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 DNS manager runs in the background which allows DNS lookups to be run asynchronously
41 as well as periodically updating the IP address. These properties allow for
42 better performance as well as recovery in the event of an IP change.
43 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
44 load/reload of large numbers of peers/users by ~40x (for large lists of peers.
45 Initially, we saw 4x improvement in call setup/destruction, but at the time
46 of merging, this gain has disappeared; further research will be done to try
47 and restore this performance improvement. Astobj2 refcounting is now used
48 for users, peers, and dialogs. Users are encouraged to assist in regression
49 testing and problem reporting!
53 * Existing DNS manager lookups extended to check for SRV records.
57 * New CLI command, "config reload <file.conf>" which reloads any module that
58 references that particular configuration file. Also added "config list"
59 which shows which configuration files are in use.
63 * Addresses managed by DNS manager now can check to see if there is a DNS
64 SRV record for a given domain and will use that hostname/port if present.
66 ------------------------------------------------------------------------------
67 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
68 ------------------------------------------------------------------------------
70 AMI - The manager (TCP/TLS/HTTP)
71 --------------------------------
72 * Manager has undergone a lot of changes, all of them documented
73 in doc/manager_1_1.txt
74 * Manager version has changed to 1.1
75 * Added a new action 'CoreShowChannels' to list currently defined channels
76 and some information about them.
77 * Added a new action 'SIPshowregistry' to list SIP registrations.
78 * Added TLS support for the manager interface and HTTP server
79 * Added the URI redirect option for the built-in HTTP server
80 * The output of CallerID in Manager events is now more consistent.
81 CallerIDNum is used for number and CallerIDName for name.
82 * Enable https support for builtin web server.
83 See configs/http.conf.sample for details.
84 * Added a new action, GetConfigJSON, which can return the contents of an
85 Asterisk configuration file in JSON format. This is intended to help
86 improve the performance of AJAX applications using the manager interface
88 * SIP and IAX manager events now use "ChannelType" in all cases where we
89 indicate channel driver. Previously, we used a mixture of "Channel"
90 and "ChannelDriver" headers.
91 * Added a "Bridge" action which allows you to bridge any two channels that
92 are currently active on the system.
93 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
94 the voicemail users setup.
95 * Added 'DBDel' and 'DBDelTree' manager commands.
96 * cdr_manager now reports events via the "cdr" level, separating it from
97 the very verbose "call" level.
98 * Manager users are now stored in memory. If you change the manager account
99 list (delete or add accounts) you need to reload manager.
100 * Added Masquerade manager event for when a masquerade happens between
102 * Added "manager reload" command for the CLI
103 * Lots of commands that only provided information are now allowed under the
104 Reporting privilege, instead of only under Call or System.
105 * The IAX* commands now require either System or Reporting privilege, to
106 mirror the privileges of the SIP* commands.
107 * Added ability to retrieve list of categories in a config file.
108 * Added ability to retrieve the content of a particular category.
109 * Added ability to empty a context.
110 * Created new action to create a new file.
111 * Updated delete action to allow deletion by line number with respect to category.
112 * Added new action insert to add new variable to category at specified line.
113 * Updated action newcat to allow new category to be inserted in file above another
115 * Added new event "JitterBufStats" in the IAX2 channel
116 * Originate now requires the Originate privilege and, if you want to call out
117 to a subshell, it requires the System privilege, as well. This was done to
118 enhance manager security.
119 * New command: Atxfer. See doc/manager_1_1.txt for more details or
120 manager show command Atxfer from the CLI
124 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
125 state in the dialplan, as well as creating custom device states that are
126 controllable from the dialplan.
127 * Extend CALLERID() function with "pres" and "ton" parameters to
128 fetch string representation of calling number presentation indicator
129 and numeric representation of type of calling number value.
130 * MailboxExists converted to dialplan function
131 * A new option to Dial() for telling IP phones not to count the call
132 as "missed" when dial times out and cancels.
133 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
134 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
135 held for any given channel. Also, locks are automatically freed when a
137 * Added HINT() dialplan function that allows retrieving hint information.
138 Hints are mappings between extensions and devices for the sake of
139 determining the state of an extension. This function can retrieve the list
140 of devices or the name associated with a hint.
141 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
143 * Added SYSINFO() dialplan function which allows retrieval of system information
144 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
145 the existence of a dialplan target.
