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.
10 Zaptel channel driver (chan_zap) Changes
11 ----------------------------------------
12 * Channels can now be configured using named sections in zapata.conf, just
13 like other channel drivers, including the use of templates.
17 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
18 to something that matches the pattern a hint will be created using the contents
19 and variables evaluated.
23 * Directory now permits both first and last names to be matched at the same
24 time. In addition, the number of digits to enter of the name can be set in
25 the arguments to Directory; previously, you could enter only 3, regardless
26 of how many names are in your company. For large companies, this should be
28 * Voicemail now permits a mailbox setting to wrap around from first to last
29 messages, if the "messagewrap" option is set to a true value.
30 * Dial has a new option: F(context^extension^pri), which permits a callee to
31 continue in the dialplan, at the specified label, if the caller hangs up.
32 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
33 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
37 * The ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using setvar to cause a given
38 audio file to be played upon completion of an attended transfer.
39 * Added DNS manager support to registrations for peers referencing peer entries.
40 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
41 load/reload of large numbers of peers/users by ~40x (for large lists of peers.
42 Initially, we saw 4x improvement in call setup/destruction, but at the time
43 of merging, this gain has disappeared; further research will be done to try
44 and restore this performance improvement. Astobj2 refcounting is now used
45 for users, peers, and dialogs. Users are encouraged to assist in regression
46 testing and problem reporting!
50 * Existing DNS manager lookups extended to check for SRV records.
54 * New CLI command, "config reload <file.conf>" which reloads any module that
55 references that particular configuration file. Also added "config list"
56 which shows which configuration files are in use.
60 * Addresses managed by DNS manager now can check to see if there is a DNS
61 SRV record for a given domain and will use that hostname/port if present.
63 ------------------------------------------------------------------------------
64 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
65 ------------------------------------------------------------------------------
67 AMI - The manager (TCP/TLS/HTTP)
68 --------------------------------
69 * Manager has undergone a lot of changes, all of them documented
70 in doc/manager_1_1.txt
71 * Manager version has changed to 1.1
72 * Added a new action 'CoreShowChannels' to list currently defined channels
73 and some information about them.
74 * Added a new action 'SIPshowregistry' to list SIP registrations.
75 * Added TLS support for the manager interface and HTTP server
76 * Added the URI redirect option for the built-in HTTP server
77 * The output of CallerID in Manager events is now more consistent.
78 CallerIDNum is used for number and CallerIDName for name.
79 * Enable https support for builtin web server.
80 See configs/http.conf.sample for details.
81 * Added a new action, GetConfigJSON, which can return the contents of an
82 Asterisk configuration file in JSON format. This is intended to help
83 improve the performance of AJAX applications using the manager interface
85 * SIP and IAX manager events now use "ChannelType" in all cases where we
86 indicate channel driver. Previously, we used a mixture of "Channel"
87 and "ChannelDriver" headers.
88 * Added a "Bridge" action which allows you to bridge any two channels that
89 are currently active on the system.
90 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
91 the voicemail users setup.
92 * Added 'DBDel' and 'DBDelTree' manager commands.
93 * cdr_manager now reports events via the "cdr" level, separating it from
94 the very verbose "call" level.
95 * Manager users are now stored in memory. If you change the manager account
96 list (delete or add accounts) you need to reload manager.
97 * Added Masquerade manager event for when a masquerade happens between
99 * Added "manager reload" command for the CLI
100 * Lots of commands that only provided information are now allowed under the
101 Reporting privilege, instead of only under Call or System.
102 * The IAX* commands now require either System or Reporting privilege, to
103 mirror the privileges of the SIP* commands.
104 * Added ability to retrieve list of categories in a config file.
105 * Added ability to retrieve the content of a particular category.
106 * Added ability to empty a context.
107 * Created new action to create a new file.
108 * Updated delete action to allow deletion by line number with respect to category.
109 * Added new action insert to add new variable to category at specified line.
110 * Updated action newcat to allow new category to be inserted in file above another
112 * Added new event "JitterBufStats" in the IAX2 channel
113 * Originate now requires the Originate privilege and, if you want to call out
114 to a subshell, it requires the System privilege, as well. This was done to
115 enhance manager security.
116 * New command: Atxfer. See doc/manager_1_1.txt for more details or
117 manager show command Atxfer from the CLI
121 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
122 state in the dialplan, as well as creating custom device states that are
123 controllable from the dialplan.
