer copies the language used by the caller to the callee's
206 channel. If you desire for the caller's channel's language to be used for file playback
207 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
208 * SendImage() no longer hangs up the channel on error; instead, it sets the
209 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
210 'UNSUPPORTED'. This change makes SendImage() more consistent with other
212 * Park has a new option, 's', which silences the announcement of the parking space number.
213 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
214 invalid input and will be assumed to mean that no timeout is desired.
218 * Added DNS manager support to registrations for peers referencing peer entries.
219 DNS manager runs in the background which allows DNS lookups to be run asynchronously
220 as well as periodically updating the IP address. These properties allow for
221 better performance as well as recovery in the event of an IP change.
222 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
223 load/reload of large numbers of peers/users by ~40x (for large lists of peers.
224 Initially, we saw 4x improvement in call setup/destruction, but at the time
225 of merging, this gain has disappeared; further research will be done to try
226 and restore this performance improvement. Astobj2 refcounting is now used
227 for users, peers, and dialogs. Users are encouraged to assist in regression
228 testing and problem reporting!
229 * Added ability to specify registration expiry time on a per registration basis in
231 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
233 * Added t38pt_usertpsource option. See sip.conf.sample for details.
234 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
235 * 'sip show peers' and 'sip show users' display their entries sorted in
236 alphabetical order, as opposed to the order they were in, in the config
238 * Videosupport now supports an additional option, "always", which always sets
239 up video RTP ports, even on clients that don't support it. This helps with
240 callfiles and certain transfers to ensure that if two video phones are
241 connected, they will always share video feeds.
245 * Existing DNS manager lookups extended to check for SRV records.
246 * IAX2 encryption support has been improved to support periodic key rotation
247 within a call for enhanced security. The option "keyrotate" has been
248 provided to disable this functionality to preserve backwards compatibility
249 with older versions of IAX2 that do not support key rotation.
253 * New CLI command, "config reload <file.conf>" which reloads any module that
254 references that particular configuration file. Also added "config list"
255 which shows which configuration files are in use.
256 * New CLI commands, "pri show version" and "ss7 show version" that will
257 display which version of libpri and libss7 are being used, respectively.
258 A new API call was added so trunk will now have to be compiled against
259 a versions of libpri and libss7 that have them or it will not know that
260 these libraries exist.
261 * The commands "core show globals", "core set global" and "core set chanvar" has
262 been deprecated in favor of the more semanticly correct "dialplan show globals",
263 "dialplan set chanvar" and "dialplan set global".
264 * New CLI command "dialplan show chanvar" to list all variables associated
265 with a given channel.
269 * Addresses managed by DNS manager now can check to see if there is a DNS
270 SRV record for a given domain and will use that hostname/port if present.
272 AMI - The manager (TCP/TLS/HTTP)
273 --------------------------------
274 * The Status command now takes an optional list of variables to display
275 along with channel status.
276 * The QueueEntry event now also includes the channel's uniqueid
280 * res_odbc no longer has a limit of 1023 total possible unshared connections,
281 as some people were running into this limit. This limit has been increased
286 * The TRANSFER queue log entry now includes the the caller's original
287 position in the transferred-from queue.
288 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
289 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
290 as well as an explanation about timeout options in general
294 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
295 adaptive capabilities. What this means in practical terms is that if your
296 realtime table lacks critical fields, Asterisk will now emit warnings to
297 that effect. Also, some of the realtime drivers have the ability (if
298 configured) to automatically add those columns to the table with the
299 correct type and length.
303 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
304 the 'setvar' option to cause a given audio file to be played upon completion
305 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
306 Skinny channels only.
307 * You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt
308 for more information.
309 * Config file variables may now be appended to, by using the '+=' append
310 operator. This is most helpful when working with long SQL queries in
311 func_odbc.conf, as the queries no longer need to be specified on a single
313 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
314 which will add a second to the billsec when the ending
315 time is set, if the number in the microseconds field of the end time is
316 greater than the number of microseconds in the answer time. This allows
317 users to count the 'initiated' seconds in their billing records.