146 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
147 upper and lower case, respectively.
148 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
149 ID for the call (not the Asterisk call ID or unique ID), provided that the
150 channel driver supports this. For SIP, you get the SIP call-ID for the
151 bridged channel which you can store in the CDR with a custom field.
155 * New CLI command "core show hint" (usage: core show hint <exten>)
156 * New CLI command "core show settings"
157 * Added 'core show channels count' CLI command.
158 * Added the ability to set the core debug and verbose values on a per-file basis.
159 * Added 'queue pause member' and 'queue unpause member' CLI commands
160 * Ability to set process limits ("ulimit") without restarting Asterisk
161 * Enhanced "agi debug" to print the channel name as a prefix to the debug
162 output to make debugging on busy systems much easier.
163 * New CLI commands "dialplan set extenpatternmatching true/false"
164 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
165 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
166 listed in the startup_commands section of cli.conf will get executed.
167 * Added a CLI command, "devstate change", which allows you to set custom device
168 states from the func_devstate module that provides the DEVICE_STATE() function
169 and handling of the "Custom:" devices.
170 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
171 sorted into the different possible callbacks, with the number of entries
172 currently scheduled for each. Gives you a feel for how busy the sip channel
177 * Improved NAT and STUN support.
178 chan_sip now can use port numbers in bindaddr, externip and externhost
179 options, as well as contact a STUN server to detect its external address
180 for the SIP socket. See sip.conf.sample, 'NAT' section.
181 * The default SIP useragent= identifier now includes the Asterisk version
182 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
183 If set, and the incoming request carries authentication info,
184 the username to match in the users list is taken from the Digest header
185 rather than from the From: field. This feature is considered experimental.
186 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
187 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
188 * The "localmask" setting was removed in version 1.2 and the reminder about it
189 being removed is now also removed.
190 * A new option "busylevel" for setting a level of calls where asterisk reports
191 a device as busy, to separate it from call-limit. This value is also added
192 to the SIP_PEER dialplan function.
193 * A new realtime family called "sipregs" is now supported to store SIP registration
194 data. If this family is defined, "sippeers" will be used for configuration and
195 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
196 registration data, as before.
197 * The SIPPEER function have new options for port address, call and pickup groups
198 * Added support for T.140 realtime text in SIP/RTP
199 * The "checkmwi" option has been removed from sip.conf, as it is no longer
200 required due to the restructuring of how MWI is handled. See the descriptions
201 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
202 for more information.
203 * Added rtpdest option to CHANNEL() dialplan function.
204 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
205 * SIP now adds a header to the CANCEL if the call was answered by another phone
206 in the same dial command, or if the new c option in dial() is used.
207 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
208 states it is not needed. For phones, however, that do require it the "registertrying" option
209 has been added so it can be enabled.
210 * A new option called "callcounter" (global/peer/user level) enables call counters needed
211 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
212 used to enable this functionality).
213 * New settings for timer T1 and timer B on a global level or per device. This makes it
214 possible to force timeout faster on non-responsive SIP servers. These settings are
215 considered advanced, so don't use them unless you have a problem.
216 * Added a dial string option to be able to set the To: header in an INVITE to any
218 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
219 the qualify frequency.
220 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
221 were not properly torn down due to network or endpoint failures during an established
223 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
224 configs/sip.conf.sample for more information on how it is used.
225 * Added a new configuration option "authfailureevents" that enables manager events when
226 a peer can't authenticate properly.
227 * Added DNS manager support to registrations for peers not referencing a peer entry.
231 * Added the trunkmaxsize configuration option to chan_iax2.
232 * Added the srvlookup option to iax.conf
233 * Added support for OSP. The token is set and retrieved through the CHANNEL()
236 XMPP Google Talk/Jingle changes
237 -------------------------------
238 * Added the bindaddr option to gtalk.conf.
242 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
243 * Proper codec support in chan_skinny.
244 * Added settings for IP and Ethernet QoS requests
248 * Added separate settings for media QoS in mgcp.conf
250 Console Channel Driver changes
251 ------------------------------
252 * Added experimental support for video send & receive to chan_oss.
253 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
256 Phone channel changes (chan_phone)
257 ----------------------------------
258 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
260 H.323 channel Changes
261 ---------------------
262 * H323 remote hold notification support added (by NOTIFY message
263 and/or H.450 supplementary service)
265 Local channel changes
266 ---------------------
267 * The device state functionality in the Local channel driver has been updated
268 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
269 to just UNKNOWN if the extension exists.
