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 * Scheduled meetme conferences may now have their end times extended by
26 * res_jabber: autoprune has been disabled by default, to avoid misconfiguration
27 that would end up being interpreted as a bug once Asterisk started removing
28 the contacts from a user list.
30 ------------------------------------------------------------------------------
31 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
32 ------------------------------------------------------------------------------
36 * The event infrastructure in Asterisk got another big update to help support
37 distributed events. It currently supports distributed device state and
38 distributed Voicemail MWI (Message Waiting Indication). A new module has
39 been merged, res_ais, which facilitates communicating events between servers.
40 It uses the SAForum AIS (Service Availability Forum Application Interface
41 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
42 a cluster of Asterisk servers, and to share events between them. For more
43 information on setting this up, see doc/distributed_devstate.txt.
47 * Added a new dialplan function, AST_CONFIG(), which allows you to access
48 variables from an Asterisk configuration file.
49 * The JACK_HOOK function now has a c() option to supply a custom client name.
50 * Added two new dialplan functions from libspeex for audio gain control and
51 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
52 rx directions of a channel from the dialplan.
53 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
54 based on other parameters. The default is still to search based on the
55 forwarding station ID. However, there are new options that allow you to search
56 based on the message desk terminal ID, or the message desk number.
57 * TIMEOUT() has been modified to be accurate down to the millisecond.
58 * ENUM*() functions now include the following new options:
59 - 'u' returns the full URI and does not strip off the URI-scheme.
60 - 's' triggers ISN specific rewriting
61 - 'i' looks for branches into an Infrastructure ENUM tree
62 - 'd' for a direct DNS lookup without any flipping of digits.
63 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
64 * CHANNEL() now has options for the maximum, minimum, and standard or normal
65 deviation of jitter, rtt, and loss for a call using chan_sip.
67 DAHDI channel driver (chan_dahdi) Changes
68 ----------------------------------------
69 * Channels can now be configured using named sections in chan_dahdi.conf, just
70 like other channel drivers, including the use of templates.
71 * The default for pridialplan has changed from 'national' to 'unknown'.
75 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
76 to something that matches the pattern a hint will be created using the contents
77 and variables evaluated.
78 * Dialplan matching has been extended to allow an extension to return to the
79 PBX core to wait for more digits. This is done by using the new dialplan
80 application called "Incomplete". This will permit a whole new level of
81 extension control, by giving the administrator more control over early
82 matches employing one of the short-circuit pattern match operators. Note
83 that custom applications can trigger this same behavior by returning the
84 special value AST_PBX_INCOMPLETE.
88 * Directory now permits both first and last names to be matched at the same
89 time. In addition, the number of digits to enter of the name can be set in
90 the arguments to Directory; previously, you could enter only 3, regardless
91 of how many names are in your company. For large companies, this should be
93 * Voicemail now permits a mailbox setting to wrap around from first to last
94 messages, if the "messagewrap" option is set to a true value.
95 * Voicemail now permits an external script to be run, for password validation.
96 The script should output "VALID" or "INVALID" on stdout, depending upon the
97 wish to validate or invalidate the password given. Arguments are:
98 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
100 * Dial has a new option: F(context^extension^pri), which permits a callee to
101 continue in the dialplan, at the specified label, if the caller hangs up.
102 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
103 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
104 * The Jack application now has a c() option to supply a custom client name.
105 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
106 like the pre-existing whisper mode, except that the spy can also talk to the
107 participant on the bridged channel as well.
108 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
109 to be spoken instead of the channel name or number. For more information on the
110 use of this option, issue the command "core show application ChanSpy" from the
112 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
113 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
114 words, if using the 'd' option, it is not possible to enter a number to append to
115 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
116 change to whisper mode, and pressing 6 will change to barge mode.
117 * ExternalIVR now takes several options that affect the way it performs, as
118 well as having several new commands. Please see doc/externalivr.txt for the
119 complete documentation.
120 * Added ability to communicate over a TCP socket instead of forking a child process for the
121 ExternalIVR application.
122 * ChanIsAvail has a new option, 'a', which will return all available channels instead
123 of just the first one if you give the function more then one channel to check.
124 * PrivacyManager now takes an option where you can specify a context where the
125 given number will be matched. This way you have more control over who is allowed
126 and it stops the people who blindly enter 10 digits.