319 ------------------------------------------------------------------------------
320 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
321 ------------------------------------------------------------------------------
323 AMI - The manager (TCP/TLS/HTTP)
324 --------------------------------
325 * Manager has undergone a lot of changes, all of them documented
326 in doc/manager_1_1.txt
327 * Manager version has changed to 1.1
328 * Added a new action 'CoreShowChannels' to list currently defined channels
329 and some information about them.
330 * Added a new action 'SIPshowregistry' to list SIP registrations.
331 * Added TLS support for the manager interface and HTTP server
332 * Added the URI redirect option for the built-in HTTP server
333 * The output of CallerID in Manager events is now more consistent.
334 CallerIDNum is used for number and CallerIDName for name.
335 * Enable https support for builtin web server.
336 See configs/http.conf.sample for details.
337 * Added a new action, GetConfigJSON, which can return the contents of an
338 Asterisk configuration file in JSON format. This is intended to help
339 improve the performance of AJAX applications using the manager interface
341 * SIP and IAX manager events now use "ChannelType" in all cases where we
342 indicate channel driver. Previously, we used a mixture of "Channel"
343 and "ChannelDriver" headers.
344 * Added a "Bridge" action which allows you to bridge any two channels that
345 are currently active on the system.
346 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
347 the voicemail users setup.
348 * Added 'DBDel' and 'DBDelTree' manager commands.
349 * cdr_manager now reports events via the "cdr" level, separating it from
350 the very verbose "call" level.
351 * Manager users are now stored in memory. If you change the manager account
352 list (delete or add accounts) you need to reload manager.
353 * Added Masquerade manager event for when a masquerade happens between
355 * Added "manager reload" command for the CLI
356 * Lots of commands that only provided information are now allowed under the
357 Reporting privilege, instead of only under Call or System.
358 * The IAX* commands now require either System or Reporting privilege, to
359 mirror the privileges of the SIP* commands.
360 * Added ability to retrieve list of categories in a config file.
361 * Added ability to retrieve the content of a particular category.
362 * Added ability to empty a context.
363 * Created new action to create a new file.
364 * Updated delete action to allow deletion by line number with respect to category.
365 * Added new action insert to add new variable to category at specified line.
366 * Updated action newcat to allow new category to be inserted in file above another
368 * Added new event "JitterBufStats" in the IAX2 channel
369 * Originate now requires the Originate privilege and, if you want to call out
370 to a subshell, it requires the System privilege, as well. This was done to
371 enhance manager security.
372 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
373 * New command: Atxfer. See doc/manager_1_1.txt for more details or
374 manager show command Atxfer from the CLI
375 * New command: IAXregistry. See doc/manager_1_1.txt for more details or
376 manager show command IAXregistry from the CLI
380 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
381 state in the dialplan, as well as creating custom device states that are
382 controllable from the dialplan.
383 * Extend CALLERID() function with "pres" and "ton" parameters to
384 fetch string representation of calling number presentation indicator
385 and numeric representation of type of calling number value.
386 * MailboxExists converted to dialplan function
387 * A new option to Dial() for telling IP phones not to count the call
388 as "missed" when dial times out and cancels.
389 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
390 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
391 held for any given channel. Also, locks are automatically freed when a
393 * Added HINT() dialplan function that allows retrieving hint information.
394 Hints are mappings between extensions and devices for the sake of
395 determining the state of an extension. This function can retrieve the list
396 of devices or the name associated with a hint.
397 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
399 * Added SYSINFO() dialplan function which allows retrieval of system information
400 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
401 the existence of a dialplan target.
402 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
403 upper and lower case, respectively.
404 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
405 ID for the call (not the Asterisk call ID or unique ID), provided that the
406 channel driver supports this. For SIP, you get the SIP call-ID for the
407 bridged channel which you can store in the CDR with a custom field.