270 * Added jitterbuffer support for chan_local. This allows you to use the
271 generic jitterbuffer on incoming calls going to Asterisk applications.
272 For example, this would allow you to use a jitterbuffer for an incoming
273 SIP call to Voicemail by putting a Local channel in the middle. This
274 feature is enabled by using the 'j' option in the Dial string to the Local
275 channel in conjunction with the existing 'n' option for local channels.
276 * A 'b' option has been added which causes chan_local to return the actual channel
277 that is behind it when queried. This is useful for transfer scenarios as the
278 actual channel will be transferred, not the Local channel.
280 Zaptel channel driver (chan_zap) Changes
281 ----------------------------------------
282 * SS7 support in chan_zap (via libss7 library)
283 * In India, some carriers transmit CID via dtmf. Some code has been added
284 that will handle some situations. The cidstart=polarity_IN choice has been added for
285 those carriers that transmit CID via dtmf after a polarity change.
286 * CID matching information is now shown when doing 'dialplan show'.
287 * Added zap show version CLI command to chan_zap.
288 * Added setvar support to zapata.conf channel entries.
289 * Added two new options: mwimonitor and mwimonitornotify. These options allow
290 you to enable MWI monitoring on FXO lines. When the MWI state changes,
291 the script specified in the mwimonitornotify option is executed. An internal
292 event indicating the new state of the mailbox is also generated, so that
293 the normal MWI facilities in Asterisk work as usual.
294 * Added signalling type 'auto', which attempts to use the same signalling type
295 for a channel as configured in Zaptel. This is primarily designed for analog
296 ports, but will also work for digital ports that are configured for FXS or FXO
297 signalling types. This mode is also the default now, so if your zapata.conf
298 does not specify signalling for a channel (which is unlikely as the sample
299 configuration file has always recommended specifying it for every channel) then
300 the 'auto' mode will be used for that channel if possible.
301 * Added a 'zap set dnd' command to allow CLI control of the Do-Not-Disturb
302 state for a channel; also ensured that the DNDState Manager event is
303 emitted no matter how the DND state is set or cleared.
307 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
308 configs/unistim.conf.sample for details. This new channel driver allows
309 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
310 * Added a new channel driver, chan_console, which uses portaudio as a cross
311 platform audio interface. It was written as a channel driver that would
312 work with Mac CoreAudio, but portaudio supports a number of other audio
313 interfaces, as well. Note that this channel driver requires v19 or higher
314 of portaudio; older versions have a different API.
318 * Added the ability to specify arguments to the Dial application when using
319 the DUNDi switch in the dialplan.
320 * Added the ability to set weights for responses dynamically. This can be
321 done using a global variable or a dialplan function. Using the SHELL()
322 function would allow you to have an external script set the weight for
324 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
325 functions will allow you to initiate a DUNDi query from the dialplan,
326 find out how many results there are, and access each one.
330 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
331 functions will allow you to initiate an ENUM lookup from the dialplan,
332 and Asterisk will cache the results. ENUMRESULT can be used to access
333 the results without doing multiple DNS queries.
337 * Added the ability to customize which sound files are used for some of the
338 prompts within the Voicemail application by changing them in voicemail.conf
339 * Added the ability for the "voicemail show users" CLI command to show users
340 configured by the dynamic realtime configuration method.
341 * MWI (Message Waiting Indication) handling has been significantly
342 restructured internally to Asterisk. It is now totally event based
343 instead of polling based. The voicemail application will notify other
344 modules that have subscribed to MWI events when something in the mailbox
346 This also means that if any other entity outside of Asterisk is changing
347 the contents of mailboxes, then the voicemail application still needs to
348 poll for changes. Examples of situations that would require this option
349 are web interfaces to voicemail or an email client in the case of using
350 IMAP storage. So, two new options have been added to voicemail.conf
351 to account for this: "pollmailboxes" and "pollfreq". See the sample
352 configuration file for details.
353 * Added "tw" language support
354 * Added support for storage of greetings using an IMAP server
355 * Added ability to customize forward, reverse, stop, and pause keys for message playback
356 * SMDI is now enabled in voicemail using the smdienable option.