127 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
128 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
129 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
130 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
131 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
132 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
133 * The Dial() application no longer copies the language used by the caller to the callee's
134 channel. If you desire for the caller's channel's language to be used for file playback
135 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
136 * SendImage() no longer hangs up the channel on error; instead, it sets the
137 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
138 'UNSUPPORTED'. This change makes SendImage() more consistent with other
140 * Park has a new option, 's', which silences the announcement of the parking space number.
144 * Added DNS manager support to registrations for peers referencing peer entries.
145 DNS manager runs in the background which allows DNS lookups to be run asynchronously
146 as well as periodically updating the IP address. These properties allow for
147 better performance as well as recovery in the event of an IP change.
148 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
149 load/reload of large numbers of peers/users by ~40x (for large lists of peers.
150 Initially, we saw 4x improvement in call setup/destruction, but at the time
151 of merging, this gain has disappeared; further research will be done to try
152 and restore this performance improvement. Astobj2 refcounting is now used
153 for users, peers, and dialogs. Users are encouraged to assist in regression
154 testing and problem reporting!
155 * Added ability to specify registration expiry time on a per registration basis in
157 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
159 * Added t38pt_usertpsource option. See sip.conf.sample for details.
160 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
161 * 'sip show peers' and 'sip show users' display their entries sorted in
162 alphabetical order, as opposed to the order they were in, in the config
164 * Videosupport now supports an additional option, "always", which always sets
165 up video RTP ports, even on clients that don't support it. This helps with
166 callfiles and certain transfers to ensure that if two video phones are
167 connected, they will always share video feeds.
171 * Existing DNS manager lookups extended to check for SRV records.
172 * IAX2 encryption support has been improved to support periodic key rotation
173 within a call for enhanced security. The option "keyrotate" has been
174 provided to disable this functionality to preserve backwards compatibility
175 with older versions of IAX2 that do not support key rotation.
179 * New CLI command, "config reload <file.conf>" which reloads any module that
180 references that particular configuration file. Also added "config list"
181 which shows which configuration files are in use.
182 * New CLI commands, "pri show version" and "ss7 show version" that will
183 display which version of libpri and libss7 are being used, respectively.
184 A new API call was added so trunk will now have to be compiled against
185 a versions of libpri and libss7 that have them or it will not know that
186 these libraries exist.
187 * The commands "core show globals", "core set global" and "core set chanvar" has
188 been deprecated in favor of the more semanticly correct "dialplan show globals",
189 "dialplan set chanvar" and "dialplan set global".
190 * New CLI command "dialplan show chanvar" to list all variables associated
191 with a given channel.
195 * Addresses managed by DNS manager now can check to see if there is a DNS
196 SRV record for a given domain and will use that hostname/port if present.
198 AMI - The manager (TCP/TLS/HTTP)
199 --------------------------------
200 * The Status command now takes an optional list of variables to display
201 along with channel status.
205 * res_odbc no longer has a limit of 1023 total possible unshared connections,
206 as some people were running into this limit. This limit has been increased
211 * The TRANSFER queue log entry now includes the the caller's original
212 position in the transferred-from queue.
213 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
214 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
215 as well as an explanation about timeout options in general
219 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
220 adaptive capabilities. What this means in practical terms is that if your
221 realtime table lacks critical fields, Asterisk will now emit warnings to
222 that effect. Also, some of the realtime drivers have the ability (if
223 configured) to automatically add those columns to the table with the
224 correct type and length.
228 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
229 the 'setvar' option to cause a given audio file to be played upon completion
230 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
231 Skinny channels only.
232 * You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt
233 for more information.
234 * Config file variables may now be appended to, by using the '+=' append
235 operator. This is most helpful when working with long SQL queries in
236 func_odbc.conf, as the queries no longer need to be specified on a single
238 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
239 which will add a second to the billsec when the ending
240 time is set, if the number in the microseconds field of the end time is
241 greater than the number of microseconds in the answer time. This allows
242 users to count the 'initiated' seconds in their billing records.
244 ------------------------------------------------------------------------------
245 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
246 ------------------------------------------------------------------------------
248 AMI - The manager (TCP/TLS/HTTP)
249 --------------------------------
250 * Manager has undergone a lot of changes, all of them documented
251 in doc/manager_1_1.txt
252 * Manager version has changed to 1.1
253 * Added a new action 'CoreShowChannels' to list currently defined channels
254 and some information about them.
255 * Added a new action 'SIPshowregistry' to list SIP registrations.