124 * Extend CALLERID() function with "pres" and "ton" parameters to
125 fetch string representation of calling number presentation indicator
126 and numeric representation of type of calling number value.
127 * MailboxExists converted to dialplan function
128 * A new option to Dial() for telling IP phones not to count the call
129 as "missed" when dial times out and cancels.
130 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
131 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
132 held for any given channel. Also, locks are automatically freed when a
134 * Added HINT() dialplan function that allows retrieving hint information.
135 Hints are mappings between extensions and devices for the sake of
136 determining the state of an extension. This function can retrieve the list
137 of devices or the name associated with a hint.
138 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
140 * Added SYSINFO() dialplan function which allows retrieval of system information
141 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
142 the existence of a dialplan target.
143 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
144 upper and lower case, respectively.
145 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
146 ID for the call (not the Asterisk call ID or unique ID), provided that the
147 channel driver supports this. For SIP, you get the SIP call-ID for the
148 bridged channel which you can store in the CDR with a custom field.
152 * New CLI command "core show hint" (usage: core show hint <exten>)
153 * New CLI command "core show settings"
154 * Added 'core show channels count' CLI command.
155 * Added the ability to set the core debug and verbose values on a per-file basis.
156 * Added 'queue pause member' and 'queue unpause member' CLI commands
157 * Ability to set process limits ("ulimit") without restarting Asterisk
158 * Enhanced "agi debug" to print the channel name as a prefix to the debug
159 output to make debugging on busy systems much easier.
160 * New CLI commands "dialplan set extenpatternmatching true/false"
161 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
162 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
163 listed in the startup_commands section of cli.conf will get executed.
164 * Added a CLI command, "devstate change", which allows you to set custom device
165 states from the func_devstate module that provides the DEVICE_STATE() function
166 and handling of the "Custom:" devices.
170 * Improved NAT and STUN support.
171 chan_sip now can use port numbers in bindaddr, externip and externhost
172 options, as well as contact a STUN server to detect its external address
173 for the SIP socket. See sip.conf.sample, 'NAT' section.
174 * The default SIP useragent= identifier now includes the Asterisk version
175 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
176 If set, and the incoming request carries authentication info,
177 the username to match in the users list is taken from the Digest header
178 rather than from the From: field. This feature is considered experimental.
179 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
180 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
181 * The "localmask" setting was removed in version 1.2 and the reminder about it
182 being removed is now also removed.
183 * A new option "busylevel" for setting a level of calls where asterisk reports
184 a device as busy, to separate it from call-limit. This value is also added
185 to the SIP_PEER dialplan function.
186 * A new realtime family called "sipregs" is now supported to store SIP registration
187 data. If this family is defined, "sippeers" will be used for configuration and
188 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
189 registration data, as before.
190 * The SIPPEER function have new options for port address, call and pickup groups
191 * Added support for T.140 realtime text in SIP/RTP
192 * The "checkmwi" option has been removed from sip.conf, as it is no longer
193 required due to the restructuring of how MWI is handled. See the descriptions
194 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
195 for more information.
196 * Added rtpdest option to CHANNEL() dialplan function.
197 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
198 * SIP now adds a header to the CANCEL if the call was answered by another phone
199 in the same dial command, or if the new c option in dial() is used.
200 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
201 states it is not needed. For phones, however, that do require it the "registertrying" option
202 has been added so it can be enabled.
203 * A new option called "callcounter" (global/peer/user level) enables call counters needed
204 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
205 used to enable this functionality).
206 * New settings for timer T1 and timer B on a global level or per device. This makes it
207 possible to force timeout faster on non-responsive SIP servers. These settings are
208 considered advanced, so don't use them unless you have a problem.
209 * Added a dial string option to be able to set the To: header in an INVITE to any
211 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
212 the qualify frequency.
213 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
214 were not properly torn down due to network or endpoint failures during an established
216 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
217 configs/sip.conf.sample for more information on how it is used.
218 * Added a new configuration option "authfailureevents" that enables manager events when
219 a peer can't authenticate properly.
220 * Added DNS manager support to registrations for peers not referencing a peer entry.
224 * Added the trunkmaxsize configuration option to chan_iax2.
225 * Added the srvlookup option to iax.conf
226 * Added support for OSP. The token is set and retrieved through the CHANNEL()
229 XMPP Google Talk/Jingle changes
230 -------------------------------
231 * Added the bindaddr option to gtalk.conf.