357 * A "lockmode" option has been added to asterisk.conf to configure the file
358 locking method used for voicemail, and potentially other things in the
359 future. The default is the old behavior, lockfile. However, there is a
360 new method, "flock", that uses a different method for situations where the
361 lockfile will not work, such as on SMB/CIFS mounts.
362 * Added the ability to backup deleted messages, to ease recovery in the case
363 that a user accidentally deletes a message, and discovers that they need it.
364 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
365 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
366 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
367 voicemail boxes. The SMDI interface can also poll for MWI changes when some
368 outside entity is modifying the state of the mailbox (such as IMAP storage or
369 a web interface of some kind).
373 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
374 used across multiple queues.
375 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
376 setqueueentryvar options for each queue, see queues.conf.sample for details.
377 * Added keepstats option to queues.conf which will keep queue
378 statistics during a reload.
379 * setinterfacevar option in queues.conf also now sets a variable
380 called MEMBERNAME which contains the member's name.
381 * Added 'Strategy' field to manager event QueueParams which represents
382 the queue strategy in use.
383 * Added option to run macro when a queue member is connected to a caller,
384 see queues.conf.sample for details.
385 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
386 does not count paused queue members as unavailable.
387 * Added min-announce-frequency option to queues.conf which allows you to control the
388 minimum amount of time between queue announcements for use when the caller's queue
389 position changes frequently.
390 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
392 * Added ability for non-realtime queues to have realtime members
393 * Added the "linear" strategy to queues.
394 * Added the "wrandom" strategy to queues.
395 * Added new channel variable QUEUE_MIN_PENALTY
396 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
397 rules in queuerules.conf. See configs/queuerules.conf.sample for details
398 * Added a new parameter for member definition, called state_interface. This may be
399 used so that a member may be called via one interface but have a different interface's
400 device state reported.
401 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
402 specified by the periodic-announce option, then one will be chosen randomly when it is time
403 to play a periodic announcment
407 * The 'o' option to provide an optimization has been removed and its functionality
408 has been enabled by default.
409 * When a conference is created, the UNIQUEID of the channel that caused it to be
410 created is stored. Then, every channel that joins the conference will have the
411 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
412 callers that come and go from long standing conferences.
413 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
414 except it does operations on a channel by name, instead of number in a conference.
415 This is a very useful feature in combination with the 'X' option to ChanSpy.
416 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
418 * Added new RealTime functionality to provide support for scheduled conferencing.
419 This includes optional messages to the caller if they attempt to join before
420 the schedule start time, or to allow the caller to join the conference early.
421 Also included is optional support for limiting the number of callers per
423 * Added the S() and L() options to the MeetMe application. These are pretty
424 much identical to the S() and L() options to Dial(). They let you set
425 timeouts for the conference, as well as have warning sounds played to
426 let the caller know how much time is left, and when it is running out.
427 * Added the ability to do "meetme concise" with the "meetme" CLI command.
428 This extends the concise capabilities of this CLI command to include
429 listing all conferences, instead of an addition to the other sub commands
430 for the "meetme" command.
431 * Added the ability to specify the music on hold class used to play into the
432 conference when there is only one member and the M option is used.
433 * Added MEETME_INFO dialplan function which provides a way to query
434 various properties of a Meetme conference.
436 Other Dialplan Application Changes
437 ----------------------------------
438 * Argument support for Gosub application
439 * From the to-do lists: straighten out the app timeout args:
440 Wait() app now really does 0.3 seconds- was truncating arg to an int.
441 WaitExten() same as Wait().
442 Congestion() - Now takes floating pt. argument.
443 Busy() - now takes floating pt. argument.
444 Read() - timeout now can be floating pt.
445 WaitForRing() now takes floating pt timeout arg.
446 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
447 * Added 's' option to Page application.
448 * Added 'E' and 'V' commands to ExternalIVR.
449 * Added 'o' and 'X' options to Chanspy.
450 * Added a new dialplan application, Bridge, which allows you to bridge the
451 calling channel to any other active channel on the system.
452 * Added the ability to specify a music on hold class to play instead of ringing
453 for the SLATrunk application.
454 * The Read application no longer exits the dialplan on error. Instead, it sets
455 READSTATUS to ERROR, which you can catch and handle separately.
456 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
457 of asking for verification of each name, one at a time.
458 * Privacy() no longer uses privacy.conf, as all options are specifyable as
459 direct options to the app.
