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)}/" .
108 * The ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using setvar to cause a given
109 audio file to be played upon completion of an attended transfer.
110 * Added DNS manager support to registrations for peers referencing peer entries.
111 DNS manager runs in the background which allows DNS lookups to be run asynchronously
112 as well as periodically updating the IP address. These properties allow for
113 better performance as well as recovery in the event of an IP change.
114 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
115 load/reload of large numbers of peers/users by ~40x (for large lists of peers.
116 Initially, we saw 4x improvement in call setup/destruction, but at the time
117 of merging, this gain has disappeared; further research will be done to try
118 and restore this performance improvement. Astobj2 refcounting is now used
119 for users, peers, and dialogs. Users are encouraged to assist in regression
120 testing and problem reporting!
121 * Added ability to specify registration expiry time on a per registration basis in
123 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
125 * Added t38pt_usertpsource option. See sip.conf.sample for details.
126 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
127 * 'sip show peers' and 'sip show users' display their entries sorted in
128 alphabetical order, as opposed to the order they were in, in the config
130 * Videosupport now supports an additional option, "always", which always sets
131 up video RTP ports, even on clients that don't support it. This helps with
132 callfiles and certain transfers to ensure that if two video phones are
133 connected, they will always share video feeds.
137 * Existing DNS manager lookups extended to check for SRV records.
141 * New CLI command, "config reload <file.conf>" which reloads any module that
142 references that particular configuration file. Also added "config list"
143 which shows which configuration files are in use.
144 * New CLI commands, "pri show version" and "ss7 show version" that will
145 display which version of libpri and libss7 are being used, respectively.
146 A new API call was added so trunk will now have to be compiled against
147 a versions of libpri and libss7 that have them or it will not know that
148 these libraries exist.
149 * The commands "core show globals", "core set global" and "core set chanvar" has
150 been deprecated in favor of the more semanticly correct "dialplan show globals",
151 "dialplan set chanvar" and "dialplan set global".
152 * New CLI command "dialplan show chanvar" to list all variables associated
153 with a given channel.
157 * Addresses managed by DNS manager now can check to see if there is a DNS
158 SRV record for a given domain and will use that hostname/port if present.
160 AMI - The manager (TCP/TLS/HTTP)
161 --------------------------------
162 * The Status command now takes an optional list of variables to display
163 along with channel status.
167 * res_odbc no longer has a limit of 1023 total possible unshared connections,
168 as some people were running into this limit. This limit has been increased
173 * The TRANSFER queue log entry now includes the the caller's original
174 position in the transferred-from queue.
175 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
176 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
177 as well as an explanation about timeout options in general
179 ------------------------------------------------------------------------------
180 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
181 ------------------------------------------------------------------------------
183 AMI - The manager (TCP/TLS/HTTP)
184 --------------------------------
185 * Manager has undergone a lot of changes, all of them documented
186 in doc/manager_1_1.txt
187 * Manager version has changed to 1.1
188 * Added a new action 'CoreShowChannels' to list currently defined channels
189 and some information about them.
190 * Added a new action 'SIPshowregistry' to list SIP registrations.
191 * Added TLS support for the manager interface and HTTP server
192 * Added the URI redirect option for the built-in HTTP server
193 * The output of CallerID in Manager events is now more consistent.
194 CallerIDNum is used for number and CallerIDName for name.
195 * Enable https support for builtin web server.
196 See configs/http.conf.sample for details.
197 * Added a new action, GetConfigJSON, which can return the contents of an
198 Asterisk configuration file in JSON format. This is intended to help
199 improve the performance of AJAX applications using the manager interface
201 * SIP and IAX manager events now use "ChannelType" in all cases where we
202 indicate channel driver. Previously, we used a mixture of "Channel"
203 and "ChannelDriver" headers.
204 * Added a "Bridge" action which allows you to bridge any two channels that
205 are currently active on the system.
206 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
207 the voicemail users setup.
208 * Added 'DBDel' and 'DBDelTree' manager commands.
209 * cdr_manager now reports events via the "cdr" level, separating it from
210 the very verbose "call" level.
211 * Manager users are now stored in memory. If you change the manager account
212 list (delete or add accounts) you need to reload manager.
