1 ------------------------------------------------------------------------------
2 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
3 ------------------------------------------------------------------------------
7 * The event infrastructure in Asterisk got another big update to help support
8 distributed events. It currently supports distributed device state and
9 distributed Voicemail MWI (Message Waiting Indication). A new module has
10 been merged, res_ais, which facilitates communicating events between servers.
11 It uses the SAForum AIS (Service Availability Forum Application Interface
12 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
13 a cluster of Asterisk servers, and to share events between them. For more
14 information on setting this up, see doc/distributed_devstate.txt.
18 * Added a new dialplan function, AST_CONFIG(), which allows you to access
19 variables from an Asterisk configuration file.
20 * The JACK_HOOK function now has a c() option to supply a custom client name.
21 * Added two new dialplan functions from libspeex for audio gain control and
22 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
23 rx directions of a channel from the dialplan.
24 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
25 based on other parameters. The default is still to search based on the
26 forwarding station ID. However, there are new options that allow you to search
27 based on the message desk terminal ID, or the message desk number.
28 * TIMEOUT() has been modified to be accurate down to the millisecond.
29 * ENUM*() functions now include the following new options:
30 - 'u' returns the full URI and does not strip off the URI-scheme.
31 - 's' triggers ISN specific rewriting
32 - 'i' looks for branches into an Infrastructure ENUM tree
33 - 'd' for a direct DNS lookup without any flipping of digits.
34 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
35 * CHANNEL() now has options for the maximum, minimum, and standard or normal
36 deviation of jitter, rtt, and loss for a call using chan_sip.
38 DAHDI channel driver (chan_dahdi) Changes
39 ----------------------------------------
40 * Channels can now be configured using named sections in chan_dahdi.conf, just
41 like other channel drivers, including the use of templates.
42 * The default for pridialplan has changed from 'national' to 'unknown'.
46 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
47 to something that matches the pattern a hint will be created using the contents
48 and variables evaluated.
49 * Dialplan matching has been extended to allow an extension to return to the
50 PBX core to wait for more digits. This is done by using the new dialplan
51 application called "Incomplete". This will permit a whole new level of
52 extension control, by giving the administrator more control over early
53 matches employing one of the short-circuit pattern match operators. Note
54 that custom applications can trigger this same behavior by returning the
55 special value AST_PBX_INCOMPLETE.
59 * Directory now permits both first and last names to be matched at the same
60 time. In addition, the number of digits to enter of the name can be set in
61 the arguments to Directory; previously, you could enter only 3, regardless
62 of how many names are in your company. For large companies, this should be
64 * Voicemail now permits a mailbox setting to wrap around from first to last
65 messages, if the "messagewrap" option is set to a true value.
66 * Voicemail now permits an external script to be run, for password validation.
67 The script should output "VALID" or "INVALID" on stdout, depending upon the
68 wish to validate or invalidate the password given. Arguments are:
69 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
71 * Dial has a new option: F(context^extension^pri), which permits a callee to
72 continue in the dialplan, at the specified label, if the caller hangs up.
73 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
74 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
75 * The Jack application now has a c() option to supply a custom client name.
76 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
77 like the pre-existing whisper mode, except that the spy can also talk to the
78 participant on the bridged channel as well.
79 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
80 to be spoken instead of the channel name or number. For more information on the
81 use of this option, issue the command "core show application ChanSpy" from the
83 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
84 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
85 words, if using the 'd' option, it is not possible to enter a number to append to
86 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
87 change to whisper mode, and pressing 6 will change to barge mode.
88 * ExternalIVR now takes several options that affect the way it performs, as
89 well as having several new commands. Please see doc/externalivr.txt for the
90 complete documentation.
91 * ChanIsAvail has a new option, 'a', which will return all available channels instead
92 of just the first one if you give the function more then one channel to check.
93 * PrivacyManager now takes an option where you can specify a context where the
94 given number will be matched. This way you have more control over who is allowed
95 and it stops the people who blindly enter 10 digits.
96 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
97 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
98 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
99 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
100 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
101 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
102 * The Dial() application no longer copies the language used by the caller to the callee's
103 channel. If you desire for the caller's channel's language to be used for file playback
104 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
105 * SendImage() no longer hangs up the channel on error; instead, it sets the
106 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
107 'UNSUPPORTED'. This change makes SendImage() more consistent with other
112 * Added DNS manager support to registrations for peers referencing peer entries.
