1 ------------------------------------------------------------------------------
2 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
3 ------------------------------------------------------------------------------
7 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
8 Snom phones use this for call pickup of extensions that the phone is
13 * Added a new dialplan function, CURLOPT, which permits setting various
14 options that may be useful with the CURL dialplan function, such as
15 cookies, proxies, connection timeouts, passwords, etc.
16 * Permit the syntax and synopsis fields of the corresponding dialplan
17 functions to be individually set from func_odbc.conf.
21 * res_jabber: autoprune has been disabled by default, to avoid misconfiguration
22 that would end up being interpreted as a bug once Asterisk started removing
23 the contacts from a user list.
25 ------------------------------------------------------------------------------
26 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
27 ------------------------------------------------------------------------------
31 * The event infrastructure in Asterisk got another big update to help support
32 distributed events. It currently supports distributed device state and
33 distributed Voicemail MWI (Message Waiting Indication). A new module has
34 been merged, res_ais, which facilitates communicating events between servers.
35 It uses the SAForum AIS (Service Availability Forum Application Interface
36 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
37 a cluster of Asterisk servers, and to share events between them. For more
38 information on setting this up, see doc/distributed_devstate.txt.
42 * Added a new dialplan function, AST_CONFIG(), which allows you to access
43 variables from an Asterisk configuration file.
44 * The JACK_HOOK function now has a c() option to supply a custom client name.
45 * Added two new dialplan functions from libspeex for audio gain control and
46 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
47 rx directions of a channel from the dialplan.
48 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
49 based on other parameters. The default is still to search based on the
50 forwarding station ID. However, there are new options that allow you to search
51 based on the message desk terminal ID, or the message desk number.
52 * TIMEOUT() has been modified to be accurate down to the millisecond.
53 * ENUM*() functions now include the following new options:
54 - 'u' returns the full URI and does not strip off the URI-scheme.
55 - 's' triggers ISN specific rewriting
56 - 'i' looks for branches into an Infrastructure ENUM tree
57 - 'd' for a direct DNS lookup without any flipping of digits.
58 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
59 * CHANNEL() now has options for the maximum, minimum, and standard or normal
60 deviation of jitter, rtt, and loss for a call using chan_sip.
62 DAHDI channel driver (chan_dahdi) Changes
63 ----------------------------------------
64 * Channels can now be configured using named sections in chan_dahdi.conf, just
65 like other channel drivers, including the use of templates.
66 * The default for pridialplan has changed from 'national' to 'unknown'.
70 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
71 to something that matches the pattern a hint will be created using the contents
72 and variables evaluated.
73 * Dialplan matching has been extended to allow an extension to return to the
74 PBX core to wait for more digits. This is done by using the new dialplan
75 application called "Incomplete". This will permit a whole new level of
76 extension control, by giving the administrator more control over early
77 matches employing one of the short-circuit pattern match operators. Note
78 that custom applications can trigger this same behavior by returning the
79 special value AST_PBX_INCOMPLETE.
83 * Directory now permits both first and last names to be matched at the same
84 time. In addition, the number of digits to enter of the name can be set in
85 the arguments to Directory; previously, you could enter only 3, regardless
86 of how many names are in your company. For large companies, this should be
88 * Voicemail now permits a mailbox setting to wrap around from first to last
89 messages, if the "messagewrap" option is set to a true value.
90 * Voicemail now permits an external script to be run, for password validation.
91 The script should output "VALID" or "INVALID" on stdout, depending upon the
92 wish to validate or invalidate the password given. Arguments are:
93 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
95 * Dial has a new option: F(context^extension^pri), which permits a callee to
96 continue in the dialplan, at the specified label, if the caller hangs up.
97 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
98 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
99 * The Jack application now has a c() option to supply a custom client name.
100 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
101 like the pre-existing whisper mode, except that the spy can also talk to the
102 participant on the bridged channel as well.
103 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
104 to be spoken instead of the channel name or number. For more information on the
105 use of this option, issue the command "core show application ChanSpy" from the
107 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
108 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
109 words, if using the 'd' option, it is not possible to enter a number to append to
110 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
111 change to whisper mode, and pressing 6 will change to barge mode.
112 * ExternalIVR now takes several options that affect the way it performs, as
113 well as having several new commands. Please see doc/externalivr.txt for the
114 complete documentation.
115 * Added ability to communicate over a TCP socket instead of forking a child process for the
116 ExternalIVR application.
117 * ChanIsAvail has a new option, 'a', which will return all available channels instead
118 of just the first one if you give the function more then one channel to check.