256 * Added TLS support for the manager interface and HTTP server
257 * Added the URI redirect option for the built-in HTTP server
258 * The output of CallerID in Manager events is now more consistent.
259 CallerIDNum is used for number and CallerIDName for name.
260 * Enable https support for builtin web server.
261 See configs/http.conf.sample for details.
262 * Added a new action, GetConfigJSON, which can return the contents of an
263 Asterisk configuration file in JSON format. This is intended to help
264 improve the performance of AJAX applications using the manager interface
266 * SIP and IAX manager events now use "ChannelType" in all cases where we
267 indicate channel driver. Previously, we used a mixture of "Channel"
268 and "ChannelDriver" headers.
269 * Added a "Bridge" action which allows you to bridge any two channels that
270 are currently active on the system.
271 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
272 the voicemail users setup.
273 * Added 'DBDel' and 'DBDelTree' manager commands.
274 * cdr_manager now reports events via the "cdr" level, separating it from
275 the very verbose "call" level.
276 * Manager users are now stored in memory. If you change the manager account
277 list (delete or add accounts) you need to reload manager.
278 * Added Masquerade manager event for when a masquerade happens between
280 * Added "manager reload" command for the CLI
281 * Lots of commands that only provided information are now allowed under the
282 Reporting privilege, instead of only under Call or System.
283 * The IAX* commands now require either System or Reporting privilege, to
284 mirror the privileges of the SIP* commands.
285 * Added ability to retrieve list of categories in a config file.
286 * Added ability to retrieve the content of a particular category.
287 * Added ability to empty a context.
288 * Created new action to create a new file.
289 * Updated delete action to allow deletion by line number with respect to category.
290 * Added new action insert to add new variable to category at specified line.
291 * Updated action newcat to allow new category to be inserted in file above another
293 * Added new event "JitterBufStats" in the IAX2 channel
294 * Originate now requires the Originate privilege and, if you want to call out
295 to a subshell, it requires the System privilege, as well. This was done to
296 enhance manager security.
297 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
298 * New command: Atxfer. See doc/manager_1_1.txt for more details or
299 manager show command Atxfer from the CLI
303 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
304 state in the dialplan, as well as creating custom device states that are
305 controllable from the dialplan.
306 * Extend CALLERID() function with "pres" and "ton" parameters to
307 fetch string representation of calling number presentation indicator
308 and numeric representation of type of calling number value.
309 * MailboxExists converted to dialplan function
310 * A new option to Dial() for telling IP phones not to count the call
311 as "missed" when dial times out and cancels.
312 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
313 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
314 held for any given channel. Also, locks are automatically freed when a
316 * Added HINT() dialplan function that allows retrieving hint information.
317 Hints are mappings between extensions and devices for the sake of
318 determining the state of an extension. This function can retrieve the list
319 of devices or the name associated with a hint.
320 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
322 * Added SYSINFO() dialplan function which allows retrieval of system information
323 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
324 the existence of a dialplan target.
325 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
326 upper and lower case, respectively.
327 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
328 ID for the call (not the Asterisk call ID or unique ID), provided that the
329 channel driver supports this. For SIP, you get the SIP call-ID for the
330 bridged channel which you can store in the CDR with a custom field.
334 * New CLI command "core show hint" (usage: core show hint <exten>)
335 * New CLI command "core show settings"
336 * Added 'core show channels count' CLI command.
337 * Added the ability to set the core debug and verbose values on a per-file basis.
338 * Added 'queue pause member' and 'queue unpause member' CLI commands
339 * Ability to set process limits ("ulimit") without restarting Asterisk
340 * Enhanced "agi debug" to print the channel name as a prefix to the debug
341 output to make debugging on busy systems much easier.
342 * New CLI commands "dialplan set extenpatternmatching true/false"
343 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
344 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
345 listed in the startup_commands section of cli.conf will get executed.
346 * Added a CLI command, "devstate change", which allows you to set custom device
347 states from the func_devstate module that provides the DEVICE_STATE() function
348 and handling of the "Custom:" devices.
349 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
350 sorted into the different possible callbacks, with the number of entries
351 currently scheduled for each. Gives you a feel for how busy the sip channel
353 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
354 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
355 (Done by lmadsen, junky and mvanbaak during the AstriDevCon)
359 * Improved NAT and STUN support.