113 DNS manager runs in the background which allows DNS lookups to be run asynchronously
114 as well as periodically updating the IP address. These properties allow for
115 better performance as well as recovery in the event of an IP change.
116 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
117 load/reload of large numbers of peers/users by ~40x (for large lists of peers.
118 Initially, we saw 4x improvement in call setup/destruction, but at the time
119 of merging, this gain has disappeared; further research will be done to try
120 and restore this performance improvement. Astobj2 refcounting is now used
121 for users, peers, and dialogs. Users are encouraged to assist in regression
122 testing and problem reporting!
123 * Added ability to specify registration expiry time on a per registration basis in
125 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
127 * Added t38pt_usertpsource option. See sip.conf.sample for details.
128 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
129 * 'sip show peers' and 'sip show users' display their entries sorted in
130 alphabetical order, as opposed to the order they were in, in the config
132 * Videosupport now supports an additional option, "always", which always sets
133 up video RTP ports, even on clients that don't support it. This helps with
134 callfiles and certain transfers to ensure that if two video phones are
135 connected, they will always share video feeds.
139 * Existing DNS manager lookups extended to check for SRV records.
143 * New CLI command, "config reload <file.conf>" which reloads any module that
144 references that particular configuration file. Also added "config list"
145 which shows which configuration files are in use.
146 * New CLI commands, "pri show version" and "ss7 show version" that will
147 display which version of libpri and libss7 are being used, respectively.
148 A new API call was added so trunk will now have to be compiled against
149 a versions of libpri and libss7 that have them or it will not know that
150 these libraries exist.
151 * The commands "core show globals", "core set global" and "core set chanvar" has
152 been deprecated in favor of the more semanticly correct "dialplan show globals",
153 "dialplan set chanvar" and "dialplan set global".
154 * New CLI command "dialplan show chanvar" to list all variables associated
155 with a given channel.
159 * Addresses managed by DNS manager now can check to see if there is a DNS
160 SRV record for a given domain and will use that hostname/port if present.
162 AMI - The manager (TCP/TLS/HTTP)
163 --------------------------------
164 * The Status command now takes an optional list of variables to display
165 along with channel status.
169 * res_odbc no longer has a limit of 1023 total possible unshared connections,
170 as some people were running into this limit. This limit has been increased
175 * The TRANSFER queue log entry now includes the the caller's original
176 position in the transferred-from queue.
177 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
178 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
179 as well as an explanation about timeout options in general
183 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
184 the 'setvar' option to cause a given audio file to be played upon completion
185 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
186 Skinny channels only.
188 ------------------------------------------------------------------------------
189 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
190 ------------------------------------------------------------------------------
192 AMI - The manager (TCP/TLS/HTTP)
193 --------------------------------
194 * Manager has undergone a lot of changes, all of them documented
195 in doc/manager_1_1.txt
196 * Manager version has changed to 1.1
197 * Added a new action 'CoreShowChannels' to list currently defined channels
198 and some information about them.
199 * Added a new action 'SIPshowregistry' to list SIP registrations.
200 * Added TLS support for the manager interface and HTTP server
201 * Added the URI redirect option for the built-in HTTP server
202 * The output of CallerID in Manager events is now more consistent.
203 CallerIDNum is used for number and CallerIDName for name.
204 * Enable https support for builtin web server.
205 See configs/http.conf.sample for details.
206 * Added a new action, GetConfigJSON, which can return the contents of an
207 Asterisk configuration file in JSON format. This is intended to help
208 improve the performance of AJAX applications using the manager interface
210 * SIP and IAX manager events now use "ChannelType" in all cases where we
211 indicate channel driver. Previously, we used a mixture of "Channel"
212 and "ChannelDriver" headers.
213 * Added a "Bridge" action which allows you to bridge any two channels that
214 are currently active on the system.
215 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
216 the voicemail users setup.
217 * Added 'DBDel' and 'DBDelTree' manager commands.
218 * cdr_manager now reports events via the "cdr" level, separating it from
219 the very verbose "call" level.