235 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
236 * Proper codec support in chan_skinny.
237 * Added settings for IP and Ethernet QoS requests
241 * Added separate settings for media QoS in mgcp.conf
243 Console Channel Driver changes
244 ------------------------------
245 * Added experimental support for video send & receive to chan_oss.
246 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
249 Phone channel changes (chan_phone)
250 ----------------------------------
251 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
253 H.323 channel Changes
254 ---------------------
255 * H323 remote hold notification support added (by NOTIFY message
256 and/or H.450 supplementary service)
258 Local channel changes
259 ---------------------
260 * The device state functionality in the Local channel driver has been updated
261 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
262 to just UNKNOWN if the extension exists.
263 * Added jitterbuffer support for chan_local. This allows you to use the
264 generic jitterbuffer on incoming calls going to Asterisk applications.
265 For example, this would allow you to use a jitterbuffer for an incoming
266 SIP call to Voicemail by putting a Local channel in the middle. This
267 feature is enabled by using the 'j' option in the Dial string to the Local
268 channel in conjunction with the existing 'n' option for local channels.
269 * A 'b' option has been added which causes chan_local to return the actual channel
270 that is behind it when queried. This is useful for transfer scenarios as the
271 actual channel will be transferred, not the Local channel.
273 Zaptel channel driver (chan_zap) Changes
274 ----------------------------------------
275 * SS7 support in chan_zap (via libss7 library)
276 * In India, some carriers transmit CID via dtmf. Some code has been added
277 that will handle some situations. The cidstart=polarity_IN choice has been added for
278 those carriers that transmit CID via dtmf after a polarity change.
279 * CID matching information is now shown when doing 'dialplan show'.
280 * Added zap show version CLI command to chan_zap.
281 * Added setvar support to zapata.conf channel entries.
282 * Added two new options: mwimonitor and mwimonitornotify. These options allow
283 you to enable MWI monitoring on FXO lines. When the MWI state changes,
284 the script specified in the mwimonitornotify option is executed. An internal
285 event indicating the new state of the mailbox is also generated, so that
286 the normal MWI facilities in Asterisk work as usual.
287 * Added signalling type 'auto', which attempts to use the same signalling type
288 for a channel as configured in Zaptel. This is primarily designed for analog
289 ports, but will also work for digital ports that are configured for FXS or FXO
290 signalling types. This mode is also the default now, so if your zapata.conf
291 does not specify signalling for a channel (which is unlikely as the sample
292 configuration file has always recommended specifying it for every channel) then
293 the 'auto' mode will be used for that channel if possible.
294 * Added a 'zap set dnd' command to allow CLI control of the Do-Not-Disturb
295 state for a channel; also ensured that the DNDState Manager event is
296 emitted no matter how the DND state is set or cleared.
300 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
301 configs/unistim.conf.sample for details. This new channel driver allows
302 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
303 * Added a new channel driver, chan_console, which uses portaudio as a cross
304 platform audio interface. It was written as a channel driver that would
305 work with Mac CoreAudio, but portaudio supports a number of other audio
306 interfaces, as well. Note that this channel driver requires v19 or higher
307 of portaudio; older versions have a different API.
311 * Added the ability to specify arguments to the Dial application when using
312 the DUNDi switch in the dialplan.
313 * Added the ability to set weights for responses dynamically. This can be
314 done using a global variable or a dialplan function. Using the SHELL()
315 function would allow you to have an external script set the weight for
317 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
318 functions will allow you to initiate a DUNDi query from the dialplan,
319 find out how many results there are, and access each one.
323 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
324 functions will allow you to initiate an ENUM lookup from the dialplan,
325 and Asterisk will cache the results. ENUMRESULT can be used to access
326 the results without doing multiple DNS queries.
330 * Added the ability to customize which sound files are used for some of the
331 prompts within the Voicemail application by changing them in voicemail.conf
332 * Added the ability for the "voicemail show users" CLI command to show users
333 configured by the dynamic realtime configuration method.