360 chan_sip now can use port numbers in bindaddr, externip and externhost
361 options, as well as contact a STUN server to detect its external address
362 for the SIP socket. See sip.conf.sample, 'NAT' section.
363 * The default SIP useragent= identifier now includes the Asterisk version
364 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
365 If set, and the incoming request carries authentication info,
366 the username to match in the users list is taken from the Digest header
367 rather than from the From: field. This feature is considered experimental.
368 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
369 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
370 * The "localmask" setting was removed in version 1.2 and the reminder about it
371 being removed is now also removed.
372 * A new option "busylevel" for setting a level of calls where asterisk reports
373 a device as busy, to separate it from call-limit. This value is also added
374 to the SIP_PEER dialplan function.
375 * A new realtime family called "sipregs" is now supported to store SIP registration
376 data. If this family is defined, "sippeers" will be used for configuration and
377 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
378 registration data, as before.
379 * The SIPPEER function have new options for port address, call and pickup groups
380 * Added support for T.140 realtime text in SIP/RTP
381 * The "checkmwi" option has been removed from sip.conf, as it is no longer
382 required due to the restructuring of how MWI is handled. See the descriptions
383 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
384 for more information.
385 * Added rtpdest option to CHANNEL() dialplan function.
386 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
387 * SIP now adds a header to the CANCEL if the call was answered by another phone
388 in the same dial command, or if the new c option in dial() is used.
389 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
390 states it is not needed. For phones, however, that do require it the "registertrying" option
391 has been added so it can be enabled.
392 * A new option called "callcounter" (global/peer/user level) enables call counters needed
393 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
394 used to enable this functionality).
395 * New settings for timer T1 and timer B on a global level or per device. This makes it
396 possible to force timeout faster on non-responsive SIP servers. These settings are
397 considered advanced, so don't use them unless you have a problem.
398 * Added a dial string option to be able to set the To: header in an INVITE to any
400 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
401 the qualify frequency.
402 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
403 were not properly torn down due to network or endpoint failures during an established
405 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
406 configs/sip.conf.sample for more information on how it is used.
407 * Added a new configuration option "authfailureevents" that enables manager events when
408 a peer can't authenticate properly.
409 * Added DNS manager support to registrations for peers not referencing a peer entry.
413 * Added the trunkmaxsize configuration option to chan_iax2.
414 * Added the srvlookup option to iax.conf
415 * Added support for OSP. The token is set and retrieved through the CHANNEL()
418 XMPP Google Talk/Jingle changes
419 -------------------------------
420 * Added the bindaddr option to gtalk.conf.
424 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
425 * Proper codec support in chan_skinny.
426 * Added settings for IP and Ethernet QoS requests
430 * Added separate settings for media QoS in mgcp.conf
432 Console Channel Driver changes
433 ------------------------------
434 * Added experimental support for video send & receive to chan_oss.
435 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
438 Phone channel changes (chan_phone)
439 ----------------------------------
440 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
442 H.323 channel Changes
443 ---------------------
444 * H323 remote hold notification support added (by NOTIFY message
445 and/or H.450 supplementary service)
447 Local channel changes
448 ---------------------
449 * The device state functionality in the Local channel driver has been updated
450 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
451 to just UNKNOWN if the extension exists.
452 * Added jitterbuffer support for chan_local. This allows you to use the
453 generic jitterbuffer on incoming calls going to Asterisk applications.
454 For example, this would allow you to use a jitterbuffer for an incoming
455 SIP call to Voicemail by putting a Local channel in the middle. This
456 feature is enabled by using the 'j' option in the Dial string to the Local
457 channel in conjunction with the existing 'n' option for local channels.
458 * A 'b' option has been added which causes chan_local to return the actual channel
459 that is behind it when queried. This is useful for transfer scenarios as the
460 actual channel will be transferred, not the Local channel.
462 Agent channel changes
463 ----------------------
464 * The ackcall and endcall options are now supplemented with options acceptdtmf
465 and enddtmf. These allow for the DTMF keypress to be configurable. The options
466 default to their old hard-coded values ('#' and '*' respectively) so this should
467 not break any existing agent installations.
469 DAHDI channel driver (chan_dahdi) Changes
470 ----------------------------------------
471 * SS7 support (via libss7 library)
472 * In India, some carriers transmit CID via dtmf. Some code has been added
473 that will handle some situations. The cidstart=polarity_IN choice has been added for
474 those carriers that transmit CID via dtmf after a polarity change.