220 * Manager users are now stored in memory. If you change the manager account
221 list (delete or add accounts) you need to reload manager.
222 * Added Masquerade manager event for when a masquerade happens between
224 * Added "manager reload" command for the CLI
225 * Lots of commands that only provided information are now allowed under the
226 Reporting privilege, instead of only under Call or System.
227 * The IAX* commands now require either System or Reporting privilege, to
228 mirror the privileges of the SIP* commands.
229 * Added ability to retrieve list of categories in a config file.
230 * Added ability to retrieve the content of a particular category.
231 * Added ability to empty a context.
232 * Created new action to create a new file.
233 * Updated delete action to allow deletion by line number with respect to category.
234 * Added new action insert to add new variable to category at specified line.
235 * Updated action newcat to allow new category to be inserted in file above another
237 * Added new event "JitterBufStats" in the IAX2 channel
238 * Originate now requires the Originate privilege and, if you want to call out
239 to a subshell, it requires the System privilege, as well. This was done to
240 enhance manager security.
241 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
242 * New command: Atxfer. See doc/manager_1_1.txt for more details or
243 manager show command Atxfer from the CLI
247 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
248 state in the dialplan, as well as creating custom device states that are
249 controllable from the dialplan.
250 * Extend CALLERID() function with "pres" and "ton" parameters to
251 fetch string representation of calling number presentation indicator
252 and numeric representation of type of calling number value.
253 * MailboxExists converted to dialplan function
254 * A new option to Dial() for telling IP phones not to count the call
255 as "missed" when dial times out and cancels.
256 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
257 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
258 held for any given channel. Also, locks are automatically freed when a
260 * Added HINT() dialplan function that allows retrieving hint information.
261 Hints are mappings between extensions and devices for the sake of
262 determining the state of an extension. This function can retrieve the list
263 of devices or the name associated with a hint.
264 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
266 * Added SYSINFO() dialplan function which allows retrieval of system information
267 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
268 the existence of a dialplan target.
269 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
270 upper and lower case, respectively.
271 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
272 ID for the call (not the Asterisk call ID or unique ID), provided that the
273 channel driver supports this. For SIP, you get the SIP call-ID for the
274 bridged channel which you can store in the CDR with a custom field.
278 * New CLI command "core show hint" (usage: core show hint <exten>)
279 * New CLI command "core show settings"
280 * Added 'core show channels count' CLI command.
281 * Added the ability to set the core debug and verbose values on a per-file basis.
282 * Added 'queue pause member' and 'queue unpause member' CLI commands
283 * Ability to set process limits ("ulimit") without restarting Asterisk
284 * Enhanced "agi debug" to print the channel name as a prefix to the debug
285 output to make debugging on busy systems much easier.
286 * New CLI commands "dialplan set extenpatternmatching true/false"
287 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
288 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
289 listed in the startup_commands section of cli.conf will get executed.
290 * Added a CLI command, "devstate change", which allows you to set custom device
291 states from the func_devstate module that provides the DEVICE_STATE() function
292 and handling of the "Custom:" devices.
293 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
294 sorted into the different possible callbacks, with the number of entries
295 currently scheduled for each. Gives you a feel for how busy the sip channel
300 * Improved NAT and STUN support.
301 chan_sip now can use port numbers in bindaddr, externip and externhost
302 options, as well as contact a STUN server to detect its external address
303 for the SIP socket. See sip.conf.sample, 'NAT' section.
304 * The default SIP useragent= identifier now includes the Asterisk version
305 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
306 If set, and the incoming request carries authentication info,
307 the username to match in the users list is taken from the Digest header
308 rather than from the From: field. This feature is considered experimental.
309 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
310 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
311 * The "localmask" setting was removed in version 1.2 and the reminder about it
312 being removed is now also removed.
313 * A new option "busylevel" for setting a level of calls where asterisk reports
314 a device as busy, to separate it from call-limit. This value is also added
315 to the SIP_PEER dialplan function.
316 * A new realtime family called "sipregs" is now supported to store SIP registration
317 data. If this family is defined, "sippeers" will be used for configuration and
318 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
319 registration data, as before.