460 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
462 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
463 * The ChannelRedirect application no longer exits the dialplan if the given channel
464 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
465 or NOCHANNEL if the given channel was not found.
466 * The silencethreshold setting that was previously configurable in multiple
467 applications is now settable globally via dsp.conf.
468 * Added ability to communicate over a TCP socket instead of forking a child process for the
469 ExternalIVR application.
471 Music On Hold Changes
472 ---------------------
473 * A new option, "digit", has been added for music on hold classes in
474 musiconhold.conf. If this is set for a music on hold class, a caller
475 listening to music on hold can press this digit to switch to listening
476 to this music on hold class.
477 * Support for realtime music on hold has been added.
478 * In conjunction with the realtime music on hold, a general section has
479 been added to musiconhold.conf, its sole variable is cachertclasses. If this
480 is set, then music on hold classes found in realtime will be cached in memory.
484 * AEL upgraded to use the Gosub with Arguments instead
485 of Macro application, to hopefully reduce the problems
486 seen with the artificially low stack ceiling that
487 Macro bumps into. Macros can only call other Macros
488 to a depth of 7. Tests run using gosub, show depths
489 limited only by virtual memory. A small test demonstrated
490 recursive call depths of 100,000 without problems.
491 -- in addition to this, all apps that allowed a macro
492 to be called, as in Dial, queues, etc, are now allowing
493 a gosub call in similar fashion.
494 * AEL now generates LOCAL(argname) declarations when it
495 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
496 etc. That makes the arguments local in scope. The user
497 can define their own local variables in macros, now,
498 by saying "local myvar=someval;" or using Set() in this
499 fashion: Set(LOCAL(myvar)=someval); ("local" is now
501 * utils/conf2ael introduced. Will convert an extensions.conf
502 file into extensions.ael. Very crude and unfinished, but
503 will be improved as time goes by. Should be useful for a
504 first pass at conversion.
505 * aelparse will now read extensions.conf to see if a referenced
506 macro or context is there before issueing a warning.
507 * AEL parser sets a local channel variable ~~EXTEN~~, to
508 preserve the value of ${EXTEN} thru switch statements.
509 * New operator in $[...] expressions: the ~~ operator serves
510 as a concatenation operator. AT THE MOMENT, it is really only
511 necessary and useful in AEL, especially in if() expressions.
512 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
513 any enclosing double-quotes, and evaluate to the value of a
514 concatenated with the value of b. For example if a is set to
515 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
519 Call Features (res_features) Changes
520 ------------------------------------
521 * Added the parkedcalltransfers option to features.conf
522 * The built-in method for doing attended transfers has been updated to
523 include some new options that allow you to have the transferee sent
524 back to the person that did the transfer if the transfer is not successful.
525 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
526 in features.conf.sample.
527 * Added support for configuring named groups of custom call features in
528 features.conf. This means that features can be written a single time, and
529 then mapped into groups of features for different key mappings or easier
531 * Updated the ParkedCall application to allow you to not specify a parking
532 extension. If you don't specify a parking space to pick up, it will grab
533 the first one available.
534 * Added cli command 'features reload' to reload call features from features.conf
535 * Moved into core asterisk binary.
537 Language Support Changes
538 ------------------------
539 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
540 * Added support for the Hungarian language for saying numbers, dates, and times.
544 * Added SPEECH commands for speech recognition. A complete listing can be found
549 * Added rotatestrategy option to logger.conf, along with two new options:
550 "timestamp" which will use the time to name the logger files instead of
551 sequence number; and "rotate", which rotates the names of the logfiles,
552 similar to the way syslog rotates files.
553 * Added exec_after_rotate option to logger.conf, which allows a system
554 command to be run after rotation. This is primarily useful with
555 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
556 and to ensure that the oldest log file gets deleted.
557 * Added realtime support for the queue log
561 * The cdr_manager module has a [mappings] feature, like cdr_custom,
562 to add fields to the manager event from the CDR variables.
563 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
564 backend database CDR table. Specifically, additional, non-standard
565 columns are supported, merely by setting the corresponding CDR variable in
566 your dialplan. In addition, you may alias any column to another name (for
567 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
568 simply "alias src => ANI" in the configuration file). Records may be
569 posted to more than one backend, simply by specifying multiple categories
570 in the configuration file. And finally, you may filter which CDRs get
571 posted to each backend, by specifying a filter (which the record must
572 match) for the particular category. Filters are additive (meaning all
573 rules must match to post that CDR).