334 * MWI (Message Waiting Indication) handling has been significantly
335 restructured internally to Asterisk. It is now totally event based
336 instead of polling based. The voicemail application will notify other
337 modules that have subscribed to MWI events when something in the mailbox
339 This also means that if any other entity outside of Asterisk is changing
340 the contents of mailboxes, then the voicemail application still needs to
341 poll for changes. Examples of situations that would require this option
342 are web interfaces to voicemail or an email client in the case of using
343 IMAP storage. So, two new options have been added to voicemail.conf
344 to account for this: "pollmailboxes" and "pollfreq". See the sample
345 configuration file for details.
346 * Added "tw" language support
347 * Added support for storage of greetings using an IMAP server
348 * Added ability to customize forward, reverse, stop, and pause keys for message playback
349 * SMDI is now enabled in voicemail using the smdienable option.
350 * A "lockmode" option has been added to asterisk.conf to configure the file
351 locking method used for voicemail, and potentially other things in the
352 future. The default is the old behavior, lockfile. However, there is a
353 new method, "flock", that uses a different method for situations where the
354 lockfile will not work, such as on SMB/CIFS mounts.
355 * Added the ability to backup deleted messages, to ease recovery in the case
356 that a user accidentally deletes a message, and discovers that they need it.
357 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
358 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
359 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
360 voicemail boxes. The SMDI interface can also poll for MWI changes when some
361 outside entity is modifying the state of the mailbox (such as IMAP storage or
362 a web interface of some kind).
366 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
367 used across multiple queues.
368 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
369 setqueueentryvar options for each queue, see queues.conf.sample for details.
370 * Added keepstats option to queues.conf which will keep queue
371 statistics during a reload.
372 * setinterfacevar option in queues.conf also now sets a variable
373 called MEMBERNAME which contains the member's name.
374 * Added 'Strategy' field to manager event QueueParams which represents
375 the queue strategy in use.
376 * Added option to run macro when a queue member is connected to a caller,
377 see queues.conf.sample for details.
378 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
379 does not count paused queue members as unavailable.
380 * Added min-announce-frequency option to queues.conf which allows you to control the
381 minimum amount of time between queue announcements for use when the caller's queue
382 position changes frequently.
383 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
385 * Added ability for non-realtime queues to have realtime members
386 * Added the "linear" strategy to queues.
387 * Added the "wrandom" strategy to queues.
388 * Added new channel variable QUEUE_MIN_PENALTY
389 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
390 rules in queuerules.conf. See configs/queuerules.conf.sample for details
391 * Added a new parameter for member definition, called state_interface. This may be
392 used so that a member may be called via one interface but have a different interface's
393 device state reported.
394 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
395 specified by the periodic-announce option, then one will be chosen randomly when it is time
396 to play a periodic announcment
400 * The 'o' option to provide an optimization has been removed and its functionality
401 has been enabled by default.
402 * When a conference is created, the UNIQUEID of the channel that caused it to be
403 created is stored. Then, every channel that joins the conference will have the
404 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
405 callers that come and go from long standing conferences.
406 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
407 except it does operations on a channel by name, instead of number in a conference.
408 This is a very useful feature in combination with the 'X' option to ChanSpy.
409 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
411 * Added new RealTime functionality to provide support for scheduled conferencing.
412 This includes optional messages to the caller if they attempt to join before
413 the schedule start time, or to allow the caller to join the conference early.
414 Also included is optional support for limiting the number of callers per
416 * Added the S() and L() options to the MeetMe application. These are pretty
417 much identical to the S() and L() options to Dial(). They let you set
418 timeouts for the conference, as well as have warning sounds played to
419 let the caller know how much time is left, and when it is running out.
420 * Added the ability to do "meetme concise" with the "meetme" CLI command.
421 This extends the concise capabilities of this CLI command to include
422 listing all conferences, instead of an addition to the other sub commands
423 for the "meetme" command.
424 * Added the ability to specify the music on hold class used to play into the
425 conference when there is only one member and the M option is used.
427 Other Dialplan Application Changes
428 ----------------------------------
429 * Argument support for Gosub application
430 * From the to-do lists: straighten out the app timeout args:
431 Wait() app now really does 0.3 seconds- was truncating arg to an int.
432 WaitExten() same as Wait().
433 Congestion() - Now takes floating pt. argument.
434 Busy() - now takes floating pt. argument.
435 Read() - timeout now can be floating pt.
436 WaitForRing() now takes floating pt timeout arg.
437 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
438 * Added 's' option to Page application.
439 * Added 'E' and 'V' commands to ExternalIVR.
440 * Added 'o' and 'X' options to Chanspy.