119 * PrivacyManager now takes an option where you can specify a context where the
120 given number will be matched. This way you have more control over who is allowed
121 and it stops the people who blindly enter 10 digits.
122 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
123 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
124 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
125 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
126 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
127 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
128 * The Dial() application no longer copies the language used by the caller to the callee's
129 channel. If you desire for the caller's channel's language to be used for file playback
130 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
131 * SendImage() no longer hangs up the channel on error; instead, it sets the
132 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
133 'UNSUPPORTED'. This change makes SendImage() more consistent with other
135 * Park has a new option, 's', which silences the announcement of the parking space number.
139 * Added DNS manager support to registrations for peers referencing peer entries.
140 DNS manager runs in the background which allows DNS lookups to be run asynchronously
141 as well as periodically updating the IP address. These properties allow for
142 better performance as well as recovery in the event of an IP change.
143 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
144 load/reload of large numbers of peers/users by ~40x (for large lists of peers.
145 Initially, we saw 4x improvement in call setup/destruction, but at the time
146 of merging, this gain has disappeared; further research will be done to try
147 and restore this performance improvement. Astobj2 refcounting is now used
148 for users, peers, and dialogs. Users are encouraged to assist in regression
149 testing and problem reporting!
150 * Added ability to specify registration expiry time on a per registration basis in
152 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
154 * Added t38pt_usertpsource option. See sip.conf.sample for details.
155 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
156 * 'sip show peers' and 'sip show users' display their entries sorted in
157 alphabetical order, as opposed to the order they were in, in the config
159 * Videosupport now supports an additional option, "always", which always sets
160 up video RTP ports, even on clients that don't support it. This helps with
161 callfiles and certain transfers to ensure that if two video phones are
162 connected, they will always share video feeds.
166 * Existing DNS manager lookups extended to check for SRV records.
167 * IAX2 encryption support has been improved to support periodic key rotation
168 within a call for enhanced security. The option "keyrotate" has been
169 provided to disable this functionality to preserve backwards compatibility
170 with older versions of IAX2 that do not support key rotation.
174 * New CLI command, "config reload <file.conf>" which reloads any module that
175 references that particular configuration file. Also added "config list"
176 which shows which configuration files are in use.
177 * New CLI commands, "pri show version" and "ss7 show version" that will
178 display which version of libpri and libss7 are being used, respectively.
179 A new API call was added so trunk will now have to be compiled against
180 a versions of libpri and libss7 that have them or it will not know that
181 these libraries exist.
182 * The commands "core show globals", "core set global" and "core set chanvar" has
183 been deprecated in favor of the more semanticly correct "dialplan show globals",
184 "dialplan set chanvar" and "dialplan set global".
185 * New CLI command "dialplan show chanvar" to list all variables associated
186 with a given channel.
190 * Addresses managed by DNS manager now can check to see if there is a DNS
191 SRV record for a given domain and will use that hostname/port if present.
193 AMI - The manager (TCP/TLS/HTTP)
194 --------------------------------
195 * The Status command now takes an optional list of variables to display
196 along with channel status.
200 * res_odbc no longer has a limit of 1023 total possible unshared connections,
201 as some people were running into this limit. This limit has been increased
206 * The TRANSFER queue log entry now includes the the caller's original
207 position in the transferred-from queue.
208 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
209 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
210 as well as an explanation about timeout options in general
214 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
215 adaptive capabilities. What this means in practical terms is that if your
216 realtime table lacks critical fields, Asterisk will now emit warnings to
217 that effect. Also, some of the realtime drivers have the ability (if
218 configured) to automatically add those columns to the table with the
219 correct type and length.
223 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
224 the 'setvar' option to cause a given audio file to be played upon completion
225 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
226 Skinny channels only.
227 * You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt
228 for more information.
229 * Config file variables may now be appended to, by using the '+=' append
230 operator. This is most helpful when working with long SQL queries in
231 func_odbc.conf, as the queries no longer need to be specified on a single
233 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
234 which will add a second to the billsec when the ending
235 time is set, if the number in the microseconds field of the end time is
236 greater than the number of microseconds in the answer time. This allows
237 users to count the 'initiated' seconds in their billing records.
