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