475 * CID matching information is now shown when doing 'dialplan show'.
476 * Added dahdi show version CLI command.
477 * Added setvar support to chan_dahdi.conf channel entries.
478 * Added two new options: mwimonitor and mwimonitornotify. These options allow
479 you to enable MWI monitoring on FXO lines. When the MWI state changes,
480 the script specified in the mwimonitornotify option is executed. An internal
481 event indicating the new state of the mailbox is also generated, so that
482 the normal MWI facilities in Asterisk work as usual.
483 * Added signalling type 'auto', which attempts to use the same signalling type
484 for a channel as configured in DAHDI. This is primarily designed for analog
485 ports, but will also work for digital ports that are configured for FXS or FXO
486 signalling types. This mode is also the default now, so if your chan_dahdi.conf
487 does not specify signalling for a channel (which is unlikely as the sample
488 configuration file has always recommended specifying it for every channel) then
489 the 'auto' mode will be used for that channel if possible.
490 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
491 state for a channel; also ensured that the DNDState Manager event is
492 emitted no matter how the DND state is set or cleared.
496 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
497 configs/unistim.conf.sample for details. This new channel driver allows
498 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
499 * Added a new channel driver, chan_console, which uses portaudio as a cross
500 platform audio interface. It was written as a channel driver that would
501 work with Mac CoreAudio, but portaudio supports a number of other audio
502 interfaces, as well. Note that this channel driver requires v19 or higher
503 of portaudio; older versions have a different API.
507 * Added the ability to specify arguments to the Dial application when using
508 the DUNDi switch in the dialplan.
509 * Added the ability to set weights for responses dynamically. This can be
510 done using a global variable or a dialplan function. Using the SHELL()
511 function would allow you to have an external script set the weight for
513 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
514 functions will allow you to initiate a DUNDi query from the dialplan,
515 find out how many results there are, and access each one.
519 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
520 functions will allow you to initiate an ENUM lookup from the dialplan,
521 and Asterisk will cache the results. ENUMRESULT can be used to access
522 the results without doing multiple DNS queries.
526 * Added the ability to customize which sound files are used for some of the
527 prompts within the Voicemail application by changing them in voicemail.conf
528 * Added the ability for the "voicemail show users" CLI command to show users
529 configured by the dynamic realtime configuration method.
530 * MWI (Message Waiting Indication) handling has been significantly
531 restructured internally to Asterisk. It is now totally event based
532 instead of polling based. The voicemail application will notify other
533 modules that have subscribed to MWI events when something in the mailbox
535 This also means that if any other entity outside of Asterisk is changing
536 the contents of mailboxes, then the voicemail application still needs to
537 poll for changes. Examples of situations that would require this option
538 are web interfaces to voicemail or an email client in the case of using
539 IMAP storage. So, two new options have been added to voicemail.conf
540 to account for this: "pollmailboxes" and "pollfreq". See the sample
541 configuration file for details.
542 * Added "tw" language support
543 * Added support for storage of greetings using an IMAP server
544 * Added ability to customize forward, reverse, stop, and pause keys for message playback
545 * SMDI is now enabled in voicemail using the smdienable option.
546 * A "lockmode" option has been added to asterisk.conf to configure the file
547 locking method used for voicemail, and potentially other things in the
548 future. The default is the old behavior, lockfile. However, there is a
549 new method, "flock", that uses a different method for situations where the
550 lockfile will not work, such as on SMB/CIFS mounts.
551 * Added the ability to backup deleted messages, to ease recovery in the case
552 that a user accidentally deletes a message, and discovers that they need it.
553 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
554 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
555 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
556 voicemail boxes. The SMDI interface can also poll for MWI changes when some
557 outside entity is modifying the state of the mailbox (such as IMAP storage or
558 a web interface of some kind).
559 * Added the support for marking messages as "urgent." There are two methods to accomplish
560 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
561 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
562 the message as urgent after he has recorded a voicemail by following the voice instructions.
563 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
568 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
569 used across multiple queues.
570 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
571 setqueueentryvar options for each queue, see queues.conf.sample for details.
572 * Added keepstats option to queues.conf which will keep queue
573 statistics during a reload.
574 * setinterfacevar option in queues.conf also now sets a variable
575 called MEMBERNAME which contains the member's name.
576 * Added 'Strategy' field to manager event QueueParams which represents
577 the queue strategy in use.