239 ------------------------------------------------------------------------------
240 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
241 ------------------------------------------------------------------------------
243 AMI - The manager (TCP/TLS/HTTP)
244 --------------------------------
245 * Manager has undergone a lot of changes, all of them documented
246 in doc/manager_1_1.txt
247 * Manager version has changed to 1.1
248 * Added a new action 'CoreShowChannels' to list currently defined channels
249 and some information about them.
250 * Added a new action 'SIPshowregistry' to list SIP registrations.
251 * Added TLS support for the manager interface and HTTP server
252 * Added the URI redirect option for the built-in HTTP server
253 * The output of CallerID in Manager events is now more consistent.
254 CallerIDNum is used for number and CallerIDName for name.
255 * Enable https support for builtin web server.
256 See configs/http.conf.sample for details.
257 * Added a new action, GetConfigJSON, which can return the contents of an
258 Asterisk configuration file in JSON format. This is intended to help
259 improve the performance of AJAX applications using the manager interface
261 * SIP and IAX manager events now use "ChannelType" in all cases where we
262 indicate channel driver. Previously, we used a mixture of "Channel"
263 and "ChannelDriver" headers.
264 * Added a "Bridge" action which allows you to bridge any two channels that
265 are currently active on the system.
266 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
267 the voicemail users setup.
268 * Added 'DBDel' and 'DBDelTree' manager commands.
269 * cdr_manager now reports events via the "cdr" level, separating it from
270 the very verbose "call" level.
271 * Manager users are now stored in memory. If you change the manager account
272 list (delete or add accounts) you need to reload manager.
273 * Added Masquerade manager event for when a masquerade happens between
275 * Added "manager reload" command for the CLI
276 * Lots of commands that only provided information are now allowed under the
277 Reporting privilege, instead of only under Call or System.
278 * The IAX* commands now require either System or Reporting privilege, to
279 mirror the privileges of the SIP* commands.
280 * Added ability to retrieve list of categories in a config file.
281 * Added ability to retrieve the content of a particular category.
282 * Added ability to empty a context.
283 * Created new action to create a new file.
284 * Updated delete action to allow deletion by line number with respect to category.
285 * Added new action insert to add new variable to category at specified line.
286 * Updated action newcat to allow new category to be inserted in file above another
288 * Added new event "JitterBufStats" in the IAX2 channel
289 * Originate now requires the Originate privilege and, if you want to call out
290 to a subshell, it requires the System privilege, as well. This was done to
291 enhance manager security.
292 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
293 * New command: Atxfer. See doc/manager_1_1.txt for more details or
294 manager show command Atxfer from the CLI
298 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
299 state in the dialplan, as well as creating custom device states that are
300 controllable from the dialplan.
301 * Extend CALLERID() function with "pres" and "ton" parameters to
302 fetch string representation of calling number presentation indicator
303 and numeric representation of type of calling number value.
304 * MailboxExists converted to dialplan function
305 * A new option to Dial() for telling IP phones not to count the call
306 as "missed" when dial times out and cancels.
307 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
308 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
309 held for any given channel. Also, locks are automatically freed when a
311 * Added HINT() dialplan function that allows retrieving hint information.
312 Hints are mappings between extensions and devices for the sake of
313 determining the state of an extension. This function can retrieve the list
314 of devices or the name associated with a hint.
315 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
317 * Added SYSINFO() dialplan function which allows retrieval of system information
318 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
319 the existence of a dialplan target.
320 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
321 upper and lower case, respectively.
322 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
323 ID for the call (not the Asterisk call ID or unique ID), provided that the
324 channel driver supports this. For SIP, you get the SIP call-ID for the
325 bridged channel which you can store in the CDR with a custom field.
329 * New CLI command "core show hint" (usage: core show hint <exten>)
330 * New CLI command "core show settings"
331 * Added 'core show channels count' CLI command.
332 * Added the ability to set the core debug and verbose values on a per-file basis.
333 * Added 'queue pause member' and 'queue unpause member' CLI commands
334 * Ability to set process limits ("ulimit") without restarting Asterisk
335 * Enhanced "agi debug" to print the channel name as a prefix to the debug
336 output to make debugging on busy systems much easier.
337 * New CLI commands "dialplan set extenpatternmatching true/false"
338 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
339 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
340 listed in the startup_commands section of cli.conf will get executed.
341 * Added a CLI command, "devstate change", which allows you to set custom device
342 states from the func_devstate module that provides the DEVICE_STATE() function
343 and handling of the "Custom:" devices.