320 * The SIPPEER function have new options for port address, call and pickup groups
321 * Added support for T.140 realtime text in SIP/RTP
322 * The "checkmwi" option has been removed from sip.conf, as it is no longer
323 required due to the restructuring of how MWI is handled. See the descriptions
324 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
325 for more information.
326 * Added rtpdest option to CHANNEL() dialplan function.
327 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
328 * SIP now adds a header to the CANCEL if the call was answered by another phone
329 in the same dial command, or if the new c option in dial() is used.
330 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
331 states it is not needed. For phones, however, that do require it the "registertrying" option
332 has been added so it can be enabled.
333 * A new option called "callcounter" (global/peer/user level) enables call counters needed
334 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
335 used to enable this functionality).
336 * New settings for timer T1 and timer B on a global level or per device. This makes it
337 possible to force timeout faster on non-responsive SIP servers. These settings are
338 considered advanced, so don't use them unless you have a problem.
339 * Added a dial string option to be able to set the To: header in an INVITE to any
341 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
342 the qualify frequency.
343 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
344 were not properly torn down due to network or endpoint failures during an established
346 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
347 configs/sip.conf.sample for more information on how it is used.
348 * Added a new configuration option "authfailureevents" that enables manager events when
349 a peer can't authenticate properly.
350 * Added DNS manager support to registrations for peers not referencing a peer entry.
354 * Added the trunkmaxsize configuration option to chan_iax2.
355 * Added the srvlookup option to iax.conf
356 * Added support for OSP. The token is set and retrieved through the CHANNEL()
359 XMPP Google Talk/Jingle changes
360 -------------------------------
361 * Added the bindaddr option to gtalk.conf.
365 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
366 * Proper codec support in chan_skinny.
367 * Added settings for IP and Ethernet QoS requests
371 * Added separate settings for media QoS in mgcp.conf
373 Console Channel Driver changes
374 ------------------------------
375 * Added experimental support for video send & receive to chan_oss.
376 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
379 Phone channel changes (chan_phone)
380 ----------------------------------
381 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
383 H.323 channel Changes
384 ---------------------
385 * H323 remote hold notification support added (by NOTIFY message
386 and/or H.450 supplementary service)
388 Local channel changes
389 ---------------------
390 * The device state functionality in the Local channel driver has been updated
391 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
392 to just UNKNOWN if the extension exists.
393 * Added jitterbuffer support for chan_local. This allows you to use the
394 generic jitterbuffer on incoming calls going to Asterisk applications.
395 For example, this would allow you to use a jitterbuffer for an incoming
396 SIP call to Voicemail by putting a Local channel in the middle. This
397 feature is enabled by using the 'j' option in the Dial string to the Local
398 channel in conjunction with the existing 'n' option for local channels.
399 * A 'b' option has been added which causes chan_local to return the actual channel
400 that is behind it when queried. This is useful for transfer scenarios as the
401 actual channel will be transferred, not the Local channel.
403 Agent channel changes
404 ----------------------
405 * The ackcall and endcall options are now supplemented with options acceptdtmf
406 and enddtmf. These allow for the DTMF keypress to be configurable. The options
407 default to their old hard-coded values ('#' and '*' respectively) so this should
408 not break any existing agent installations.
410 DAHDI channel driver (chan_dahdi) Changes
411 ----------------------------------------
412 * SS7 support (via libss7 library)
413 * In India, some carriers transmit CID via dtmf. Some code has been added
414 that will handle some situations. The cidstart=polarity_IN choice has been added for
415 those carriers that transmit CID via dtmf after a polarity change.
416 * CID matching information is now shown when doing 'dialplan show'.
417 * Added dahdi show version CLI command.
418 * Added setvar support to chan_dahdi.conf channel entries.
419 * Added two new options: mwimonitor and mwimonitornotify. These options allow
420 you to enable MWI monitoring on FXO lines. When the MWI state changes,
421 the script specified in the mwimonitornotify option is executed. An internal
422 event indicating the new state of the mailbox is also generated, so that
423 the normal MWI facilities in Asterisk work as usual.
424 * Added signalling type 'auto', which attempts to use the same signalling type
425 for a channel as configured in DAHDI. This is primarily designed for analog
426 ports, but will also work for digital ports that are configured for FXS or FXO
427 signalling types. This mode is also the default now, so if your chan_dahdi.conf
428 does not specify signalling for a channel (which is unlikely as the sample
429 configuration file has always recommended specifying it for every channel) then
430 the 'auto' mode will be used for that channel if possible.