578 * Added option to run macro when a queue member is connected to a caller,
579 see queues.conf.sample for details.
580 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
581 does not count paused queue members as unavailable.
582 * Added min-announce-frequency option to queues.conf which allows you to control the
583 minimum amount of time between queue announcements for use when the caller's queue
584 position changes frequently.
585 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
587 * Added ability for non-realtime queues to have realtime members
588 * Added the "linear" strategy to queues.
589 * Added the "wrandom" strategy to queues.
590 * Added new channel variable QUEUE_MIN_PENALTY
591 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
592 rules in queuerules.conf. See configs/queuerules.conf.sample for details
593 * Added a new parameter for member definition, called state_interface. This may be
594 used so that a member may be called via one interface but have a different interface's
595 device state reported.
596 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
597 specified by the periodic-announce option, then one will be chosen randomly when it is time
598 to play a periodic announcment
599 * New configuration options: announce-position now takes two more values in addition to "yes" and
600 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
601 announce-position-limit. By setting announce-position to "limit" callers will only have their
602 position announced if their position is less than what is specified by announce-position-limit.
603 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
604 will be told that their are more than announce-position-limit callers waiting.
605 * Two new queue log events have been added. An ADDMEMBER event will be logged
606 when a realtime queue member is added and a REMOVEMEMBER event will be logged
607 when a realtime queue member is removed. Since there is no calling channel associated
608 with these events, the string "REALTIME" is placed where the channel's unique id
613 * The 'o' option to provide an optimization has been removed and its functionality
614 has been enabled by default.
615 * When a conference is created, the UNIQUEID of the channel that caused it to be
616 created is stored. Then, every channel that joins the conference will have the
617 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
618 callers that come and go from long standing conferences.
619 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
620 except it does operations on a channel by name, instead of number in a conference.
621 This is a very useful feature in combination with the 'X' option to ChanSpy.
622 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
624 * Added new RealTime functionality to provide support for scheduled conferencing.
625 This includes optional messages to the caller if they attempt to join before
626 the schedule start time, or to allow the caller to join the conference early.
627 Also included is optional support for limiting the number of callers per
629 * Added the S() and L() options to the MeetMe application. These are pretty
630 much identical to the S() and L() options to Dial(). They let you set
631 timeouts for the conference, as well as have warning sounds played to
632 let the caller know how much time is left, and when it is running out.
633 * Added the ability to do "meetme concise" with the "meetme" CLI command.
634 This extends the concise capabilities of this CLI command to include
635 listing all conferences, instead of an addition to the other sub commands
636 for the "meetme" command.
637 * Added the ability to specify the music on hold class used to play into the
638 conference when there is only one member and the M option is used.
639 * Added MEETME_INFO dialplan function which provides a way to query
640 various properties of a Meetme conference.
642 Other Dialplan Application Changes
643 ----------------------------------
644 * Argument support for Gosub application
645 * From the to-do lists: straighten out the app timeout args:
646 Wait() app now really does 0.3 seconds- was truncating arg to an int.
647 WaitExten() same as Wait().
648 Congestion() - Now takes floating pt. argument.
649 Busy() - now takes floating pt. argument.
650 Read() - timeout now can be floating pt.
651 WaitForRing() now takes floating pt timeout arg.
652 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
653 * Added 's' option to Page application.
654 * Added 'E', 'V', and 'P' commands to ExternalIVR.
655 * Added 'o' and 'X' options to Chanspy.
656 * Added a new dialplan application, Bridge, which allows you to bridge the
657 calling channel to any other active channel on the system.
658 * Added the ability to specify a music on hold class to play instead of ringing
659 for the SLATrunk application.
660 * The Read application no longer exits the dialplan on error. Instead, it sets
661 READSTATUS to ERROR, which you can catch and handle separately.
662 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
663 of asking for verification of each name, one at a time.
664 * Privacy() no longer uses privacy.conf, as all options are specifyable as
665 direct options to the app.
666 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
668 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
669 * The ChannelRedirect application no longer exits the dialplan if the given channel
670 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
671 or NOCHANNEL if the given channel was not found.
672 * The silencethreshold setting that was previously configurable in multiple
673 applications is now settable globally via dsp.conf.
675 Music On Hold Changes
676 ---------------------
677 * A new option, "digit", has been added for music on hold classes in
678 musiconhold.conf. If this is set for a music on hold class, a caller
679 listening to music on hold can press this digit to switch to listening
680 to this music on hold class.