213 * Added Masquerade manager event for when a masquerade happens between
215 * Added "manager reload" command for the CLI
216 * Lots of commands that only provided information are now allowed under the
217 Reporting privilege, instead of only under Call or System.
218 * The IAX* commands now require either System or Reporting privilege, to
219 mirror the privileges of the SIP* commands.
220 * Added ability to retrieve list of categories in a config file.
221 * Added ability to retrieve the content of a particular category.
222 * Added ability to empty a context.
223 * Created new action to create a new file.
224 * Updated delete action to allow deletion by line number with respect to category.
225 * Added new action insert to add new variable to category at specified line.
226 * Updated action newcat to allow new category to be inserted in file above another
228 * Added new event "JitterBufStats" in the IAX2 channel
229 * Originate now requires the Originate privilege and, if you want to call out
230 to a subshell, it requires the System privilege, as well. This was done to
231 enhance manager security.
232 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
233 * New command: Atxfer. See doc/manager_1_1.txt for more details or
234 manager show command Atxfer from the CLI
238 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
239 state in the dialplan, as well as creating custom device states that are
240 controllable from the dialplan.
241 * Extend CALLERID() function with "pres" and "ton" parameters to
242 fetch string representation of calling number presentation indicator
243 and numeric representation of type of calling number value.
244 * MailboxExists converted to dialplan function
245 * A new option to Dial() for telling IP phones not to count the call
246 as "missed" when dial times out and cancels.
247 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
248 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
249 held for any given channel. Also, locks are automatically freed when a
251 * Added HINT() dialplan function that allows retrieving hint information.
252 Hints are mappings between extensions and devices for the sake of
253 determining the state of an extension. This function can retrieve the list
254 of devices or the name associated with a hint.
255 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
257 * Added SYSINFO() dialplan function which allows retrieval of system information
258 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
259 the existence of a dialplan target.
260 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
261 upper and lower case, respectively.
262 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
263 ID for the call (not the Asterisk call ID or unique ID), provided that the
264 channel driver supports this. For SIP, you get the SIP call-ID for the
265 bridged channel which you can store in the CDR with a custom field.
269 * New CLI command "core show hint" (usage: core show hint <exten>)
270 * New CLI command "core show settings"
271 * Added 'core show channels count' CLI command.
272 * Added the ability to set the core debug and verbose values on a per-file basis.
273 * Added 'queue pause member' and 'queue unpause member' CLI commands
274 * Ability to set process limits ("ulimit") without restarting Asterisk
275 * Enhanced "agi debug" to print the channel name as a prefix to the debug
276 output to make debugging on busy systems much easier.
277 * New CLI commands "dialplan set extenpatternmatching true/false"
278 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
279 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
280 listed in the startup_commands section of cli.conf will get executed.
281 * Added a CLI command, "devstate change", which allows you to set custom device
282 states from the func_devstate module that provides the DEVICE_STATE() function
283 and handling of the "Custom:" devices.
284 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
285 sorted into the different possible callbacks, with the number of entries
286 currently scheduled for each. Gives you a feel for how busy the sip channel
291 * Improved NAT and STUN support.
292 chan_sip now can use port numbers in bindaddr, externip and externhost
293 options, as well as contact a STUN server to detect its external address
294 for the SIP socket. See sip.conf.sample, 'NAT' section.
295 * The default SIP useragent= identifier now includes the Asterisk version
296 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
297 If set, and the incoming request carries authentication info,
298 the username to match in the users list is taken from the Digest header
299 rather than from the From: field. This feature is considered experimental.
300 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
301 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
302 * The "localmask" setting was removed in version 1.2 and the reminder about it
303 being removed is now also removed.
304 * A new option "busylevel" for setting a level of calls where asterisk reports
305 a device as busy, to separate it from call-limit. This value is also added
306 to the SIP_PEER dialplan function.
307 * A new realtime family called "sipregs" is now supported to store SIP registration
308 data. If this family is defined, "sippeers" will be used for configuration and
309 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
310 registration data, as before.