431 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
432 state for a channel; also ensured that the DNDState Manager event is
433 emitted no matter how the DND state is set or cleared.
437 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
438 configs/unistim.conf.sample for details. This new channel driver allows
439 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
440 * Added a new channel driver, chan_console, which uses portaudio as a cross
441 platform audio interface. It was written as a channel driver that would
442 work with Mac CoreAudio, but portaudio supports a number of other audio
443 interfaces, as well. Note that this channel driver requires v19 or higher
444 of portaudio; older versions have a different API.
448 * Added the ability to specify arguments to the Dial application when using
449 the DUNDi switch in the dialplan.
450 * Added the ability to set weights for responses dynamically. This can be
451 done using a global variable or a dialplan function. Using the SHELL()
452 function would allow you to have an external script set the weight for
454 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
455 functions will allow you to initiate a DUNDi query from the dialplan,
456 find out how many results there are, and access each one.
460 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
461 functions will allow you to initiate an ENUM lookup from the dialplan,
462 and Asterisk will cache the results. ENUMRESULT can be used to access
463 the results without doing multiple DNS queries.
467 * Added the ability to customize which sound files are used for some of the
468 prompts within the Voicemail application by changing them in voicemail.conf
469 * Added the ability for the "voicemail show users" CLI command to show users
470 configured by the dynamic realtime configuration method.
471 * MWI (Message Waiting Indication) handling has been significantly
472 restructured internally to Asterisk. It is now totally event based
473 instead of polling based. The voicemail application will notify other
474 modules that have subscribed to MWI events when something in the mailbox
476 This also means that if any other entity outside of Asterisk is changing
477 the contents of mailboxes, then the voicemail application still needs to
478 poll for changes. Examples of situations that would require this option
479 are web interfaces to voicemail or an email client in the case of using
480 IMAP storage. So, two new options have been added to voicemail.conf
481 to account for this: "pollmailboxes" and "pollfreq". See the sample
482 configuration file for details.
483 * Added "tw" language support
484 * Added support for storage of greetings using an IMAP server
485 * Added ability to customize forward, reverse, stop, and pause keys for message playback
486 * SMDI is now enabled in voicemail using the smdienable option.
487 * A "lockmode" option has been added to asterisk.conf to configure the file
488 locking method used for voicemail, and potentially other things in the
489 future. The default is the old behavior, lockfile. However, there is a
490 new method, "flock", that uses a different method for situations where the
491 lockfile will not work, such as on SMB/CIFS mounts.
492 * Added the ability to backup deleted messages, to ease recovery in the case
493 that a user accidentally deletes a message, and discovers that they need it.
494 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
495 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
496 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
497 voicemail boxes. The SMDI interface can also poll for MWI changes when some
498 outside entity is modifying the state of the mailbox (such as IMAP storage or
499 a web interface of some kind).
500 * Added the support for marking messages as "urgent." There are two methods to accomplish
501 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
502 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
503 the message as urgent after he has recorded a voicemail by following the voice instructions.
504 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
509 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
510 used across multiple queues.
511 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
512 setqueueentryvar options for each queue, see queues.conf.sample for details.
513 * Added keepstats option to queues.conf which will keep queue
514 statistics during a reload.
515 * setinterfacevar option in queues.conf also now sets a variable
516 called MEMBERNAME which contains the member's name.
517 * Added 'Strategy' field to manager event QueueParams which represents
518 the queue strategy in use.
519 * Added option to run macro when a queue member is connected to a caller,
520 see queues.conf.sample for details.
521 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
522 does not count paused queue members as unavailable.
523 * Added min-announce-frequency option to queues.conf which allows you to control the
524 minimum amount of time between queue announcements for use when the caller's queue
525 position changes frequently.
526 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
528 * Added ability for non-realtime queues to have realtime members
529 * Added the "linear" strategy to queues.
530 * Added the "wrandom" strategy to queues.
531 * Added new channel variable QUEUE_MIN_PENALTY
532 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
533 rules in queuerules.conf. See configs/queuerules.conf.sample for details
534 * Added a new parameter for member definition, called state_interface. This may be
535 used so that a member may be called via one interface but have a different interface's
536 device state reported.