681 * Support for realtime music on hold has been added.
682 * In conjunction with the realtime music on hold, a general section has
683 been added to musiconhold.conf, its sole variable is cachertclasses. If this
684 is set, then music on hold classes found in realtime will be cached in memory.
688 * AEL upgraded to use the Gosub with Arguments instead
689 of Macro application, to hopefully reduce the problems
690 seen with the artificially low stack ceiling that
691 Macro bumps into. Macros can only call other Macros
692 to a depth of 7. Tests run using gosub, show depths
693 limited only by virtual memory. A small test demonstrated
694 recursive call depths of 100,000 without problems.
695 -- in addition to this, all apps that allowed a macro
696 to be called, as in Dial, queues, etc, are now allowing
697 a gosub call in similar fashion.
698 * AEL now generates LOCAL(argname) declarations when it
699 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
700 etc. That makes the arguments local in scope. The user
701 can define their own local variables in macros, now,
702 by saying "local myvar=someval;" or using Set() in this
703 fashion: Set(LOCAL(myvar)=someval); ("local" is now
705 * utils/conf2ael introduced. Will convert an extensions.conf
706 file into extensions.ael. Very crude and unfinished, but
707 will be improved as time goes by. Should be useful for a
708 first pass at conversion.
709 * aelparse will now read extensions.conf to see if a referenced
710 macro or context is there before issueing a warning.
711 * AEL parser sets a local channel variable ~~EXTEN~~, to
712 preserve the value of ${EXTEN} thru switch statements.
713 * New operator in $[...] expressions: the ~~ operator serves
714 as a concatenation operator. AT THE MOMENT, it is really only
715 necessary and useful in AEL, especially in if() expressions.
716 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
717 any enclosing double-quotes, and evaluate to the value of a
718 concatenated with the value of b. For example if a is set to
719 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
723 Call Features (res_features) Changes
724 ------------------------------------
725 * Added the parkedcalltransfers option to features.conf
726 * The built-in method for doing attended transfers has been updated to
727 include some new options that allow you to have the transferee sent
728 back to the person that did the transfer if the transfer is not successful.
729 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
730 in features.conf.sample.
731 * Added support for configuring named groups of custom call features in
732 features.conf. This means that features can be written a single time, and
733 then mapped into groups of features for different key mappings or easier
735 * Updated the ParkedCall application to allow you to not specify a parking
736 extension. If you don't specify a parking space to pick up, it will grab
737 the first one available.
738 * Added cli command 'features reload' to reload call features from features.conf
739 * Moved into core asterisk binary.
741 Language Support Changes
742 ------------------------
743 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
744 * Added support for the Hungarian language for saying numbers, dates, and times.
748 * Added SPEECH commands for speech recognition. A complete listing can be found
750 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
751 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
752 does not behave as expected; the native command needs to be used, instead.
756 * Added rotatestrategy option to logger.conf, along with two new options:
757 "timestamp" which will use the time to name the logger files instead of
758 sequence number; and "rotate", which rotates the names of the logfiles,
759 similar to the way syslog rotates files.
760 * Added exec_after_rotate option to logger.conf, which allows a system
761 command to be run after rotation. This is primarily useful with
762 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
763 and to ensure that the oldest log file gets deleted.
764 * Added realtime support for the queue log
768 * The cdr_manager module has a [mappings] feature, like cdr_custom,
769 to add fields to the manager event from the CDR variables.
770 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
771 backend database CDR table. Specifically, additional, non-standard
772 columns are supported, merely by setting the corresponding CDR variable in
773 your dialplan. In addition, you may alias any column to another name (for
774 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
775 simply "alias src => ANI" in the configuration file). Records may be
776 posted to more than one backend, simply by specifying multiple categories
777 in the configuration file. And finally, you may filter which CDRs get
778 posted to each backend, by specifying a filter (which the record must
779 match) for the particular category. Filters are additive (meaning all
780 rules must match to post that CDR).
781 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
782 module. Specifically, you may add additional columns into the table and
783 they will be set, if you set the corresponding CDR variable name. Also,
784 if you omit columns in your database table, they will be silently skipped
785 (but a record will still be inserted, based on what columns remain). Note
786 that the other two features from cdr_adaptive_odbc (alias and filter) are
787 not currently supported.
788 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
789 has been disabled using the NoCDR application.