411 * Added CLI permissions, config file: cli_permissions.conf
412 default is to allow all commands for every local user/group.
413 Also this new feature added three new CLI commands:
414 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
415 - cli reload permissions
416 - cli show permissions
417 * New CLI command "core show hint" (usage: core show hint <exten>)
418 * New CLI command "core show settings"
419 * Added 'core show channels count' CLI command.
420 * Added the ability to set the core debug and verbose values on a per-file basis.
421 * Added 'queue pause member' and 'queue unpause member' CLI commands
422 * Ability to set process limits ("ulimit") without restarting Asterisk
423 * Enhanced "agi debug" to print the channel name as a prefix to the debug
424 output to make debugging on busy systems much easier.
425 * New CLI commands "dialplan set extenpatternmatching true/false"
426 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
427 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
428 listed in the startup_commands section of cli.conf will get executed.
429 * Added a CLI command, "devstate change", which allows you to set custom device
430 states from the func_devstate module that provides the DEVICE_STATE() function
431 and handling of the "Custom:" devices.
432 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
433 sorted into the different possible callbacks, with the number of entries
434 currently scheduled for each. Gives you a feel for how busy the sip channel
436 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
437 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
438 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
442 * Improved NAT and STUN support.
443 chan_sip now can use port numbers in bindaddr, externip and externhost
444 options, as well as contact a STUN server to detect its external address
445 for the SIP socket. See sip.conf.sample, 'NAT' section.
446 * The default SIP useragent= identifier now includes the Asterisk version
447 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
448 If set, and the incoming request carries authentication info,
449 the username to match in the users list is taken from the Digest header
450 rather than from the From: field. This feature is considered experimental.
451 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
452 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
453 * The "localmask" setting was removed in version 1.2 and the reminder about it
454 being removed is now also removed.
455 * A new option "busylevel" for setting a level of calls where asterisk reports
456 a device as busy, to separate it from call-limit. This value is also added
457 to the SIP_PEER dialplan function.
458 * A new realtime family called "sipregs" is now supported to store SIP registration
459 data. If this family is defined, "sippeers" will be used for configuration and
460 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
461 registration data, as before.
462 * The SIPPEER function have new options for port address, call and pickup groups
463 * Added support for T.140 realtime text in SIP/RTP
464 * The "checkmwi" option has been removed from sip.conf, as it is no longer
465 required due to the restructuring of how MWI is handled. See the descriptions
466 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
467 for more information.
468 * Added rtpdest option to CHANNEL() dialplan function.
469 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
470 * SIP now adds a header to the CANCEL if the call was answered by another phone
471 in the same dial command, or if the new c option in dial() is used.
472 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
473 states it is not needed. For phones, however, that do require it the "registertrying" option
474 has been added so it can be enabled.
475 * A new option called "callcounter" (global/peer/user level) enables call counters needed
476 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
477 used to enable this functionality).
478 * New settings for timer T1 and timer B on a global level or per device. This makes it
479 possible to force timeout faster on non-responsive SIP servers. These settings are
480 considered advanced, so don't use them unless you have a problem.
481 * Added a dial string option to be able to set the To: header in an INVITE to any
483 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
484 the qualify frequency.
485 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
486 were not properly torn down due to network or endpoint failures during an established
488 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
489 configs/sip.conf.sample for more information on how it is used.
490 * Added a new configuration option "authfailureevents" that enables manager events when
491 a peer can't authenticate properly.
492 * Added DNS manager support to registrations for peers not referencing a peer entry.
496 * Added the trunkmaxsize configuration option to chan_iax2.
497 * Added the srvlookup option to iax.conf
498 * Added support for OSP. The token is set and retrieved through the CHANNEL()
501 XMPP Google Talk/Jingle changes
502 -------------------------------
503 * Added the bindaddr option to gtalk.conf.
507 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
508 * Proper codec support in chan_skinny.