537 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
538 specified by the periodic-announce option, then one will be chosen randomly when it is time
539 to play a periodic announcment
540 * New configuration options: announce-position now takes two more values in addition to "yes" and
541 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
542 announce-position-limit. By setting announce-position to "limit" callers will only have their
543 position announced if their position is less than what is specified by announce-position-limit.
544 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
545 will be told that their are more than announce-position-limit callers waiting.
546 * Two new queue log events have been added. An ADDMEMBER event will be logged
547 when a realtime queue member is added and a REMOVEMEMBER event will be logged
548 when a realtime queue member is removed. Since there is no calling channel associated
549 with these events, the string "REALTIME" is placed where the channel's unique id
554 * The 'o' option to provide an optimization has been removed and its functionality
555 has been enabled by default.
556 * When a conference is created, the UNIQUEID of the channel that caused it to be
557 created is stored. Then, every channel that joins the conference will have the
558 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
559 callers that come and go from long standing conferences.
560 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
561 except it does operations on a channel by name, instead of number in a conference.
562 This is a very useful feature in combination with the 'X' option to ChanSpy.
563 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
565 * Added new RealTime functionality to provide support for scheduled conferencing.
566 This includes optional messages to the caller if they attempt to join before
567 the schedule start time, or to allow the caller to join the conference early.
568 Also included is optional support for limiting the number of callers per
570 * Added the S() and L() options to the MeetMe application. These are pretty
571 much identical to the S() and L() options to Dial(). They let you set
572 timeouts for the conference, as well as have warning sounds played to
573 let the caller know how much time is left, and when it is running out.
574 * Added the ability to do "meetme concise" with the "meetme" CLI command.
575 This extends the concise capabilities of this CLI command to include
576 listing all conferences, instead of an addition to the other sub commands
577 for the "meetme" command.
578 * Added the ability to specify the music on hold class used to play into the
579 conference when there is only one member and the M option is used.
580 * Added MEETME_INFO dialplan function which provides a way to query
581 various properties of a Meetme conference.
583 Other Dialplan Application Changes
584 ----------------------------------
585 * Argument support for Gosub application
586 * From the to-do lists: straighten out the app timeout args:
587 Wait() app now really does 0.3 seconds- was truncating arg to an int.
588 WaitExten() same as Wait().
589 Congestion() - Now takes floating pt. argument.
590 Busy() - now takes floating pt. argument.
591 Read() - timeout now can be floating pt.
592 WaitForRing() now takes floating pt timeout arg.
593 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
594 * Added 's' option to Page application.
595 * Added 'E' and 'V' commands to ExternalIVR.
596 * Added 'o' and 'X' options to Chanspy.
597 * Added a new dialplan application, Bridge, which allows you to bridge the
598 calling channel to any other active channel on the system.
599 * Added the ability to specify a music on hold class to play instead of ringing
600 for the SLATrunk application.
601 * The Read application no longer exits the dialplan on error. Instead, it sets
602 READSTATUS to ERROR, which you can catch and handle separately.
603 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
604 of asking for verification of each name, one at a time.
605 * Privacy() no longer uses privacy.conf, as all options are specifyable as
606 direct options to the app.
607 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
609 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
610 * The ChannelRedirect application no longer exits the dialplan if the given channel
611 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
612 or NOCHANNEL if the given channel was not found.
613 * The silencethreshold setting that was previously configurable in multiple
614 applications is now settable globally via dsp.conf.
615 * Added ability to communicate over a TCP socket instead of forking a child process for the
616 ExternalIVR application.
618 Music On Hold Changes
619 ---------------------
620 * A new option, "digit", has been added for music on hold classes in
621 musiconhold.conf. If this is set for a music on hold class, a caller
622 listening to music on hold can press this digit to switch to listening
623 to this music on hold class.
624 * Support for realtime music on hold has been added.
625 * In conjunction with the realtime music on hold, a general section has
626 been added to musiconhold.conf, its sole variable is cachertclasses. If this
627 is set, then music on hold classes found in realtime will be cached in memory.