311 * The SIPPEER function have new options for port address, call and pickup groups
312 * Added support for T.140 realtime text in SIP/RTP
313 * The "checkmwi" option has been removed from sip.conf, as it is no longer
314 required due to the restructuring of how MWI is handled. See the descriptions
315 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
316 for more information.
317 * Added rtpdest option to CHANNEL() dialplan function.
318 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
319 * SIP now adds a header to the CANCEL if the call was answered by another phone
320 in the same dial command, or if the new c option in dial() is used.
321 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
322 states it is not needed. For phones, however, that do require it the "registertrying" option
323 has been added so it can be enabled.
324 * A new option called "callcounter" (global/peer/user level) enables call counters needed
325 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
326 used to enable this functionality).
327 * New settings for timer T1 and timer B on a global level or per device. This makes it
328 possible to force timeout faster on non-responsive SIP servers. These settings are
329 considered advanced, so don't use them unless you have a problem.
330 * Added a dial string option to be able to set the To: header in an INVITE to any
332 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
333 the qualify frequency.
334 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
335 were not properly torn down due to network or endpoint failures during an established
337 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
338 configs/sip.conf.sample for more information on how it is used.
339 * Added a new configuration option "authfailureevents" that enables manager events when
340 a peer can't authenticate properly.
341 * Added DNS manager support to registrations for peers not referencing a peer entry.
345 * Added the trunkmaxsize configuration option to chan_iax2.
346 * Added the srvlookup option to iax.conf
347 * Added support for OSP. The token is set and retrieved through the CHANNEL()
350 XMPP Google Talk/Jingle changes
351 -------------------------------
352 * Added the bindaddr option to gtalk.conf.
356 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
357 * Proper codec support in chan_skinny.
358 * Added settings for IP and Ethernet QoS requests
362 * Added separate settings for media QoS in mgcp.conf
364 Console Channel Driver changes
365 ------------------------------
366 * Added experimental support for video send & receive to chan_oss.
367 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
370 Phone channel changes (chan_phone)
371 ----------------------------------
372 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
374 H.323 channel Changes
375 ---------------------
376 * H323 remote hold notification support added (by NOTIFY message
377 and/or H.450 supplementary service)
379 Local channel changes
380 ---------------------
381 * The device state functionality in the Local channel driver has been updated
382 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
383 to just UNKNOWN if the extension exists.
384 * Added jitterbuffer support for chan_local. This allows you to use the
385 generic jitterbuffer on incoming calls going to Asterisk applications.
386 For example, this would allow you to use a jitterbuffer for an incoming
387 SIP call to Voicemail by putting a Local channel in the middle. This
388 feature is enabled by using the 'j' option in the Dial string to the Local
389 channel in conjunction with the existing 'n' option for local channels.
390 * A 'b' option has been added which causes chan_local to return the actual channel
391 that is behind it when queried. This is useful for transfer scenarios as the
392 actual channel will be transferred, not the Local channel.
394 Agent channel changes
395 ----------------------
396 * The ackcall and endcall options are now supplemented with options acceptdtmf
397 and enddtmf. These allow for the DTMF keypress to be configurable. The options
398 default to their old hard-coded values ('#' and '*' respectively) so this should
399 not break any existing agent installations.
401 DAHDI channel driver (chan_dahdi) Changes
402 ----------------------------------------
403 * SS7 support (via libss7 library)
404 * In India, some carriers transmit CID via dtmf. Some code has been added
405 that will handle some situations. The cidstart=polarity_IN choice has been added for
406 those carriers that transmit CID via dtmf after a polarity change.
407 * CID matching information is now shown when doing 'dialplan show'.
408 * Added dahdi show version CLI command.
409 * Added setvar support to chan_dahdi.conf channel entries.
410 * Added two new options: mwimonitor and mwimonitornotify. These options allow
411 you to enable MWI monitoring on FXO lines. When the MWI state changes,
412 the script specified in the mwimonitornotify option is executed. An internal
413 event indicating the new state of the mailbox is also generated, so that
414 the normal MWI facilities in Asterisk work as usual.