344 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
345 sorted into the different possible callbacks, with the number of entries
346 currently scheduled for each. Gives you a feel for how busy the sip channel
348 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
349 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
350 (Done by lmadsen, junky and mvanbaak during the AstriDevCon)
354 * Improved NAT and STUN support.
355 chan_sip now can use port numbers in bindaddr, externip and externhost
356 options, as well as contact a STUN server to detect its external address
357 for the SIP socket. See sip.conf.sample, 'NAT' section.
358 * The default SIP useragent= identifier now includes the Asterisk version
359 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
360 If set, and the incoming request carries authentication info,
361 the username to match in the users list is taken from the Digest header
362 rather than from the From: field. This feature is considered experimental.
363 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
364 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
365 * The "localmask" setting was removed in version 1.2 and the reminder about it
366 being removed is now also removed.
367 * A new option "busylevel" for setting a level of calls where asterisk reports
368 a device as busy, to separate it from call-limit. This value is also added
369 to the SIP_PEER dialplan function.
370 * A new realtime family called "sipregs" is now supported to store SIP registration
371 data. If this family is defined, "sippeers" will be used for configuration and
372 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
373 registration data, as before.
374 * The SIPPEER function have new options for port address, call and pickup groups
375 * Added support for T.140 realtime text in SIP/RTP
376 * The "checkmwi" option has been removed from sip.conf, as it is no longer
377 required due to the restructuring of how MWI is handled. See the descriptions
378 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
379 for more information.
380 * Added rtpdest option to CHANNEL() dialplan function.
381 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
382 * SIP now adds a header to the CANCEL if the call was answered by another phone
383 in the same dial command, or if the new c option in dial() is used.
384 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
385 states it is not needed. For phones, however, that do require it the "registertrying" option
386 has been added so it can be enabled.
387 * A new option called "callcounter" (global/peer/user level) enables call counters needed
388 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
389 used to enable this functionality).
390 * New settings for timer T1 and timer B on a global level or per device. This makes it
391 possible to force timeout faster on non-responsive SIP servers. These settings are
392 considered advanced, so don't use them unless you have a problem.
393 * Added a dial string option to be able to set the To: header in an INVITE to any
395 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
396 the qualify frequency.
397 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
398 were not properly torn down due to network or endpoint failures during an established
400 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
401 configs/sip.conf.sample for more information on how it is used.
402 * Added a new configuration option "authfailureevents" that enables manager events when
403 a peer can't authenticate properly.
404 * Added DNS manager support to registrations for peers not referencing a peer entry.
408 * Added the trunkmaxsize configuration option to chan_iax2.
409 * Added the srvlookup option to iax.conf
410 * Added support for OSP. The token is set and retrieved through the CHANNEL()
413 XMPP Google Talk/Jingle changes
414 -------------------------------
415 * Added the bindaddr option to gtalk.conf.
419 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
420 * Proper codec support in chan_skinny.
421 * Added settings for IP and Ethernet QoS requests
425 * Added separate settings for media QoS in mgcp.conf
427 Console Channel Driver changes
428 ------------------------------
429 * Added experimental support for video send & receive to chan_oss.
430 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
433 Phone channel changes (chan_phone)
434 ----------------------------------
435 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
437 H.323 channel Changes
438 ---------------------
439 * H323 remote hold notification support added (by NOTIFY message
440 and/or H.450 supplementary service)
442 Local channel changes
443 ---------------------
444 * The device state functionality in the Local channel driver has been updated
445 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
446 to just UNKNOWN if the extension exists.
447 * Added jitterbuffer support for chan_local. This allows you to use the
448 generic jitterbuffer on incoming calls going to Asterisk applications.
449 For example, this would allow you to use a jitterbuffer for an incoming
450 SIP call to Voicemail by putting a Local channel in the middle. This
451 feature is enabled by using the 'j' option in the Dial string to the Local
452 channel in conjunction with the existing 'n' option for local channels.
453 * A 'b' option has been added which causes chan_local to return the actual channel
454 that is behind it when queried. This is useful for transfer scenarios as the
455 actual channel will be transferred, not the Local channel.
457 Agent channel changes
458 ----------------------
459 * The ackcall and endcall options are now supplemented with options acceptdtmf
460 and enddtmf. These allow for the DTMF keypress to be configurable. The options
461 default to their old hard-coded values ('#' and '*' respectively) so this should
462 not break any existing agent installations.