509 * Added settings for IP and Ethernet QoS requests
513 * Added separate settings for media QoS in mgcp.conf
515 Console Channel Driver changes
516 ------------------------------
517 * Added experimental support for video send & receive to chan_oss.
518 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
521 Phone channel changes (chan_phone)
522 ----------------------------------
523 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
525 H.323 channel Changes
526 ---------------------
527 * H323 remote hold notification support added (by NOTIFY message
528 and/or H.450 supplementary service)
530 Local channel changes
531 ---------------------
532 * The device state functionality in the Local channel driver has been updated
533 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
534 to just UNKNOWN if the extension exists.
535 * Added jitterbuffer support for chan_local. This allows you to use the
536 generic jitterbuffer on incoming calls going to Asterisk applications.
537 For example, this would allow you to use a jitterbuffer for an incoming
538 SIP call to Voicemail by putting a Local channel in the middle. This
539 feature is enabled by using the 'j' option in the Dial string to the Local
540 channel in conjunction with the existing 'n' option for local channels.
541 * A 'b' option has been added which causes chan_local to return the actual channel
542 that is behind it when queried. This is useful for transfer scenarios as the
543 actual channel will be transferred, not the Local channel.
545 Agent channel changes
546 ----------------------
547 * The ackcall and endcall options are now supplemented with options acceptdtmf
548 and enddtmf. These allow for the DTMF keypress to be configurable. The options
549 default to their old hard-coded values ('#' and '*' respectively) so this should
550 not break any existing agent installations.
552 DAHDI channel driver (chan_dahdi) Changes
553 ----------------------------------------
554 * SS7 support (via libss7 library)
555 * In India, some carriers transmit CID via dtmf. Some code has been added
556 that will handle some situations. The cidstart=polarity_IN choice has been added for
557 those carriers that transmit CID via dtmf after a polarity change.
558 * CID matching information is now shown when doing 'dialplan show'.
559 * Added dahdi show version CLI command.
560 * Added setvar support to chan_dahdi.conf channel entries.
561 * Added two new options: mwimonitor and mwimonitornotify. These options allow
562 you to enable MWI monitoring on FXO lines. When the MWI state changes,
563 the script specified in the mwimonitornotify option is executed. An internal
564 event indicating the new state of the mailbox is also generated, so that
565 the normal MWI facilities in Asterisk work as usual.
566 * Added signalling type 'auto', which attempts to use the same signalling type
567 for a channel as configured in DAHDI. This is primarily designed for analog
568 ports, but will also work for digital ports that are configured for FXS or FXO
569 signalling types. This mode is also the default now, so if your chan_dahdi.conf
570 does not specify signalling for a channel (which is unlikely as the sample
571 configuration file has always recommended specifying it for every channel) then
572 the 'auto' mode will be used for that channel if possible.
573 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
574 state for a channel; also ensured that the DNDState Manager event is
575 emitted no matter how the DND state is set or cleared.
579 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
580 configs/unistim.conf.sample for details. This new channel driver allows
581 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
582 * Added a new channel driver, chan_console, which uses portaudio as a cross
583 platform audio interface. It was written as a channel driver that would
584 work with Mac CoreAudio, but portaudio supports a number of other audio
585 interfaces, as well. Note that this channel driver requires v19 or higher
586 of portaudio; older versions have a different API.
590 * Added the ability to specify arguments to the Dial application when using
591 the DUNDi switch in the dialplan.
592 * Added the ability to set weights for responses dynamically. This can be
593 done using a global variable or a dialplan function. Using the SHELL()
594 function would allow you to have an external script set the weight for
596 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
597 functions will allow you to initiate a DUNDi query from the dialplan,
598 find out how many results there are, and access each one.
602 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
603 functions will allow you to initiate an ENUM lookup from the dialplan,
604 and Asterisk will cache the results. ENUMRESULT can be used to access
605 the results without doing multiple DNS queries.
609 * Added the ability to customize which sound files are used for some of the
610 prompts within the Voicemail application by changing them in voicemail.conf
611 * Added the ability for the "voicemail show users" CLI command to show users
612 configured by the dynamic realtime configuration method.