415 * Added signalling type 'auto', which attempts to use the same signalling type
416 for a channel as configured in DAHDI. This is primarily designed for analog
417 ports, but will also work for digital ports that are configured for FXS or FXO
418 signalling types. This mode is also the default now, so if your chan_dahdi.conf
419 does not specify signalling for a channel (which is unlikely as the sample
420 configuration file has always recommended specifying it for every channel) then
421 the 'auto' mode will be used for that channel if possible.
422 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
423 state for a channel; also ensured that the DNDState Manager event is
424 emitted no matter how the DND state is set or cleared.
428 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
429 configs/unistim.conf.sample for details. This new channel driver allows
430 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
431 * Added a new channel driver, chan_console, which uses portaudio as a cross
432 platform audio interface. It was written as a channel driver that would
433 work with Mac CoreAudio, but portaudio supports a number of other audio
434 interfaces, as well. Note that this channel driver requires v19 or higher
435 of portaudio; older versions have a different API.
439 * Added the ability to specify arguments to the Dial application when using
440 the DUNDi switch in the dialplan.
441 * Added the ability to set weights for responses dynamically. This can be
442 done using a global variable or a dialplan function. Using the SHELL()
443 function would allow you to have an external script set the weight for
445 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
446 functions will allow you to initiate a DUNDi query from the dialplan,
447 find out how many results there are, and access each one.
451 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
452 functions will allow you to initiate an ENUM lookup from the dialplan,
453 and Asterisk will cache the results. ENUMRESULT can be used to access
454 the results without doing multiple DNS queries.
458 * Added the ability to customize which sound files are used for some of the
459 prompts within the Voicemail application by changing them in voicemail.conf
460 * Added the ability for the "voicemail show users" CLI command to show users
461 configured by the dynamic realtime configuration method.
462 * MWI (Message Waiting Indication) handling has been significantly
463 restructured internally to Asterisk. It is now totally event based
464 instead of polling based. The voicemail application will notify other
465 modules that have subscribed to MWI events when something in the mailbox
467 This also means that if any other entity outside of Asterisk is changing
468 the contents of mailboxes, then the voicemail application still needs to
469 poll for changes. Examples of situations that would require this option
470 are web interfaces to voicemail or an email client in the case of using
471 IMAP storage. So, two new options have been added to voicemail.conf
472 to account for this: "pollmailboxes" and "pollfreq". See the sample
473 configuration file for details.
474 * Added "tw" language support
475 * Added support for storage of greetings using an IMAP server
476 * Added ability to customize forward, reverse, stop, and pause keys for message playback
477 * SMDI is now enabled in voicemail using the smdienable option.
478 * A "lockmode" option has been added to asterisk.conf to configure the file
479 locking method used for voicemail, and potentially other things in the
480 future. The default is the old behavior, lockfile. However, there is a
481 new method, "flock", that uses a different method for situations where the
482 lockfile will not work, such as on SMB/CIFS mounts.
483 * Added the ability to backup deleted messages, to ease recovery in the case
484 that a user accidentally deletes a message, and discovers that they need it.
485 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
486 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
487 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
488 voicemail boxes. The SMDI interface can also poll for MWI changes when some
489 outside entity is modifying the state of the mailbox (such as IMAP storage or
490 a web interface of some kind).
491 * Added the support for marking messages as "urgent." There are two methods to accomplish
492 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
493 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
494 the message as urgent after he has recorded a voicemail by following the voice instructions.
495 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
500 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
501 used across multiple queues.
502 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
503 setqueueentryvar options for each queue, see queues.conf.sample for details.
504 * Added keepstats option to queues.conf which will keep queue
505 statistics during a reload.
506 * setinterfacevar option in queues.conf also now sets a variable
507 called MEMBERNAME which contains the member's name.
508 * Added 'Strategy' field to manager event QueueParams which represents
509 the queue strategy in use.
510 * Added option to run macro when a queue member is connected to a caller,
511 see queues.conf.sample for details.
512 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
513 does not count paused queue members as unavailable.
514 * Added min-announce-frequency option to queues.conf which allows you to control the
515 minimum amount of time between queue announcements for use when the caller's queue
516 position changes frequently.
517 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
519 * Added ability for non-realtime queues to have realtime members
520 * Added the "linear" strategy to queues.
521 * Added the "wrandom" strategy to queues.