441 * Added a new dialplan application, Bridge, which allows you to bridge the
442 calling channel to any other active channel on the system.
443 * Added the ability to specify a music on hold class to play instead of ringing
444 for the SLATrunk application.
445 * The Read application no longer exits the dialplan on error. Instead, it sets
446 READSTATUS to ERROR, which you can catch and handle separately.
447 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
448 of asking for verification of each name, one at a time.
449 * Privacy() no longer uses privacy.conf, as all options are specifyable as
450 direct options to the app.
451 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
453 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
454 * The ChannelRedirect application no longer exits the dialplan if the given channel
455 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
456 or NOCHANNEL if the given channel was not found.
457 * The silencethreshold setting that was previously configurable in multiple
458 applications is now settable globally via dsp.conf.
459 * Added ability to communicate over a TCP socket instead of forking a child process for the
460 ExternalIVR application.
462 Music On Hold Changes
463 ---------------------
464 * A new option, "digit", has been added for music on hold classes in
465 musiconhold.conf. If this is set for a music on hold class, a caller
466 listening to music on hold can press this digit to switch to listening
467 to this music on hold class.
468 * Support for realtime music on hold has been added.
469 * In conjunction with the realtime music on hold, a general section has
470 been added to musiconhold.conf, its sole variable is cachertclasses. If this
471 is set, then music on hold classes found in realtime will be cached in memory.
475 * AEL upgraded to use the Gosub with Arguments instead
476 of Macro application, to hopefully reduce the problems
477 seen with the artificially low stack ceiling that
478 Macro bumps into. Macros can only call other Macros
479 to a depth of 7. Tests run using gosub, show depths
480 limited only by virtual memory. A small test demonstrated
481 recursive call depths of 100,000 without problems.
482 -- in addition to this, all apps that allowed a macro
483 to be called, as in Dial, queues, etc, are now allowing
484 a gosub call in similar fashion.
485 * AEL now generates LOCAL(argname) declarations when it
486 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
487 etc. That makes the arguments local in scope. The user
488 can define their own local variables in macros, now,
489 by saying "local myvar=someval;" or using Set() in this
490 fashion: Set(LOCAL(myvar)=someval); ("local" is now
492 * utils/conf2ael introduced. Will convert an extensions.conf
493 file into extensions.ael. Very crude and unfinished, but
494 will be improved as time goes by. Should be useful for a
495 first pass at conversion.
496 * aelparse will now read extensions.conf to see if a referenced
497 macro or context is there before issueing a warning.
499 Call Features (res_features) Changes
500 ------------------------------------
501 * Added the parkedcalltransfers option to features.conf
502 * The built-in method for doing attended transfers has been updated to
503 include some new options that allow you to have the transferee sent
504 back to the person that did the transfer if the transfer is not successful.
505 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
506 in features.conf.sample.
507 * Added support for configuring named groups of custom call features in
508 features.conf. This means that features can be written a single time, and
509 then mapped into groups of features for different key mappings or easier
511 * Updated the ParkedCall application to allow you to not specify a parking
512 extension. If you don't specify a parking space to pick up, it will grab
513 the first one available.
514 * Added cli command 'features reload' to reload call features from features.conf
515 * Moved into core asterisk binary.
517 Language Support Changes
518 ------------------------
519 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
520 * Added support for the Hungarian language for saying numbers, dates, and times.
524 * Added SPEECH commands for speech recognition. A complete listing can be found
529 * Added rotatestrategy option to logger.conf, along with two new options:
530 "timestamp" which will use the time to name the logger files instead of
531 sequence number; and "rotate", which rotates the names of the logfiles,
532 similar to the way syslog rotates files.
533 * Added exec_after_rotate option to logger.conf, which allows a system
534 command to be run after rotation. This is primarily useful with
535 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
536 and to ensure that the oldest log file gets deleted.
537 * Added realtime support for the queue log
541 * The cdr_manager module has a [mappings] feature, like cdr_custom,
542 to add fields to the manager event from the CDR variables.
543 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
544 backend database CDR table. Specifically, additional, non-standard
545 columns are supported, merely by setting the corresponding CDR variable in
546 your dialplan. In addition, you may alias any column to another name (for
547 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
548 simply "alias src => ANI" in the configuration file). Records may be
549 posted to more than one backend, simply by specifying multiple categories
550 in the configuration file. And finally, you may filter which CDRs get
551 posted to each backend, by specifying a filter (which the record must
552 match) for the particular category. Filters are additive (meaning all
553 rules must match to post that CDR).