613 * MWI (Message Waiting Indication) handling has been significantly
614 restructured internally to Asterisk. It is now totally event based
615 instead of polling based. The voicemail application will notify other
616 modules that have subscribed to MWI events when something in the mailbox
618 This also means that if any other entity outside of Asterisk is changing
619 the contents of mailboxes, then the voicemail application still needs to
620 poll for changes. Examples of situations that would require this option
621 are web interfaces to voicemail or an email client in the case of using
622 IMAP storage. So, two new options have been added to voicemail.conf
623 to account for this: "pollmailboxes" and "pollfreq". See the sample
624 configuration file for details.
625 * Added "tw" language support
626 * Added support for storage of greetings using an IMAP server
627 * Added ability to customize forward, reverse, stop, and pause keys for message playback
628 * SMDI is now enabled in voicemail using the smdienable option.
629 * A "lockmode" option has been added to asterisk.conf to configure the file
630 locking method used for voicemail, and potentially other things in the
631 future. The default is the old behavior, lockfile. However, there is a
632 new method, "flock", that uses a different method for situations where the
633 lockfile will not work, such as on SMB/CIFS mounts.
634 * Added the ability to backup deleted messages, to ease recovery in the case
635 that a user accidentally deletes a message, and discovers that they need it.
636 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
637 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
638 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
639 voicemail boxes. The SMDI interface can also poll for MWI changes when some
640 outside entity is modifying the state of the mailbox (such as IMAP storage or
641 a web interface of some kind).
642 * Added the support for marking messages as "urgent." There are two methods to accomplish
643 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
644 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
645 the message as urgent after he has recorded a voicemail by following the voice instructions.
646 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
651 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
652 used across multiple queues.
653 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
654 setqueueentryvar options for each queue, see queues.conf.sample for details.
655 * Added keepstats option to queues.conf which will keep queue
656 statistics during a reload.
657 * setinterfacevar option in queues.conf also now sets a variable
658 called MEMBERNAME which contains the member's name.
659 * Added 'Strategy' field to manager event QueueParams which represents
660 the queue strategy in use.
661 * Added option to run macro when a queue member is connected to a caller,
662 see queues.conf.sample for details.
663 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
664 does not count paused queue members as unavailable.
665 * Added min-announce-frequency option to queues.conf which allows you to control the
666 minimum amount of time between queue announcements for use when the caller's queue
667 position changes frequently.
668 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
670 * Added ability for non-realtime queues to have realtime members
671 * Added the "linear" strategy to queues.
672 * Added the "wrandom" strategy to queues.
673 * Added new channel variable QUEUE_MIN_PENALTY
674 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
675 rules in queuerules.conf. See configs/queuerules.conf.sample for details
676 * Added a new parameter for member definition, called state_interface. This may be
677 used so that a member may be called via one interface but have a different interface's
678 device state reported.
679 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
680 specified by the periodic-announce option, then one will be chosen randomly when it is time
681 to play a periodic announcment
682 * New configuration options: announce-position now takes two more values in addition to "yes" and
683 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
684 announce-position-limit. By setting announce-position to "limit" callers will only have their
685 position announced if their position is less than what is specified by announce-position-limit.
686 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
687 will be told that their are more than announce-position-limit callers waiting.
688 * Two new queue log events have been added. An ADDMEMBER event will be logged
689 when a realtime queue member is added and a REMOVEMEMBER event will be logged
690 when a realtime queue member is removed. Since there is no calling channel associated
691 with these events, the string "REALTIME" is placed where the channel's unique id
693 * The configuration method for the "joinempty" and "leavewhenempty" options has
694 changed to a comma-separated list of methods of determining member availability
695 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
696 values are still accepted for backwards-compatibility, though.
700 * The 'o' option to provide an optimization has been removed and its functionality
701 has been enabled by default.