522 * Added new channel variable QUEUE_MIN_PENALTY
523 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
524 rules in queuerules.conf. See configs/queuerules.conf.sample for details
525 * Added a new parameter for member definition, called state_interface. This may be
526 used so that a member may be called via one interface but have a different interface's
527 device state reported.
528 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
529 specified by the periodic-announce option, then one will be chosen randomly when it is time
530 to play a periodic announcment
531 * New configuration options: announce-position now takes two more values in addition to "yes" and
532 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
533 announce-position-limit. By setting announce-position to "limit" callers will only have their
534 position announced if their position is less than what is specified by announce-position-limit.
535 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
536 will be told that their are more than announce-position-limit callers waiting.
537 * Two new queue log events have been added. An ADDMEMBER event will be logged
538 when a realtime queue member is added and a REMOVEMEMBER event will be logged
539 when a realtime queue member is removed. Since there is no calling channel associated
540 with these events, the string "REALTIME" is placed where the channel's unique id
545 * The 'o' option to provide an optimization has been removed and its functionality
546 has been enabled by default.
547 * When a conference is created, the UNIQUEID of the channel that caused it to be
548 created is stored. Then, every channel that joins the conference will have the
549 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
550 callers that come and go from long standing conferences.
551 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
552 except it does operations on a channel by name, instead of number in a conference.
553 This is a very useful feature in combination with the 'X' option to ChanSpy.
554 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
556 * Added new RealTime functionality to provide support for scheduled conferencing.
557 This includes optional messages to the caller if they attempt to join before
558 the schedule start time, or to allow the caller to join the conference early.
559 Also included is optional support for limiting the number of callers per
561 * Added the S() and L() options to the MeetMe application. These are pretty
562 much identical to the S() and L() options to Dial(). They let you set
563 timeouts for the conference, as well as have warning sounds played to
564 let the caller know how much time is left, and when it is running out.
565 * Added the ability to do "meetme concise" with the "meetme" CLI command.
566 This extends the concise capabilities of this CLI command to include
567 listing all conferences, instead of an addition to the other sub commands
568 for the "meetme" command.
569 * Added the ability to specify the music on hold class used to play into the
570 conference when there is only one member and the M option is used.
571 * Added MEETME_INFO dialplan function which provides a way to query
572 various properties of a Meetme conference.
574 Other Dialplan Application Changes
575 ----------------------------------
576 * Argument support for Gosub application
577 * From the to-do lists: straighten out the app timeout args:
578 Wait() app now really does 0.3 seconds- was truncating arg to an int.
579 WaitExten() same as Wait().
580 Congestion() - Now takes floating pt. argument.
581 Busy() - now takes floating pt. argument.
582 Read() - timeout now can be floating pt.
583 WaitForRing() now takes floating pt timeout arg.
584 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
585 * Added 's' option to Page application.
586 * Added 'E' and 'V' commands to ExternalIVR.
587 * Added 'o' and 'X' options to Chanspy.
588 * Added a new dialplan application, Bridge, which allows you to bridge the
589 calling channel to any other active channel on the system.
590 * Added the ability to specify a music on hold class to play instead of ringing
591 for the SLATrunk application.
592 * The Read application no longer exits the dialplan on error. Instead, it sets
593 READSTATUS to ERROR, which you can catch and handle separately.
594 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
595 of asking for verification of each name, one at a time.
596 * Privacy() no longer uses privacy.conf, as all options are specifyable as
597 direct options to the app.
598 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
600 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
601 * The ChannelRedirect application no longer exits the dialplan if the given channel
602 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
603 or NOCHANNEL if the given channel was not found.
604 * The silencethreshold setting that was previously configurable in multiple
605 applications is now settable globally via dsp.conf.
606 * Added ability to communicate over a TCP socket instead of forking a child process for the
607 ExternalIVR application.
609 Music On Hold Changes
610 ---------------------
611 * A new option, "digit", has been added for music on hold classes in
612 musiconhold.conf. If this is set for a music on hold class, a caller
613 listening to music on hold can press this digit to switch to listening
614 to this music on hold class.
615 * Support for realtime music on hold has been added.