464 DAHDI channel driver (chan_dahdi) Changes
465 ----------------------------------------
466 * SS7 support (via libss7 library)
467 * In India, some carriers transmit CID via dtmf. Some code has been added
468 that will handle some situations. The cidstart=polarity_IN choice has been added for
469 those carriers that transmit CID via dtmf after a polarity change.
470 * CID matching information is now shown when doing 'dialplan show'.
471 * Added dahdi show version CLI command.
472 * Added setvar support to chan_dahdi.conf channel entries.
473 * Added two new options: mwimonitor and mwimonitornotify. These options allow
474 you to enable MWI monitoring on FXO lines. When the MWI state changes,
475 the script specified in the mwimonitornotify option is executed. An internal
476 event indicating the new state of the mailbox is also generated, so that
477 the normal MWI facilities in Asterisk work as usual.
478 * Added signalling type 'auto', which attempts to use the same signalling type
479 for a channel as configured in DAHDI. This is primarily designed for analog
480 ports, but will also work for digital ports that are configured for FXS or FXO
481 signalling types. This mode is also the default now, so if your chan_dahdi.conf
482 does not specify signalling for a channel (which is unlikely as the sample
483 configuration file has always recommended specifying it for every channel) then
484 the 'auto' mode will be used for that channel if possible.
485 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
486 state for a channel; also ensured that the DNDState Manager event is
487 emitted no matter how the DND state is set or cleared.
491 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
492 configs/unistim.conf.sample for details. This new channel driver allows
493 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
494 * Added a new channel driver, chan_console, which uses portaudio as a cross
495 platform audio interface. It was written as a channel driver that would
496 work with Mac CoreAudio, but portaudio supports a number of other audio
497 interfaces, as well. Note that this channel driver requires v19 or higher
498 of portaudio; older versions have a different API.
502 * Added the ability to specify arguments to the Dial application when using
503 the DUNDi switch in the dialplan.
504 * Added the ability to set weights for responses dynamically. This can be
505 done using a global variable or a dialplan function. Using the SHELL()
506 function would allow you to have an external script set the weight for
508 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
509 functions will allow you to initiate a DUNDi query from the dialplan,
510 find out how many results there are, and access each one.
514 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
515 functions will allow you to initiate an ENUM lookup from the dialplan,
516 and Asterisk will cache the results. ENUMRESULT can be used to access
517 the results without doing multiple DNS queries.
521 * Added the ability to customize which sound files are used for some of the
522 prompts within the Voicemail application by changing them in voicemail.conf
523 * Added the ability for the "voicemail show users" CLI command to show users
524 configured by the dynamic realtime configuration method.
525 * MWI (Message Waiting Indication) handling has been significantly
526 restructured internally to Asterisk. It is now totally event based
527 instead of polling based. The voicemail application will notify other
528 modules that have subscribed to MWI events when something in the mailbox
530 This also means that if any other entity outside of Asterisk is changing
531 the contents of mailboxes, then the voicemail application still needs to
532 poll for changes. Examples of situations that would require this option
533 are web interfaces to voicemail or an email client in the case of using
534 IMAP storage. So, two new options have been added to voicemail.conf
535 to account for this: "pollmailboxes" and "pollfreq". See the sample
536 configuration file for details.
537 * Added "tw" language support
538 * Added support for storage of greetings using an IMAP server
539 * Added ability to customize forward, reverse, stop, and pause keys for message playback
540 * SMDI is now enabled in voicemail using the smdienable option.
541 * A "lockmode" option has been added to asterisk.conf to configure the file
542 locking method used for voicemail, and potentially other things in the
543 future. The default is the old behavior, lockfile. However, there is a
544 new method, "flock", that uses a different method for situations where the
545 lockfile will not work, such as on SMB/CIFS mounts.
546 * Added the ability to backup deleted messages, to ease recovery in the case
547 that a user accidentally deletes a message, and discovers that they need it.
548 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
549 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
550 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
551 voicemail boxes. The SMDI interface can also poll for MWI changes when some
552 outside entity is modifying the state of the mailbox (such as IMAP storage or
553 a web interface of some kind).
554 * Added the support for marking messages as "urgent." There are two methods to accomplish
555 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
556 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
557 the message as urgent after he has recorded a voicemail by following the voice instructions.
558 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
563 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
564 used across multiple queues.
565 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
566 setqueueentryvar options for each queue, see queues.conf.sample for details.
567 * Added keepstats option to queues.conf which will keep queue
568 statistics during a reload.
569 * setinterfacevar option in queues.conf also now sets a variable
570 called MEMBERNAME which contains the member's name.
