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