616 * In conjunction with the realtime music on hold, a general section has
617 been added to musiconhold.conf, its sole variable is cachertclasses. If this
618 is set, then music on hold classes found in realtime will be cached in memory.
622 * AEL upgraded to use the Gosub with Arguments instead
623 of Macro application, to hopefully reduce the problems
624 seen with the artificially low stack ceiling that
625 Macro bumps into. Macros can only call other Macros
626 to a depth of 7. Tests run using gosub, show depths
627 limited only by virtual memory. A small test demonstrated
628 recursive call depths of 100,000 without problems.
629 -- in addition to this, all apps that allowed a macro
630 to be called, as in Dial, queues, etc, are now allowing
631 a gosub call in similar fashion.
632 * AEL now generates LOCAL(argname) declarations when it
633 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
634 etc. That makes the arguments local in scope. The user
635 can define their own local variables in macros, now,
636 by saying "local myvar=someval;" or using Set() in this
637 fashion: Set(LOCAL(myvar)=someval); ("local" is now
639 * utils/conf2ael introduced. Will convert an extensions.conf
640 file into extensions.ael. Very crude and unfinished, but
641 will be improved as time goes by. Should be useful for a
642 first pass at conversion.
643 * aelparse will now read extensions.conf to see if a referenced
644 macro or context is there before issueing a warning.
645 * AEL parser sets a local channel variable ~~EXTEN~~, to
646 preserve the value of ${EXTEN} thru switch statements.
647 * New operator in $[...] expressions: the ~~ operator serves
648 as a concatenation operator. AT THE MOMENT, it is really only
649 necessary and useful in AEL, especially in if() expressions.
650 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
651 any enclosing double-quotes, and evaluate to the value of a
652 concatenated with the value of b. For example if a is set to
653 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
657 Call Features (res_features) Changes
658 ------------------------------------
659 * Added the parkedcalltransfers option to features.conf
660 * The built-in method for doing attended transfers has been updated to
661 include some new options that allow you to have the transferee sent
662 back to the person that did the transfer if the transfer is not successful.
663 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
664 in features.conf.sample.
665 * Added support for configuring named groups of custom call features in
666 features.conf. This means that features can be written a single time, and
667 then mapped into groups of features for different key mappings or easier
669 * Updated the ParkedCall application to allow you to not specify a parking
670 extension. If you don't specify a parking space to pick up, it will grab
671 the first one available.
672 * Added cli command 'features reload' to reload call features from features.conf
673 * Moved into core asterisk binary.
675 Language Support Changes
676 ------------------------
677 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
678 * Added support for the Hungarian language for saying numbers, dates, and times.
682 * Added SPEECH commands for speech recognition. A complete listing can be found
684 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
685 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
686 does not behave as expected; the native command needs to be used, instead.
690 * Added rotatestrategy option to logger.conf, along with two new options:
691 "timestamp" which will use the time to name the logger files instead of
692 sequence number; and "rotate", which rotates the names of the logfiles,
693 similar to the way syslog rotates files.
694 * Added exec_after_rotate option to logger.conf, which allows a system
695 command to be run after rotation. This is primarily useful with
696 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
697 and to ensure that the oldest log file gets deleted.
698 * Added realtime support for the queue log
702 * The cdr_manager module has a [mappings] feature, like cdr_custom,
703 to add fields to the manager event from the CDR variables.
704 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
705 backend database CDR table. Specifically, additional, non-standard
706 columns are supported, merely by setting the corresponding CDR variable in
707 your dialplan. In addition, you may alias any column to another name (for
708 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
709 simply "alias src => ANI" in the configuration file). Records may be
710 posted to more than one backend, simply by specifying multiple categories
711 in the configuration file. And finally, you may filter which CDRs get
712 posted to each backend, by specifying a filter (which the record must
713 match) for the particular category. Filters are additive (meaning all
714 rules must match to post that CDR).
715 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
716 module. Specifically, you may add additional columns into the table and
717 they will be set, if you set the corresponding CDR variable name. Also,
718 if you omit columns in your database table, they will be silently skipped
719 (but a record will still be inserted, based on what columns remain). Note
720 that the other two features from cdr_adaptive_odbc (alias and filter) are
721 not currently supported.