631 * AEL upgraded to use the Gosub with Arguments instead
632 of Macro application, to hopefully reduce the problems
633 seen with the artificially low stack ceiling that
634 Macro bumps into. Macros can only call other Macros
635 to a depth of 7. Tests run using gosub, show depths
636 limited only by virtual memory. A small test demonstrated
637 recursive call depths of 100,000 without problems.
638 -- in addition to this, all apps that allowed a macro
639 to be called, as in Dial, queues, etc, are now allowing
640 a gosub call in similar fashion.
641 * AEL now generates LOCAL(argname) declarations when it
642 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
643 etc. That makes the arguments local in scope. The user
644 can define their own local variables in macros, now,
645 by saying "local myvar=someval;" or using Set() in this
646 fashion: Set(LOCAL(myvar)=someval); ("local" is now
648 * utils/conf2ael introduced. Will convert an extensions.conf
649 file into extensions.ael. Very crude and unfinished, but
650 will be improved as time goes by. Should be useful for a
651 first pass at conversion.
652 * aelparse will now read extensions.conf to see if a referenced
653 macro or context is there before issueing a warning.
654 * AEL parser sets a local channel variable ~~EXTEN~~, to
655 preserve the value of ${EXTEN} thru switch statements.
656 * New operator in $[...] expressions: the ~~ operator serves
657 as a concatenation operator. AT THE MOMENT, it is really only
658 necessary and useful in AEL, especially in if() expressions.
659 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
660 any enclosing double-quotes, and evaluate to the value of a
661 concatenated with the value of b. For example if a is set to
662 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
666 Call Features (res_features) Changes
667 ------------------------------------
668 * Added the parkedcalltransfers option to features.conf
669 * The built-in method for doing attended transfers has been updated to
670 include some new options that allow you to have the transferee sent
671 back to the person that did the transfer if the transfer is not successful.
672 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
673 in features.conf.sample.
674 * Added support for configuring named groups of custom call features in
675 features.conf. This means that features can be written a single time, and
676 then mapped into groups of features for different key mappings or easier
678 * Updated the ParkedCall application to allow you to not specify a parking
679 extension. If you don't specify a parking space to pick up, it will grab
680 the first one available.
681 * Added cli command 'features reload' to reload call features from features.conf
682 * Moved into core asterisk binary.
684 Language Support Changes
685 ------------------------
686 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
687 * Added support for the Hungarian language for saying numbers, dates, and times.
691 * Added SPEECH commands for speech recognition. A complete listing can be found
693 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
694 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
695 does not behave as expected; the native command needs to be used, instead.
699 * Added rotatestrategy option to logger.conf, along with two new options:
700 "timestamp" which will use the time to name the logger files instead of
701 sequence number; and "rotate", which rotates the names of the logfiles,
702 similar to the way syslog rotates files.
703 * Added exec_after_rotate option to logger.conf, which allows a system
704 command to be run after rotation. This is primarily useful with
705 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
706 and to ensure that the oldest log file gets deleted.
707 * Added realtime support for the queue log
711 * The cdr_manager module has a [mappings] feature, like cdr_custom,
712 to add fields to the manager event from the CDR variables.
713 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
714 backend database CDR table. Specifically, additional, non-standard
715 columns are supported, merely by setting the corresponding CDR variable in
716 your dialplan. In addition, you may alias any column to another name (for
717 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
718 simply "alias src => ANI" in the configuration file). Records may be
719 posted to more than one backend, simply by specifying multiple categories
720 in the configuration file. And finally, you may filter which CDRs get
721 posted to each backend, by specifying a filter (which the record must
722 match) for the particular category. Filters are additive (meaning all
723 rules must match to post that CDR).
724 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
725 module. Specifically, you may add additional columns into the table and
726 they will be set, if you set the corresponding CDR variable name. Also,
727 if you omit columns in your database table, they will be silently skipped
728 (but a record will still be inserted, based on what columns remain). Note
729 that the other two features from cdr_adaptive_odbc (alias and filter) are
730 not currently supported.
731 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
732 has been disabled using the NoCDR application.
734 Miscellaneous New Modules
735 -------------------------
736 * Added a new CDR module, cdr_sqlite3_custom.