571 * Added 'Strategy' field to manager event QueueParams which represents
572 the queue strategy in use.
573 * Added option to run macro when a queue member is connected to a caller,
574 see queues.conf.sample for details.
575 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
576 does not count paused queue members as unavailable.
577 * Added min-announce-frequency option to queues.conf which allows you to control the
578 minimum amount of time between queue announcements for use when the caller's queue
579 position changes frequently.
580 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
582 * Added ability for non-realtime queues to have realtime members
583 * Added the "linear" strategy to queues.
584 * Added the "wrandom" strategy to queues.
585 * Added new channel variable QUEUE_MIN_PENALTY
586 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
587 rules in queuerules.conf. See configs/queuerules.conf.sample for details
588 * Added a new parameter for member definition, called state_interface. This may be
589 used so that a member may be called via one interface but have a different interface's
590 device state reported.
591 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
592 specified by the periodic-announce option, then one will be chosen randomly when it is time
593 to play a periodic announcment
594 * New configuration options: announce-position now takes two more values in addition to "yes" and
595 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
596 announce-position-limit. By setting announce-position to "limit" callers will only have their
597 position announced if their position is less than what is specified by announce-position-limit.
598 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
599 will be told that their are more than announce-position-limit callers waiting.
600 * Two new queue log events have been added. An ADDMEMBER event will be logged
601 when a realtime queue member is added and a REMOVEMEMBER event will be logged
602 when a realtime queue member is removed. Since there is no calling channel associated
603 with these events, the string "REALTIME" is placed where the channel's unique id
608 * The 'o' option to provide an optimization has been removed and its functionality
609 has been enabled by default.
610 * When a conference is created, the UNIQUEID of the channel that caused it to be
611 created is stored. Then, every channel that joins the conference will have the
612 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
613 callers that come and go from long standing conferences.
614 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
615 except it does operations on a channel by name, instead of number in a conference.
616 This is a very useful feature in combination with the 'X' option to ChanSpy.
617 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
619 * Added new RealTime functionality to provide support for scheduled conferencing.
620 This includes optional messages to the caller if they attempt to join before
621 the schedule start time, or to allow the caller to join the conference early.
622 Also included is optional support for limiting the number of callers per
624 * Added the S() and L() options to the MeetMe application. These are pretty
625 much identical to the S() and L() options to Dial(). They let you set
626 timeouts for the conference, as well as have warning sounds played to
627 let the caller know how much time is left, and when it is running out.
628 * Added the ability to do "meetme concise" with the "meetme" CLI command.
629 This extends the concise capabilities of this CLI command to include
630 listing all conferences, instead of an addition to the other sub commands
631 for the "meetme" command.
632 * Added the ability to specify the music on hold class used to play into the
633 conference when there is only one member and the M option is used.
634 * Added MEETME_INFO dialplan function which provides a way to query
635 various properties of a Meetme conference.
637 Other Dialplan Application Changes
638 ----------------------------------
639 * Argument support for Gosub application
640 * From the to-do lists: straighten out the app timeout args:
641 Wait() app now really does 0.3 seconds- was truncating arg to an int.
642 WaitExten() same as Wait().
643 Congestion() - Now takes floating pt. argument.
644 Busy() - now takes floating pt. argument.
645 Read() - timeout now can be floating pt.
646 WaitForRing() now takes floating pt timeout arg.
647 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
648 * Added 's' option to Page application.
649 * Added 'E', 'V', and 'P' commands to ExternalIVR.
650 * Added 'o' and 'X' options to Chanspy.
651 * Added a new dialplan application, Bridge, which allows you to bridge the
652 calling channel to any other active channel on the system.
653 * Added the ability to specify a music on hold class to play instead of ringing
654 for the SLATrunk application.
655 * The Read application no longer exits the dialplan on error. Instead, it sets
656 READSTATUS to ERROR, which you can catch and handle separately.
657 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
658 of asking for verification of each name, one at a time.
659 * Privacy() no longer uses privacy.conf, as all options are specifyable as
660 direct options to the app.
661 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
663 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
664 * The ChannelRedirect application no longer exits the dialplan if the given channel
665 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
666 or NOCHANNEL if the given channel was not found.
667 * The silencethreshold setting that was previously configurable in multiple
668 applications is now settable globally via dsp.conf.
670 Music On Hold Changes
671 ---------------------
672 * A new option, "digit", has been added for music on hold classes in
673 musiconhold.conf. If this is set for a music on hold class, a caller
674 listening to music on hold can press this digit to switch to listening
675 to this music on hold class.