702 * When a conference is created, the UNIQUEID of the channel that caused it to be
703 created is stored. Then, every channel that joins the conference will have the
704 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
705 callers that come and go from long standing conferences.
706 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
707 except it does operations on a channel by name, instead of number in a conference.
708 This is a very useful feature in combination with the 'X' option to ChanSpy.
709 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
711 * Added new RealTime functionality to provide support for scheduled conferencing.
712 This includes optional messages to the caller if they attempt to join before
713 the schedule start time, or to allow the caller to join the conference early.
714 Also included is optional support for limiting the number of callers per
716 * Added the S() and L() options to the MeetMe application. These are pretty
717 much identical to the S() and L() options to Dial(). They let you set
718 timeouts for the conference, as well as have warning sounds played to
719 let the caller know how much time is left, and when it is running out.
720 * Added the ability to do "meetme concise" with the "meetme" CLI command.
721 This extends the concise capabilities of this CLI command to include
722 listing all conferences, instead of an addition to the other sub commands
723 for the "meetme" command.
724 * Added the ability to specify the music on hold class used to play into the
725 conference when there is only one member and the M option is used.
726 * Added MEETME_INFO dialplan function which provides a way to query
727 various properties of a Meetme conference.
729 Other Dialplan Application Changes
730 ----------------------------------
731 * Argument support for Gosub application
732 * From the to-do lists: straighten out the app timeout args:
733 Wait() app now really does 0.3 seconds- was truncating arg to an int.
734 WaitExten() same as Wait().
735 Congestion() - Now takes floating pt. argument.
736 Busy() - now takes floating pt. argument.
737 Read() - timeout now can be floating pt.
738 WaitForRing() now takes floating pt timeout arg.
739 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
740 * Added 's' option to Page application.
741 * Added an optional timeout argument to the Page application.
742 * Added 'E', 'V', and 'P' commands to ExternalIVR.
743 * Added 'o' and 'X' options to Chanspy.
744 * Added a new dialplan application, Bridge, which allows you to bridge the
745 calling channel to any other active channel on the system.
746 * Added the ability to specify a music on hold class to play instead of ringing
747 for the SLATrunk application.
748 * The Read application no longer exits the dialplan on error. Instead, it sets
749 READSTATUS to ERROR, which you can catch and handle separately.
750 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
751 of asking for verification of each name, one at a time.
752 * Privacy() no longer uses privacy.conf, as all options are specifyable as
753 direct options to the app.
754 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
756 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
757 * The ChannelRedirect application no longer exits the dialplan if the given channel
758 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
759 or NOCHANNEL if the given channel was not found.
760 * The silencethreshold setting that was previously configurable in multiple
761 applications is now settable globally via dsp.conf.
763 Music On Hold Changes
764 ---------------------
765 * A new option, "digit", has been added for music on hold classes in
766 musiconhold.conf. If this is set for a music on hold class, a caller
767 listening to music on hold can press this digit to switch to listening
768 to this music on hold class.
769 * Support for realtime music on hold has been added.
770 * In conjunction with the realtime music on hold, a general section has
771 been added to musiconhold.conf, its sole variable is cachertclasses. If this
772 is set, then music on hold classes found in realtime will be cached in memory.
776 * AEL upgraded to use the Gosub with Arguments instead
777 of Macro application, to hopefully reduce the problems
778 seen with the artificially low stack ceiling that
779 Macro bumps into. Macros can only call other Macros
780 to a depth of 7. Tests run using gosub, show depths
781 limited only by virtual memory. A small test demonstrated
782 recursive call depths of 100,000 without problems.
783 -- in addition to this, all apps that allowed a macro
784 to be called, as in Dial, queues, etc, are now allowing
785 a gosub call in similar fashion.
786 * AEL now generates LOCAL(argname) declarations when it
787 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
788 etc. That makes the arguments local in scope. The user
789 can define their own local variables in macros, now,
790 by saying "local myvar=someval;" or using Set() in this
791 fashion: Set(LOCAL(myvar)=someval); ("local" is now
793 * utils/conf2ael introduced. Will convert an extensions.conf
794 file into extensions.ael. Very crude and unfinished, but
795 will be improved as time goes by. Should be useful for a
796 first pass at conversion.