722 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
723 has been disabled using the NoCDR application.
725 Miscellaneous New Modules
726 -------------------------
727 * Added a new CDR module, cdr_sqlite3_custom.
728 * Added a new realtime configuration module, res_config_sqlite
729 * Added a new codec translation module, codec_resample, which re-samples
730 signed linear audio between 8 kHz and 16 kHz to help support wideband
732 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
733 based on configuration templates that use Asterisk dialplan function and
734 variable substitution. It should be possible to create phone profiles and
735 templates that work for the majority of phones provisioned over http. It
736 is currently only intended to provision a single user account per phone.
737 An example profile and set of templates for Polycom phones is provided.
738 NOTE: Polycom firmware is not included, but should be placed in
739 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
740 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
741 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
742 provided; there is a JACK() application, and a JACK_HOOK() function. Both
743 interfaces create an input and output JACK port. The application makes
744 these ports the endpoint of the call. The audio coming from the channel
745 goes out the output port and whatever comes back in on the input port is
746 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
747 audiohook on the channel. This lets you run the audio coming from a
748 channel through JACK, and whatever comes back in is what gets forwarded
749 on as the channel's audio. This is very useful for building custom
750 vocoders or doing recording or analysis of the channel's audio in another
752 * Added a new module, res_config_curl, which permits using a HTTP POST url
753 to retrieve, create, update, and delete realtime information from a remote
754 web server. Note that this module requires func_curl.so to be loaded for
755 backend functionality.
756 * Added a new module, res_config_ldap, which permits the use of an LDAP
757 server for realtime data access.
758 * Added support for writing and running your dialplan in lua using the pbx_lua
759 module. See configs/extensions.lua.sample for examples of how to do this.
763 * Ability to use libcap to set high ToS bits when non-root
764 on Linux. If configure is unable to find libcap then you
765 can use --with-cap to specify the path.
766 * Added maxfiles option to options section of asterisk.conf which allows you to specify
767 what Asterisk should set as the maximum number of open files when it loads.
768 * Added the jittertargetextra configuration option.
769 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
770 configuration files for the IP channel drivers. The new option is "cos".
771 This information is also documented in doc/qos.tex, or the IP Quality of Service
772 section of asterisk.pdf.
773 * When originating a call using AMI or pbx_spool that fails the reason for failure
774 will now be available in the failed extension using the REASON dialplan variable.
775 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
776 It allows you to configure a prefix for auto-monitor recordings.
777 * A new extension pattern matching algorithm, based on a trie, is introduced
778 here, that could noticeably speed up mid-sized to large dialplans.
779 It is NOT used by default, as duplicating the behaviour of the old pattern
780 matcher is still under development. A config file option, in extensions.conf,
781 in the [general] section, called "extenpatternmatchingnew", is by default
782 set to false; setting that to true will force the use of the new algorithm.
783 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
784 be used to switch the algorithms at run time.
785 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
786 specifying which socket to use to connect to the running Asterisk daemon
788 * Performance enhancements to the sched facility, which is used in
789 the channel drivers, etc. Added hashtabs and doubly-linked lists
790 to speed up deletion; start at the beginning or end of list to
792 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
793 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
794 Added regression tests to the tests/ dir, also.
795 * Added a refcount trace feature to astobj2 for those trying to balance
796 object creation, deletion; work, play; space and time. See the
797 notes in astobj2.h. Also, see utils/refcounter as well, as a
798 quick way to find unbalanced refcounts in what could be a sea
799 of objects that were balanced.
800 * Added logging to 'make update' command. See update.log
801 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
802 do not come from the remote party.
803 * Added the 'n' option to the SpeechBackground application to tell it to not
804 answer the channel if it has not already been answered.
805 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
806 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
808 * iLBC source code no longer included (see UPGRADE.txt for details)
809 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
810 deadlock is detected, a backtrace of the stack which led to the lock calls
811 will be output to the CLI.
812 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
813 the "core show locks" CLI command will give lock information output as well
814 as a backtrace of the stack which led to the lock calls.
815 * users.conf now sports an optional alternateexts property, which permits
816 allocation of additional extensions which will reach the specified user.