676 * Support for realtime music on hold has been added.
677 * In conjunction with the realtime music on hold, a general section has
678 been added to musiconhold.conf, its sole variable is cachertclasses. If this
679 is set, then music on hold classes found in realtime will be cached in memory.
683 * AEL upgraded to use the Gosub with Arguments instead
684 of Macro application, to hopefully reduce the problems
685 seen with the artificially low stack ceiling that
686 Macro bumps into. Macros can only call other Macros
687 to a depth of 7. Tests run using gosub, show depths
688 limited only by virtual memory. A small test demonstrated
689 recursive call depths of 100,000 without problems.
690 -- in addition to this, all apps that allowed a macro
691 to be called, as in Dial, queues, etc, are now allowing
692 a gosub call in similar fashion.
693 * AEL now generates LOCAL(argname) declarations when it
694 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
695 etc. That makes the arguments local in scope. The user
696 can define their own local variables in macros, now,
697 by saying "local myvar=someval;" or using Set() in this
698 fashion: Set(LOCAL(myvar)=someval); ("local" is now
700 * utils/conf2ael introduced. Will convert an extensions.conf
701 file into extensions.ael. Very crude and unfinished, but
702 will be improved as time goes by. Should be useful for a
703 first pass at conversion.
704 * aelparse will now read extensions.conf to see if a referenced
705 macro or context is there before issueing a warning.
706 * AEL parser sets a local channel variable ~~EXTEN~~, to
707 preserve the value of ${EXTEN} thru switch statements.
708 * New operator in $[...] expressions: the ~~ operator serves
709 as a concatenation operator. AT THE MOMENT, it is really only
710 necessary and useful in AEL, especially in if() expressions.
711 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
712 any enclosing double-quotes, and evaluate to the value of a
713 concatenated with the value of b. For example if a is set to
714 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
718 Call Features (res_features) Changes
719 ------------------------------------
720 * Added the parkedcalltransfers option to features.conf
721 * The built-in method for doing attended transfers has been updated to
722 include some new options that allow you to have the transferee sent
723 back to the person that did the transfer if the transfer is not successful.
724 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
725 in features.conf.sample.
726 * Added support for configuring named groups of custom call features in
727 features.conf. This means that features can be written a single time, and
728 then mapped into groups of features for different key mappings or easier
730 * Updated the ParkedCall application to allow you to not specify a parking
731 extension. If you don't specify a parking space to pick up, it will grab
732 the first one available.
733 * Added cli command 'features reload' to reload call features from features.conf
734 * Moved into core asterisk binary.
736 Language Support Changes
737 ------------------------
738 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
739 * Added support for the Hungarian language for saying numbers, dates, and times.
743 * Added SPEECH commands for speech recognition. A complete listing can be found
745 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
746 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
747 does not behave as expected; the native command needs to be used, instead.
751 * Added rotatestrategy option to logger.conf, along with two new options:
752 "timestamp" which will use the time to name the logger files instead of
753 sequence number; and "rotate", which rotates the names of the logfiles,
754 similar to the way syslog rotates files.
755 * Added exec_after_rotate option to logger.conf, which allows a system
756 command to be run after rotation. This is primarily useful with
757 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
758 and to ensure that the oldest log file gets deleted.
759 * Added realtime support for the queue log
763 * The cdr_manager module has a [mappings] feature, like cdr_custom,
764 to add fields to the manager event from the CDR variables.
765 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
766 backend database CDR table. Specifically, additional, non-standard
767 columns are supported, merely by setting the corresponding CDR variable in
768 your dialplan. In addition, you may alias any column to another name (for
769 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
770 simply "alias src => ANI" in the configuration file). Records may be
771 posted to more than one backend, simply by specifying multiple categories
772 in the configuration file. And finally, you may filter which CDRs get
773 posted to each backend, by specifying a filter (which the record must
774 match) for the particular category. Filters are additive (meaning all
775 rules must match to post that CDR).
776 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
777 module. Specifically, you may add additional columns into the table and
778 they will be set, if you set the corresponding CDR variable name. Also,
779 if you omit columns in your database table, they will be silently skipped
780 (but a record will still be inserted, based on what columns remain). Note
781 that the other two features from cdr_adaptive_odbc (alias and filter) are
782 not currently supported.
783 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
784 has been disabled using the NoCDR application.