791 Miscellaneous New Modules
792 -------------------------
793 * Added a new CDR module, cdr_sqlite3_custom.
794 * Added a new realtime configuration module, res_config_sqlite
795 * Added a new codec translation module, codec_resample, which re-samples
796 signed linear audio between 8 kHz and 16 kHz to help support wideband
798 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
799 based on configuration templates that use Asterisk dialplan function and
800 variable substitution. It should be possible to create phone profiles and
801 templates that work for the majority of phones provisioned over http. It
802 is currently only intended to provision a single user account per phone.
803 An example profile and set of templates for Polycom phones is provided.
804 NOTE: Polycom firmware is not included, but should be placed in
805 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
806 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
807 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
808 provided; there is a JACK() application, and a JACK_HOOK() function. Both
809 interfaces create an input and output JACK port. The application makes
810 these ports the endpoint of the call. The audio coming from the channel
811 goes out the output port and whatever comes back in on the input port is
812 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
813 audiohook on the channel. This lets you run the audio coming from a
814 channel through JACK, and whatever comes back in is what gets forwarded
815 on as the channel's audio. This is very useful for building custom
816 vocoders or doing recording or analysis of the channel's audio in another
818 * Added a new module, res_config_curl, which permits using a HTTP POST url
819 to retrieve, create, update, and delete realtime information from a remote
820 web server. Note that this module requires func_curl.so to be loaded for
821 backend functionality.
822 * Added a new module, res_config_ldap, which permits the use of an LDAP
823 server for realtime data access.
824 * Added support for writing and running your dialplan in lua using the pbx_lua
825 module. See configs/extensions.lua.sample for examples of how to do this.
829 * Ability to use libcap to set high ToS bits when non-root
830 on Linux. If configure is unable to find libcap then you
831 can use --with-cap to specify the path.
832 * Added maxfiles option to options section of asterisk.conf which allows you to specify
833 what Asterisk should set as the maximum number of open files when it loads.
834 * Added the jittertargetextra configuration option.
835 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
836 configuration files for the IP channel drivers. The new option is "cos".
837 This information is also documented in doc/qos.tex, or the IP Quality of Service
838 section of asterisk.pdf.
839 * When originating a call using AMI or pbx_spool that fails the reason for failure
840 will now be available in the failed extension using the REASON dialplan variable.
841 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
842 It allows you to configure a prefix for auto-monitor recordings.
843 * A new extension pattern matching algorithm, based on a trie, is introduced
844 here, that could noticeably speed up mid-sized to large dialplans.
845 It is NOT used by default, as duplicating the behaviour of the old pattern
846 matcher is still under development. A config file option, in extensions.conf,
847 in the [general] section, called "extenpatternmatchingnew", is by default
848 set to false; setting that to true will force the use of the new algorithm.
849 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
850 be used to switch the algorithms at run time.
851 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
852 specifying which socket to use to connect to the running Asterisk daemon
854 * Performance enhancements to the sched facility, which is used in
855 the channel drivers, etc. Added hashtabs and doubly-linked lists
856 to speed up deletion; start at the beginning or end of list to
858 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
859 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
860 Added regression tests to the tests/ dir, also.
861 * Added a refcount trace feature to astobj2 for those trying to balance
862 object creation, deletion; work, play; space and time. See the
863 notes in astobj2.h. Also, see utils/refcounter as well, as a
864 quick way to find unbalanced refcounts in what could be a sea
865 of objects that were balanced.
866 * Added logging to 'make update' command. See update.log
867 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
868 do not come from the remote party.
869 * Added the 'n' option to the SpeechBackground application to tell it to not
870 answer the channel if it has not already been answered.
871 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
872 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
874 * iLBC source code no longer included (see UPGRADE.txt for details)
875 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
876 deadlock is detected, a backtrace of the stack which led to the lock calls
877 will be output to the CLI.
878 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
879 the "core show locks" CLI command will give lock information output as well
880 as a backtrace of the stack which led to the lock calls.
881 * users.conf now sports an optional alternateexts property, which permits
882 allocation of additional extensions which will reach the specified user.
883 * A new option for the configure script, --enable-internal-poll, has been added
884 for use with systems which may have a buggy implementation of the poll system
885 call. If you notice odd behavior such as the CLI being unresponsive on remote
886 consoles, you may want to try using this option. This option is enabled by default
887 on Darwin systems since it is known that the Darwin poll() implementation has
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