797 * aelparse will now read extensions.conf to see if a referenced
798 macro or context is there before issueing a warning.
799 * AEL parser sets a local channel variable ~~EXTEN~~, to
800 preserve the value of ${EXTEN} thru switch statements.
801 * New operator in $[...] expressions: the ~~ operator serves
802 as a concatenation operator. AT THE MOMENT, it is really only
803 necessary and useful in AEL, especially in if() expressions.
804 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
805 any enclosing double-quotes, and evaluate to the value of a
806 concatenated with the value of b. For example if a is set to
807 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
811 Call Features (res_features) Changes
812 ------------------------------------
813 * Added the parkedcalltransfers option to features.conf
814 * The built-in method for doing attended transfers has been updated to
815 include some new options that allow you to have the transferee sent
816 back to the person that did the transfer if the transfer is not successful.
817 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
818 in features.conf.sample.
819 * Added support for configuring named groups of custom call features in
820 features.conf. This means that features can be written a single time, and
821 then mapped into groups of features for different key mappings or easier
823 * Updated the ParkedCall application to allow you to not specify a parking
824 extension. If you don't specify a parking space to pick up, it will grab
825 the first one available.
826 * Added cli command 'features reload' to reload call features from features.conf
827 * Moved into core asterisk binary.
828 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
830 Language Support Changes
831 ------------------------
832 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
833 * Added support for the Hungarian language for saying numbers, dates, and times.
837 * Added SPEECH commands for speech recognition. A complete listing can be found
839 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
840 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
841 does not behave as expected; the native command needs to be used, instead.
845 * Added rotatestrategy option to logger.conf, along with two new options:
846 "timestamp" which will use the time to name the logger files instead of
847 sequence number; and "rotate", which rotates the names of the logfiles,
848 similar to the way syslog rotates files.
849 * Added exec_after_rotate option to logger.conf, which allows a system
850 command to be run after rotation. This is primarily useful with
851 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
852 and to ensure that the oldest log file gets deleted.
853 * Added realtime support for the queue log
857 * The cdr_manager module has a [mappings] feature, like cdr_custom,
858 to add fields to the manager event from the CDR variables.
859 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
860 backend database CDR table. Specifically, additional, non-standard
861 columns are supported, merely by setting the corresponding CDR variable in
862 your dialplan. In addition, you may alias any column to another name (for
863 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
864 simply "alias src => ANI" in the configuration file). Records may be
865 posted to more than one backend, simply by specifying multiple categories
866 in the configuration file. And finally, you may filter which CDRs get
867 posted to each backend, by specifying a filter (which the record must
868 match) for the particular category. Filters are additive (meaning all
869 rules must match to post that CDR).
870 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
871 module. Specifically, you may add additional columns into the table and
872 they will be set, if you set the corresponding CDR variable name. Also,
873 if you omit columns in your database table, they will be silently skipped
874 (but a record will still be inserted, based on what columns remain). Note
875 that the other two features from cdr_adaptive_odbc (alias and filter) are
876 not currently supported.
877 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
878 has been disabled using the NoCDR application.
880 Miscellaneous New Modules
881 -------------------------
882 * Added a new CDR module, cdr_sqlite3_custom.
883 * Added a new realtime configuration module, res_config_sqlite
884 * Added a new codec translation module, codec_resample, which re-samples
885 signed linear audio between 8 kHz and 16 kHz to help support wideband
887 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
888 based on configuration templates that use Asterisk dialplan function and
889 variable substitution. It should be possible to create phone profiles and
890 templates that work for the majority of phones provisioned over http. It
891 is currently only intended to provision a single user account per phone.