786 Miscellaneous New Modules
787 -------------------------
788 * Added a new CDR module, cdr_sqlite3_custom.
789 * Added a new realtime configuration module, res_config_sqlite
790 * Added a new codec translation module, codec_resample, which re-samples
791 signed linear audio between 8 kHz and 16 kHz to help support wideband
793 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
794 based on configuration templates that use Asterisk dialplan function and
795 variable substitution. It should be possible to create phone profiles and
796 templates that work for the majority of phones provisioned over http. It
797 is currently only intended to provision a single user account per phone.
798 An example profile and set of templates for Polycom phones is provided.
799 NOTE: Polycom firmware is not included, but should be placed in
800 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
801 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
802 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
803 provided; there is a JACK() application, and a JACK_HOOK() function. Both
804 interfaces create an input and output JACK port. The application makes
805 these ports the endpoint of the call. The audio coming from the channel
806 goes out the output port and whatever comes back in on the input port is
807 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
808 audiohook on the channel. This lets you run the audio coming from a
809 channel through JACK, and whatever comes back in is what gets forwarded
810 on as the channel's audio. This is very useful for building custom
811 vocoders or doing recording or analysis of the channel's audio in another
813 * Added a new module, res_config_curl, which permits using a HTTP POST url
814 to retrieve, create, update, and delete realtime information from a remote
815 web server. Note that this module requires func_curl.so to be loaded for
816 backend functionality.
817 * Added a new module, res_config_ldap, which permits the use of an LDAP
818 server for realtime data access.
819 * Added support for writing and running your dialplan in lua using the pbx_lua
820 module. See configs/extensions.lua.sample for examples of how to do this.
824 * Ability to use libcap to set high ToS bits when non-root
825 on Linux. If configure is unable to find libcap then you
826 can use --with-cap to specify the path.
827 * Added maxfiles option to options section of asterisk.conf which allows you to specify
828 what Asterisk should set as the maximum number of open files when it loads.
829 * Added the jittertargetextra configuration option.
830 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
831 configuration files for the IP channel drivers. The new option is "cos".
832 This information is also documented in doc/qos.tex, or the IP Quality of Service
833 section of asterisk.pdf.
834 * When originating a call using AMI or pbx_spool that fails the reason for failure
835 will now be available in the failed extension using the REASON dialplan variable.
836 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
837 It allows you to configure a prefix for auto-monitor recordings.
838 * A new extension pattern matching algorithm, based on a trie, is introduced
839 here, that could noticeably speed up mid-sized to large dialplans.
840 It is NOT used by default, as duplicating the behaviour of the old pattern
841 matcher is still under development. A config file option, in extensions.conf,
842 in the [general] section, called "extenpatternmatchingnew", is by default
843 set to false; setting that to true will force the use of the new algorithm.
844 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
845 be used to switch the algorithms at run time.
846 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
847 specifying which socket to use to connect to the running Asterisk daemon
849 * Performance enhancements to the sched facility, which is used in
850 the channel drivers, etc. Added hashtabs and doubly-linked lists
851 to speed up deletion; start at the beginning or end of list to
853 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
854 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
855 Added regression tests to the tests/ dir, also.
856 * Added a refcount trace feature to astobj2 for those trying to balance
857 object creation, deletion; work, play; space and time. See the
858 notes in astobj2.h. Also, see utils/refcounter as well, as a
859 quick way to find unbalanced refcounts in what could be a sea
860 of objects that were balanced.
861 * Added logging to 'make update' command. See update.log
862 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
863 do not come from the remote party.
864 * Added the 'n' option to the SpeechBackground application to tell it to not
865 answer the channel if it has not already been answered.
866 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
867 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
869 * iLBC source code no longer included (see UPGRADE.txt for details)
870 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
871 deadlock is detected, a backtrace of the stack which led to the lock calls
872 will be output to the CLI.
873 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
874 the "core show locks" CLI command will give lock information output as well
875 as a backtrace of the stack which led to the lock calls.
876 * users.conf now sports an optional alternateexts property, which permits
877 allocation of additional extensions which will reach the specified user.
878 * A new option for the configure script, --enable-internal-poll, has been added
879 for use with systems which may have a buggy implementation of the poll system
880 call. If you notice odd behavior such as the CLI being unresponsive on remote
881 consoles, you may want to try using this option. This option is enabled by default
882 on Darwin systems since it is known that the Darwin poll() implementation has
projects / asterisk/asterisk.git / blob