574 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
575 module. Specifically, you may add additional columns into the table and
576 they will be set, if you set the corresponding CDR variable name. Also,
577 if you omit columns in your database table, they will be silently skipped
578 (but a record will still be inserted, based on what columns remain). Note
579 that the other two features from cdr_adaptive_odbc (alias and filter) are
580 not currently supported.
581 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
582 has been disabled using the NoCDR application.
584 Miscellaneous New Modules
585 -------------------------
586 * Added a new CDR module, cdr_sqlite3_custom.
587 * Added a new realtime configuration module, res_config_sqlite
588 * Added a new codec translation module, codec_resample, which re-samples
589 signed linear audio between 8 kHz and 16 kHz to help support wideband
591 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
592 based on configuration templates that use Asterisk dialplan function and
593 variable substitution. It should be possible to create phone profiles and
594 templates that work for the majority of phones provisioned over http. It
595 is currently only intended to provision a single user account per phone.
596 An example profile and set of templates for Polycom phones is provided.
597 NOTE: Polycom firmware is not included, but should be placed in
598 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
599 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
600 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
601 provided; there is a JACK() application, and a JACK_HOOK() function. Both
602 interfaces create an input and output JACK port. The application makes
603 these ports the endpoint of the call. The audio coming from the channel
604 goes out the output port and whatever comes back in on the input port is
605 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
606 audiohook on the channel. This lets you run the audio coming from a
607 channel through JACK, and whatever comes back in is what gets forwarded
608 on as the channel's audio. This is very useful for building custom
609 vocoders or doing recording or analysis of the channel's audio in another
611 * Added a new module, res_config_curl, which permits using a HTTP POST url
612 to retrieve, create, update, and delete realtime information from a remote
613 web server. Note that this module requires func_curl.so to be loaded for
614 backend functionality.
615 * Added a new module, res_config_ldap, which permits the use of an LDAP
616 server for realtime data access.
617 * Added support for writing and running your dialplan in lua using the pbx_lua
618 module. See configs/extensions.lua.sample for examples of how to do this.
622 * Ability to use libcap to set high ToS bits when non-root
623 on Linux. If configure is unable to find libcap then you
624 can use --with-cap to specify the path.
625 * Added maxfiles option to options section of asterisk.conf which allows you to specify
626 what Asterisk should set as the maximum number of open files when it loads.
627 * Added the jittertargetextra configuration option.
628 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
629 configuration files for the IP channel drivers. The new option is "cos".
630 This information is also documented in doc/qos.tex, or the IP Quality of Service
631 section of asterisk.pdf.
632 * When originating a call using AMI or pbx_spool that fails the reason for failure
633 will now be available in the failed extension using the REASON dialplan variable.
634 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
635 It allows you to configure a prefix for auto-monitor recordings.
636 * A new extension pattern matching algorithm, based on a trie, is introduced
637 here, that could noticeably speed up mid-sized to large dialplans.
638 It is NOT used by default, as duplicating the behaviour of the old pattern
639 matcher is still under development. A config file option, in extensions.conf,
640 in the [general] section, called "extenpatternmatchingnew", is by default
641 set to false; setting that to true will force the use of the new algorithm.
642 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
643 be used to switch the algorithms at run time.
644 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
645 specifying which socket to use to connect to the running Asterisk daemon
647 * Performance enhancements to the sched facility, which is used in
648 the channel drivers, etc. Added hashtabs and doubly-linked lists
649 to speed up deletion; start at the beginning or end of list to
651 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
652 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
653 Added regression tests to the tests/ dir, also.
654 * Added a refcount trace feature to astobj2 for those trying to balance
655 object creation, deletion; work, play; space and time. See the
656 notes in astobj2.h. Also, see utils/refcounter as well, as a
657 quick way to find unbalanced refcounts in what could be a sea
658 of objects that were balanced.
659 * Added logging to 'make update' command. See update.log
660 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
661 do not come from the remote party.
662 * Added the 'n' option to the SpeechBackground application to tell it to not
663 answer the channel if it has not already been answered.
664 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
665 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
667 * iLBC source code no longer included (see UPGRADE.txt for details)
projects / asterisk/asterisk.git / blob