737 * Added a new realtime configuration module, res_config_sqlite
738 * Added a new codec translation module, codec_resample, which re-samples
739 signed linear audio between 8 kHz and 16 kHz to help support wideband
741 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
742 based on configuration templates that use Asterisk dialplan function and
743 variable substitution. It should be possible to create phone profiles and
744 templates that work for the majority of phones provisioned over http. It
745 is currently only intended to provision a single user account per phone.
746 An example profile and set of templates for Polycom phones is provided.
747 NOTE: Polycom firmware is not included, but should be placed in
748 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
749 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
750 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
751 provided; there is a JACK() application, and a JACK_HOOK() function. Both
752 interfaces create an input and output JACK port. The application makes
753 these ports the endpoint of the call. The audio coming from the channel
754 goes out the output port and whatever comes back in on the input port is
755 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
756 audiohook on the channel. This lets you run the audio coming from a
757 channel through JACK, and whatever comes back in is what gets forwarded
758 on as the channel's audio. This is very useful for building custom
759 vocoders or doing recording or analysis of the channel's audio in another
761 * Added a new module, res_config_curl, which permits using a HTTP POST url
762 to retrieve, create, update, and delete realtime information from a remote
763 web server. Note that this module requires func_curl.so to be loaded for
764 backend functionality.
765 * Added a new module, res_config_ldap, which permits the use of an LDAP
766 server for realtime data access.
767 * Added support for writing and running your dialplan in lua using the pbx_lua
768 module. See configs/extensions.lua.sample for examples of how to do this.
772 * Ability to use libcap to set high ToS bits when non-root
773 on Linux. If configure is unable to find libcap then you
774 can use --with-cap to specify the path.
775 * Added maxfiles option to options section of asterisk.conf which allows you to specify
776 what Asterisk should set as the maximum number of open files when it loads.
777 * Added the jittertargetextra configuration option.
778 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
779 configuration files for the IP channel drivers. The new option is "cos".
780 This information is also documented in doc/qos.tex, or the IP Quality of Service
781 section of asterisk.pdf.
782 * When originating a call using AMI or pbx_spool that fails the reason for failure
783 will now be available in the failed extension using the REASON dialplan variable.
784 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
785 It allows you to configure a prefix for auto-monitor recordings.
786 * A new extension pattern matching algorithm, based on a trie, is introduced
787 here, that could noticeably speed up mid-sized to large dialplans.
788 It is NOT used by default, as duplicating the behaviour of the old pattern
789 matcher is still under development. A config file option, in extensions.conf,
790 in the [general] section, called "extenpatternmatchingnew", is by default
791 set to false; setting that to true will force the use of the new algorithm.
792 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
793 be used to switch the algorithms at run time.
794 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
795 specifying which socket to use to connect to the running Asterisk daemon
797 * Performance enhancements to the sched facility, which is used in
798 the channel drivers, etc. Added hashtabs and doubly-linked lists
799 to speed up deletion; start at the beginning or end of list to
801 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
802 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
803 Added regression tests to the tests/ dir, also.
804 * Added a refcount trace feature to astobj2 for those trying to balance
805 object creation, deletion; work, play; space and time. See the
806 notes in astobj2.h. Also, see utils/refcounter as well, as a
807 quick way to find unbalanced refcounts in what could be a sea
808 of objects that were balanced.
809 * Added logging to 'make update' command. See update.log
810 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
811 do not come from the remote party.
812 * Added the 'n' option to the SpeechBackground application to tell it to not
813 answer the channel if it has not already been answered.
814 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
815 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
817 * iLBC source code no longer included (see UPGRADE.txt for details)
818 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
819 deadlock is detected, a backtrace of the stack which led to the lock calls
820 will be output to the CLI.
821 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
822 the "core show locks" CLI command will give lock information output as well
823 as a backtrace of the stack which led to the lock calls.
824 * users.conf now sports an optional alternateexts property, which permits
825 allocation of additional extensions which will reach the specified user.
826 * A new option for the configure script, --enable-internal-poll, has been added
827 for use with systems which may have a buggy implementation of the poll system
828 call. If you notice odd behavior such as the CLI being unresponsive on remote
829 consoles, you may want to try using this option. This option is enabled by default
830 on Darwin systems since it is known that the Darwin poll() implementation has
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