554 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
555 module. Specifically, you may add additional columns into the table and
556 they will be set, if you set the corresponding CDR variable name. Also,
557 if you omit columns in your database table, they will be silently skipped
558 (but a record will still be inserted, based on what columns remain). Note
559 that the other two features from cdr_adaptive_odbc (alias and filter) are
560 not currently supported.
561 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
562 has been disabled using the NoCDR application.
564 Miscellaneous New Modules
565 -------------------------
566 * Added a new CDR module, cdr_sqlite3_custom.
567 * Added a new realtime configuration module, res_config_sqlite
568 * Added a new codec translation module, codec_resample, which re-samples
569 signed linear audio between 8 kHz and 16 kHz to help support wideband
571 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
572 based on configuration templates that use Asterisk dialplan function and
573 variable substitution. It should be possible to create phone profiles and
574 templates that work for the majority of phones provisioned over http. It
575 is currently only intended to provision a single user account per phone.
576 An example profile and set of templates for Polycom phones is provided.
577 NOTE: Polycom firmware is not included, but should be placed in
578 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
579 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
580 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
581 provided; there is a JACK() application, and a JACK_HOOK() function. Both
582 interfaces create an input and output JACK port. The application makes
583 these ports the endpoint of the call. The audio coming from the channel
584 goes out the output port and whatever comes back in on the input port is
585 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
586 audiohook on the channel. This lets you run the audio coming from a
587 channel through JACK, and whatever comes back in is what gets forwarded
588 on as the channel's audio. This is very useful for building custom
589 vocoders or doing recording or analysis of the channel's audio in another
591 * Added a new module, res_config_curl, which permits using a HTTP POST url
592 to retrieve, create, update, and delete realtime information from a remote
593 web server. Note that this module requires func_curl.so to be loaded for
594 backend functionality.
595 * Added a new module, res_config_ldap, which permits the use of an LDAP
596 server for realtime data access.
597 * Added support for writing and running your dialplan in lua using the pbx_lua
598 module. See configs/extensions.lua.sample for examples of how to do this.
602 * Ability to use libcap to set high ToS bits when non-root
603 on Linux. If configure is unable to find libcap then you
604 can use --with-cap to specify the path.
605 * Added maxfiles option to options section of asterisk.conf which allows you to specify
606 what Asterisk should set as the maximum number of open files when it loads.
607 * Added the jittertargetextra configuration option.
608 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
609 configuration files for the IP channel drivers. The new option is "cos".
610 This information is also documented in doc/qos.tex, or the IP Quality of Service
611 section of asterisk.pdf.
612 * When originating a call using AMI or pbx_spool that fails the reason for failure
613 will now be available in the failed extension using the REASON dialplan variable.
614 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
615 It allows you to configure a prefix for auto-monitor recordings.
616 * A new extension pattern matching algorithm, based on a trie, is introduced
617 here, that could noticeably speed up mid-sized to large dialplans.
618 It is NOT used by default, as duplicating the behaviour of the old pattern
619 matcher is still under development. A config file option, in extensions.conf,
620 in the [general] section, called "extenpatternmatchingnew", is by default
621 set to false; setting that to true will force the use of the new algorithm.
622 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
623 be used to switch the algorithms at run time.
624 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
625 specifying which socket to use to connect to the running Asterisk daemon
627 * Performance enhancements to the sched facility, which is used in
628 the channel drivers, etc. Added hashtabs and doubly-linked lists
629 to speed up deletion; start at the beginning or end of list to
631 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
632 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
633 Added regression tests to the tests/ dir, also.
634 * Added a refcount trace feature to astobj2 for those trying to balance
635 object creation, deletion; work, play; space and time. See the
636 notes in astobj2.h. Also, see utils/refcounter as well, as a
637 quick way to find unbalanced refcounts in what could be a sea
638 of objects that were balanced.
639 * Added logging to 'make update' command. See update.log
640 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
641 do not come from the remote party.
642 * Added the 'n' option to the SpeechBackground application to tell it to not
643 answer the channel if it has not already been answered.
644 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
645 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
647 * iLBC source code no longer included (see UPGRADE.txt for details)
projects / asterisk/asterisk.git / blob