892 An example profile and set of templates for Polycom phones is provided.
893 NOTE: Polycom firmware is not included, but should be placed in
894 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
895 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
896 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
897 provided; there is a JACK() application, and a JACK_HOOK() function. Both
898 interfaces create an input and output JACK port. The application makes
899 these ports the endpoint of the call. The audio coming from the channel
900 goes out the output port and whatever comes back in on the input port is
901 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
902 audiohook on the channel. This lets you run the audio coming from a
903 channel through JACK, and whatever comes back in is what gets forwarded
904 on as the channel's audio. This is very useful for building custom
905 vocoders or doing recording or analysis of the channel's audio in another
907 * Added a new module, res_config_curl, which permits using a HTTP POST url
908 to retrieve, create, update, and delete realtime information from a remote
909 web server. Note that this module requires func_curl.so to be loaded for
910 backend functionality.
911 * Added a new module, res_config_ldap, which permits the use of an LDAP
912 server for realtime data access.
913 * Added support for writing and running your dialplan in lua using the pbx_lua
914 module. See configs/extensions.lua.sample for examples of how to do this.
918 * Ability to use libcap to set high ToS bits when non-root
919 on Linux. If configure is unable to find libcap then you
920 can use --with-cap to specify the path.
921 * Added maxfiles option to options section of asterisk.conf which allows you to specify
922 what Asterisk should set as the maximum number of open files when it loads.
923 * Added the jittertargetextra configuration option.
924 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
925 configuration files for the IP channel drivers. The new option is "cos".
926 This information is also documented in doc/qos.tex, or the IP Quality of Service
927 section of asterisk.pdf.
928 * When originating a call using AMI or pbx_spool that fails the reason for failure
929 will now be available in the failed extension using the REASON dialplan variable.
930 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
931 It allows you to configure a prefix for auto-monitor recordings.
932 * A new extension pattern matching algorithm, based on a trie, is introduced
933 here, that could noticeably speed up mid-sized to large dialplans.
934 It is NOT used by default, as duplicating the behaviour of the old pattern
935 matcher is still under development. A config file option, in extensions.conf,
936 in the [general] section, called "extenpatternmatchingnew", is by default
937 set to false; setting that to true will force the use of the new algorithm.
938 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
939 be used to switch the algorithms at run time.
940 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
941 specifying which socket to use to connect to the running Asterisk daemon
943 * Performance enhancements to the sched facility, which is used in
944 the channel drivers, etc. Added hashtabs and doubly-linked lists
945 to speed up deletion; start at the beginning or end of list to
947 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
948 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
949 Added regression tests to the tests/ dir, also.
950 * Added a refcount trace feature to astobj2 for those trying to balance
951 object creation, deletion; work, play; space and time. See the
952 notes in astobj2.h. Also, see utils/refcounter as well, as a
953 quick way to find unbalanced refcounts in what could be a sea
954 of objects that were balanced.
955 * Added logging to 'make update' command. See update.log
956 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
957 do not come from the remote party.
958 * Added the 'n' option to the SpeechBackground application to tell it to not
959 answer the channel if it has not already been answered.
960 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
961 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
963 * iLBC source code no longer included (see UPGRADE.txt for details)
964 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
965 deadlock is detected, a backtrace of the stack which led to the lock calls
966 will be output to the CLI.
967 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
968 the "core show locks" CLI command will give lock information output as well
969 as a backtrace of the stack which led to the lock calls.
970 * users.conf now sports an optional alternateexts property, which permits
971 allocation of additional extensions which will reach the specified user.
972 * A new option for the configure script, --enable-internal-poll, has been added
973 for use with systems which may have a buggy implementation of the poll system
974 call. If you notice odd behavior such as the CLI being unresponsive on remote
975 consoles, you may want to try using this option. This option is enabled by default
976 on Darwin systems since it is known that the Darwin poll() implementation has
981 * In addition to timing from DAHDI, there is a new timing module called
982 res_timing_timerfd. In order to use this, you must be running Linux with
983 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
984 script will be able to tell if you have the requirements. From menuselect, select
985 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob