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1 ------------------------------------------------------------------------------
2 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
3 ------------------------------------------------------------------------------
5 Dialplan Functions
6 ------------------
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.
15 PBX Changes
16 -----------
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.
21 ------------------------------------------------------------------------------
22 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
23 ------------------------------------------------------------------------------
25 AMI - The manager (TCP/TLS/HTTP)
26 --------------------------------
27 * Manager has undergone a lot of changes, all of them documented
28 in doc/manager_1_1.txt
29 * Manager version has changed to 1.1
30 * Added a new action 'CoreShowChannels' to list currently defined channels
31 and some information about them.
32 * Added a new action 'SIPshowregistry' to list SIP registrations.
33 * Added TLS support for the manager interface and HTTP server
34 * Added the URI redirect option for the built-in HTTP server
35 * The output of CallerID in Manager events is now more consistent.
36 CallerIDNum is used for number and CallerIDName for name.
37 * Enable https support for builtin web server.
38 See configs/http.conf.sample for details.
39 * Added a new action, GetConfigJSON, which can return the contents of an
40 Asterisk configuration file in JSON format. This is intended to help
41 improve the performance of AJAX applications using the manager interface
42 over HTTP.
43 * SIP and IAX manager events now use "ChannelType" in all cases where we
44 indicate channel driver. Previously, we used a mixture of "Channel"
45 and "ChannelDriver" headers.
46 * Added a "Bridge" action which allows you to bridge any two channels that
47 are currently active on the system.
48 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
49 the voicemail users setup.
50 * Added 'DBDel' and 'DBDelTree' manager commands.
51 * cdr_manager now reports events via the "cdr" level, separating it from
52 the very verbose "call" level.
53 * Manager users are now stored in memory. If you change the manager account
54 list (delete or add accounts) you need to reload manager.
55 * Added Masquerade manager event for when a masquerade happens between
56 two channels.
57 * Added "manager reload" command for the CLI
58 * Lots of commands that only provided information are now allowed under the
59 Reporting privilege, instead of only under Call or System.
60 * The IAX* commands now require either System or Reporting privilege, to
61 mirror the privileges of the SIP* commands.
62 * Added ability to retrieve list of categories in a config file.
63 * Added ability to retrieve the content of a particular category.
64 * Added ability to empty a context.
65 * Created new action to create a new file.
66 * Updated delete action to allow deletion by line number with respect to category.
67 * Added new action insert to add new variable to category at specified line.
68 * Updated action newcat to allow new category to be inserted in file above another
69 existing category.
70 * Added new event "JitterBufStats" in the IAX2 channel
71 * Originate now requires the Originate privilege and, if you want to call out
72 to a subshell, it requires the System privilege, as well. This was done to
73 enhance manager security.
74 * New command: Atxfer. See doc/manager_1_1.txt for more details or
75 manager show command Atxfer from the CLI
77 Dialplan functions
78 ------------------
79 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
80 state in the dialplan, as well as creating custom device states that are
81 controllable from the dialplan.
82 * Extend CALLERID() function with "pres" and "ton" parameters to
83 fetch string representation of calling number presentation indicator
84 and numeric representation of type of calling number value.
85 * MailboxExists converted to dialplan function
86 * A new option to Dial() for telling IP phones not to count the call
87 as "missed" when dial times out and cancels.
88 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
89 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
90 held for any given channel. Also, locks are automatically freed when a
91 channel is hung up.
92 * Added HINT() dialplan function that allows retrieving hint information.
93 Hints are mappings between extensions and devices for the sake of
94 determining the state of an extension. This function can retrieve the list
95 of devices or the name associated with a hint.
96 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
97 of any extension.
98 * Added SYSINFO() dialplan function which allows retrieval of system information
99 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
100 the existence of a dialplan target.
101 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
102 upper and lower case, respectively.
103 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
104 ID for the call (not the Asterisk call ID or unique ID), provided that the
105 channel driver supports this. For SIP, you get the SIP call-ID for the
106 bridged channel which you can store in the CDR with a custom field.
108 CLI Changes
109 -----------
110 * New CLI command "core show hint" (usage: core show hint <exten>)
111 * New CLI command "core show settings"
112 * Added 'core show channels count' CLI command.
113 * Added the ability to set the core debug and verbose values on a per-file basis.
114 * Added 'queue pause member' and 'queue unpause member' CLI commands
115 * Ability to set process limits ("ulimit") without restarting Asterisk
116 * Enhanced "agi debug" to print the channel name as a prefix to the debug
117 output to make debugging on busy systems much easier.
118 * New CLI commands "dialplan set extenpatternmatching true/false"
119 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
120 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
121 listed in the startup_commands section of cli.conf will get executed.
122 * Added a CLI command, "devstate change", which allows you to set custom device
123 states from the func_devstate module that provides the DEVICE_STATE() function
124 and handling of the "Custom:" devices.
126 SIP changes
127 -----------
128 * Improved NAT and STUN support.
129 chan_sip now can use port numbers in bindaddr, externip and externhost
130 options, as well as contact a STUN server to detect its external address
131 for the SIP socket. See sip.conf.sample, 'NAT' section.
132 * The default SIP useragent= identifier now includes the Asterisk version
133 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
134 If set, and the incoming request carries authentication info,
135 the username to match in the users list is taken from the Digest header
136 rather than from the From: field. This feature is considered experimental.
137 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
138 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
139 * The "localmask" setting was removed in version 1.2 and the reminder about it
140 being removed is now also removed.
141 * A new option "busylevel" for setting a level of calls where asterisk reports
142 a device as busy, to separate it from call-limit. This value is also added
143 to the SIP_PEER dialplan function.
144 * A new realtime family called "sipregs" is now supported to store SIP registration
145 data. If this family is defined, "sippeers" will be used for configuration and
146 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
147 registration data, as before.
148 * The SIPPEER function have new options for port address, call and pickup groups
149 * Added support for T.140 realtime text in SIP/RTP
150 * The "checkmwi" option has been removed from sip.conf, as it is no longer
151 required due to the restructuring of how MWI is handled. See the descriptions
152 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
153 for more information.
154 * Added rtpdest option to CHANNEL() dialplan function.
155 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
156 * SIP now adds a header to the CANCEL if the call was answered by another phone
157 in the same dial command, or if the new c option in dial() is used.
158 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
159 states it is not needed. For phones, however, that do require it the "registertrying" option
160 has been added so it can be enabled.
161 * A new option called "callcounter" (global/peer/user level) enables call counters needed
162 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
163 used to enable this functionality).
164 * New settings for timer T1 and timer B on a global level or per device. This makes it
165 possible to force timeout faster on non-responsive SIP servers. These settings are
166 considered advanced, so don't use them unless you have a problem.
167 * Added a dial string option to be able to set the To: header in an INVITE to any
168 SIP uri.
169 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
170 the qualify frequency.
171 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
172 were not properly torn down due to network or endpoint failures during an established
173 SIP session.
174 * Added TCP and TLS support for SIP. See doc/siptls.txt and configs/sip.conf.sample for
175 more information on how it is used.
176 * Added a new configuration option "authfailureevents" that enables manager events when
177 a peer can't authenticate properly.
179 IAX2 changes
180 ------------
181 * Added the trunkmaxsize configuration option to chan_iax2.
182 * Added the srvlookup option to iax.conf
183 * Added support for OSP. The token is set and retrieved through the CHANNEL()
184 dialplan function.
186 XMPP Google Talk/Jingle changes
187 -------------------------------
188 * Added the bindaddr option to gtalk.conf.
190 Skinny changes
191 -------------
192 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
193 * Proper codec support in chan_skinny.
194 * Added settings for IP and Ethernet QoS requests
196 MGCP changes
197 ------------
198 * Added separate settings for media QoS in mgcp.conf
200 Console Channel Driver changes
201 ------------------------------
202 * Added experimental support for video send & receive to chan_oss.
203 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
204 a video source.
206 Phone channel changes (chan_phone)
207 ----------------------------------
208 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
210 H.323 channel Changes
211 ---------------------
212 * H323 remote hold notification support added (by NOTIFY message
213 and/or H.450 supplementary service)
215 Local channel changes
216 ---------------------
217 * The device state functionality in the Local channel driver has been updated
218 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
219 to just UNKNOWN if the extension exists.
220 * Added jitterbuffer support for chan_local. This allows you to use the
221 generic jitterbuffer on incoming calls going to Asterisk applications.
222 For example, this would allow you to use a jitterbuffer for an incoming
223 SIP call to Voicemail by putting a Local channel in the middle. This
224 feature is enabled by using the 'j' option in the Dial string to the Local
225 channel in conjunction with the existing 'n' option for local channels.
227 Zaptel channel driver (chan_zap) Changes
228 ----------------------------------------
229 * SS7 support in chan_zap (via libss7 library)
230 * In India, some carriers transmit CID via dtmf. Some code has been added
231 that will handle some situations. The cidstart=polarity_IN choice has been added for
232 those carriers that transmit CID via dtmf after a polarity change.
233 * CID matching information is now shown when doing 'dialplan show'.
234 * Added zap show version CLI command to chan_zap.
235 * Added setvar support to zapata.conf channel entries.
236 * Added two new options: mwimonitor and mwimonitornotify. These options allow
237 you to enable MWI monitoring on FXO lines. When the MWI state changes,
238 the script specified in the mwimonitornotify option is executed. An internal
239 event indicating the new state of the mailbox is also generated, so that
240 the normal MWI facilities in Asterisk work as usual.
241 * Added signalling type 'auto', which attempts to use the same signalling type
242 for a channel as configured in Zaptel. This is primarily designed for analog
243 ports, but will also work for digital ports that are configured for FXS or FXO
244 signalling types. This mode is also the default now, so if your zapata.conf
245 does not specify signalling for a channel (which is unlikely as the sample
246 configuration file has always recommended specifying it for every channel) then
247 the 'auto' mode will be used for that channel if possible.
248 * Added a 'zap set dnd' command to allow CLI control of the Do-Not-Disturb
249 state for a channel; also ensured that the DNDState Manager event is
250 emitted no matter how the DND state is set or cleared.
252 New Channel Drivers
253 -------------------
254 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
255 configs/unistim.conf.sample for details. This new channel driver allows
256 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
257 * Added a new channel driver, chan_console, which uses portaudio as a cross
258 platform audio interface. It was written as a channel driver that would
259 work with Mac CoreAudio, but portaudio supports a number of other audio
260 interfaces, as well. Note that this channel driver requires v19 or higher
261 of portaudio; older versions have a different API.
263 DUNDi changes
264 -------------
265 * Added the ability to specify arguments to the Dial application when using
266 the DUNDi switch in the dialplan.
267 * Added the ability to set weights for responses dynamically. This can be
268 done using a global variable or a dialplan function. Using the SHELL()
269 function would allow you to have an external script set the weight for
270 each response.
271 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
272 functions will allow you to initiate a DUNDi query from the dialplan,
273 find out how many results there are, and access each one.
275 ENUM changes
276 ------------
277 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
278 functions will allow you to initiate an ENUM lookup from the dialplan,
279 and Asterisk will cache the results. ENUMRESULT can be used to access
280 the results without doing multiple DNS queries.
282 Voicemail Changes
283 -----------------
284 * Added the ability to customize which sound files are used for some of the
285 prompts within the Voicemail application by changing them in voicemail.conf
286 * Added the ability for the "voicemail show users" CLI command to show users
287 configured by the dynamic realtime configuration method.
288 * MWI (Message Waiting Indication) handling has been significantly
289 restructured internally to Asterisk. It is now totally event based
290 instead of polling based. The voicemail application will notify other
291 modules that have subscribed to MWI events when something in the mailbox
292 changes.
293 This also means that if any other entity outside of Asterisk is changing
294 the contents of mailboxes, then the voicemail application still needs to
295 poll for changes. Examples of situations that would require this option
296 are web interfaces to voicemail or an email client in the case of using
297 IMAP storage. So, two new options have been added to voicemail.conf
298 to account for this: "pollmailboxes" and "pollfreq". See the sample
299 configuration file for details.
300 * Added "tw" language support
301 * Added support for storage of greetings using an IMAP server
302 * Added ability to customize forward, reverse, stop, and pause keys for message playback
303 * SMDI is now enabled in voicemail using the smdienable option.
304 * A "lockmode" option has been added to asterisk.conf to configure the file
305 locking method used for voicemail, and potentially other things in the
306 future. The default is the old behavior, lockfile. However, there is a
307 new method, "flock", that uses a different method for situations where the
308 lockfile will not work, such as on SMB/CIFS mounts.
309 * Added the ability to backup deleted messages, to ease recovery in the case
310 that a user accidentally deletes a message, and discovers that they need it.
311 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
312 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
313 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
314 voicemail boxes. The SMDI interface can also poll for MWI changes when some
315 outside entity is modifying the state of the mailbox (such as IMAP storage or
316 a web interface of some kind).
318 Queue changes
319 -------------
320 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
321 used across multiple queues.
322 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
323 setqueueentryvar options for each queue, see queues.conf.sample for details.
324 * Added keepstats option to queues.conf which will keep queue
325 statistics during a reload.
326 * setinterfacevar option in queues.conf also now sets a variable
327 called MEMBERNAME which contains the member's name.
328 * Added 'Strategy' field to manager event QueueParams which represents
329 the queue strategy in use.
330 * Added option to run macro when a queue member is connected to a caller,
331 see queues.conf.sample for details.
332 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
333 does not count paused queue members as unavailable.
334 * Added min-announce-frequency option to queues.conf which allows you to control the
335 minimum amount of time between queue announcements for use when the caller's queue
336 position changes frequently.
337 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
338 queue log.
339 * Added ability for non-realtime queues to have realtime members
340 * Added the "linear" strategy to queues.
341 * Added the "wrandom" strategy to queues.
342 * Added new channel variable QUEUE_MIN_PENALTY
343 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
344 rules in queuerules.conf. See configs/queuerules.conf.sample for details
345 * Added a new parameter for member definition, called state_interface. This may be
346 used so that a member may be called via one interface but have a different interface's
347 device state reported.
348 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
349 specified by the periodic-announce option, then one will be chosen randomly when it is time
350 to play a periodic announcment
352 MeetMe Changes
353 --------------
354 * The 'o' option to provide an optimization has been removed and its functionality
355 has been enabled by default.
356 * When a conference is created, the UNIQUEID of the channel that caused it to be
357 created is stored. Then, every channel that joins the conference will have the
358 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
359 callers that come and go from long standing conferences.
360 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
361 except it does operations on a channel by name, instead of number in a conference.
362 This is a very useful feature in combination with the 'X' option to ChanSpy.
363 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
364 when kicked out.
365 * Added new RealTime functionality to provide support for scheduled conferencing.
366 This includes optional messages to the caller if they attempt to join before
367 the schedule start time, or to allow the caller to join the conference early.
368 Also included is optional support for limiting the number of callers per
369 RealTime conference.
370 * Added the S() and L() options to the MeetMe application. These are pretty
371 much identical to the S() and L() options to Dial(). They let you set
372 timeouts for the conference, as well as have warning sounds played to
373 let the caller know how much time is left, and when it is running out.
374 * Added the ability to do "meetme concise" with the "meetme" CLI command.
375 This extends the concise capabilities of this CLI command to include
376 listing all conferences, instead of an addition to the other sub commands
377 for the "meetme" command.
378 * Added the ability to specify the music on hold class used to play into the
379 conference when there is only one member and the M option is used.
381 Other Dialplan Application Changes
382 ----------------------------------
383 * Argument support for Gosub application
384 * From the to-do lists: straighten out the app timeout args:
385 Wait() app now really does 0.3 seconds- was truncating arg to an int.
386 WaitExten() same as Wait().
387 Congestion() - Now takes floating pt. argument.
388 Busy() - now takes floating pt. argument.
389 Read() - timeout now can be floating pt.
390 WaitForRing() now takes floating pt timeout arg.
391 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
392 * Added 's' option to Page application.
393 * Added 'E' and 'V' commands to ExternalIVR.
394 * Added 'o' and 'X' options to Chanspy.
395 * Added a new dialplan application, Bridge, which allows you to bridge the
396 calling channel to any other active channel on the system.
397 * Added the ability to specify a music on hold class to play instead of ringing
398 for the SLATrunk application.
399 * The Read application no longer exits the dialplan on error. Instead, it sets
400 READSTATUS to ERROR, which you can catch and handle separately.
401 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
402 of asking for verification of each name, one at a time.
403 * Privacy() no longer uses privacy.conf, as all options are specifyable as
404 direct options to the app.
405 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
406 for more details
407 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
408 * The ChannelRedirect application no longer exits the dialplan if the given channel
409 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
410 or NOCHANNEL if the given channel was not found.
411 * The silencethreshold setting that was previously configurable in multiple
412 applications is now settable globally via dsp.conf.
413 * Added ability to communicate over a TCP socket instead of forking a child process for the
414 ExternalIVR application.
416 Music On Hold Changes
417 ---------------------
418 * A new option, "digit", has been added for music on hold classes in
419 musiconhold.conf. If this is set for a music on hold class, a caller
420 listening to music on hold can press this digit to switch to listening
421 to this music on hold class.
422 * Support for realtime music on hold has been added.
423 * In conjunction with the realtime music on hold, a general section has
424 been added to musiconhold.conf, its sole variable is cachertclasses. If this
425 is set, then music on hold classes found in realtime will be cached in memory.
427 AEL Changes
428 -----------
429 * AEL upgraded to use the Gosub with Arguments instead
430 of Macro application, to hopefully reduce the problems
431 seen with the artificially low stack ceiling that
432 Macro bumps into. Macros can only call other Macros
433 to a depth of 7. Tests run using gosub, show depths
434 limited only by virtual memory. A small test demonstrated
435 recursive call depths of 100,000 without problems.
436 -- in addition to this, all apps that allowed a macro
437 to be called, as in Dial, queues, etc, are now allowing
438 a gosub call in similar fashion.
439 * AEL now generates LOCAL(argname) declarations when it
440 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
441 etc. That makes the arguments local in scope. The user
442 can define their own local variables in macros, now,
443 by saying "local myvar=someval;" or using Set() in this
444 fashion: Set(LOCAL(myvar)=someval); ("local" is now
445 an AEL keyword).
446 * utils/conf2ael introduced. Will convert an extensions.conf
447 file into extensions.ael. Very crude and unfinished, but
448 will be improved as time goes by. Should be useful for a
449 first pass at conversion.
450 * aelparse will now read extensions.conf to see if a referenced
451 macro or context is there before issueing a warning.
453 Call Features (res_features) Changes
454 ------------------------------------
455 * Added the parkedcalltransfers option to features.conf
456 * The built-in method for doing attended transfers has been updated to
457 include some new options that allow you to have the transferee sent
458 back to the person that did the transfer if the transfer is not successful.
459 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
460 in features.conf.sample.
461 * Added support for configuring named groups of custom call features in
462 features.conf. This means that features can be written a single time, and
463 then mapped into groups of features for different key mappings or easier
464 access control.
465 * Updated the ParkedCall application to allow you to not specify a parking
466 extension. If you don't specify a parking space to pick up, it will grab
467 the first one available.
468 * Added cli command 'features reload' to reload call features from features.conf
469 * Moved into core asterisk binary.
471 Language Support Changes
472 ------------------------
473 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
474 * Added support for the Hungarian language for saying numbers, dates, and times.
476 AGI Changes
477 -----------
478 * Added SPEECH commands for speech recognition. A complete listing can be found
479 using agi show.
481 Logger changes
482 --------------
483 * Added rotatestrategy option to logger.conf, along with two new options:
484 "timestamp" which will use the time to name the logger files instead of
485 sequence number; and "rotate", which rotates the names of the logfiles,
486 similar to the way syslog rotates files.
487 * Added exec_after_rotate option to logger.conf, which allows a system
488 command to be run after rotation. This is primarily useful with
489 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
490 and to ensure that the oldest log file gets deleted.
491 * Added realtime support for the queue log
493 Call Detail Records
494 -------------------
495 * The cdr_manager module has a [mappings] feature, like cdr_custom,
496 to add fields to the manager event from the CDR variables.
497 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
498 backend database CDR table. Specifically, additional, non-standard
499 columns are supported, merely by setting the corresponding CDR variable in
500 your dialplan. In addition, you may alias any column to another name (for
501 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
502 simply "alias src => ANI" in the configuration file). Records may be
503 posted to more than one backend, simply by specifying multiple categories
504 in the configuration file. And finally, you may filter which CDRs get
505 posted to each backend, by specifying a filter (which the record must
506 match) for the particular category. Filters are additive (meaning all
507 rules must match to post that CDR).
508 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
509 module. Specifically, you may add additional columns into the table and
510 they will be set, if you set the corresponding CDR variable name. Also,
511 if you omit columns in your database table, they will be silently skipped
512 (but a record will still be inserted, based on what columns remain). Note
513 that the other two features from cdr_adaptive_odbc (alias and filter) are
514 not currently supported.
515 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
516 has been disabled using the NoCDR application.
518 Miscellaneous New Modules
519 -------------------------
520 * Added a new CDR module, cdr_sqlite3_custom.
521 * Added a new realtime configuration module, res_config_sqlite
522 * Added a new codec translation module, codec_resample, which re-samples
523 signed linear audio between 8 kHz and 16 kHz to help support wideband
524 codecs.
525 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
526 based on configuration templates that use Asterisk dialplan function and
527 variable substitution. It should be possible to create phone profiles and
528 templates that work for the majority of phones provisioned over http. It
529 is currently only intended to provision a single user account per phone.
530 An example profile and set of templates for Polycom phones is provided.
531 NOTE: Polycom firmware is not included, but should be placed in
532 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
533 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
534 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
535 provided; there is a JACK() application, and a JACK_HOOK() function. Both
536 interfaces create an input and output JACK port. The application makes
537 these ports the endpoint of the call. The audio coming from the channel
538 goes out the output port and whatever comes back in on the input port is
539 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
540 audiohook on the channel. This lets you run the audio coming from a
541 channel through JACK, and whatever comes back in is what gets forwarded
542 on as the channel's audio. This is very useful for building custom
543 vocoders or doing recording or analysis of the channel's audio in another
544 application.
545 * Added a new module, res_config_curl, which permits using a HTTP POST url
546 to retrieve, create, update, and delete realtime information from a remote
547 web server. Note that this module requires func_curl.so to be loaded for
548 backend functionality.
549 * Added a new module, res_config_ldap, which permits the use of an LDAP
550 server for realtime data access.
551 * Added support for writing and running your dialplan in lua using the pbx_lua
552 module. See configs/extensions.lua.sample for examples of how to do this.
554 Miscellaneous
555 -------------
556 * Ability to use libcap to set high ToS bits when non-root
557 on Linux. If configure is unable to find libcap then you
558 can use --with-cap to specify the path.
559 * Added maxfiles option to options section of asterisk.conf which allows you to specify
560 what Asterisk should set as the maximum number of open files when it loads.
561 * Added the jittertargetextra configuration option.
562 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
563 configuration files for the IP channel drivers. The new option is "cos".
564 This information is also documented in doc/qos.tex, or the IP Quality of Service
565 section of asterisk.pdf.
566 * When originating a call using AMI or pbx_spool that fails the reason for failure
567 will now be available in the failed extension using the REASON dialplan variable.
568 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
569 It allows you to configure a prefix for auto-monitor recordings.
570 * A new extension pattern matching algorithm, based on a trie, is introduced
571 here, that could noticeably speed up mid-sized to large dialplans.
572 It is NOT used by default, as duplicating the behaviour of the old pattern
573 matcher is still under development. A config file option, in extensions.conf,
574 in the [general] section, called "extenpatternmatchingnew", is by default
575 set to false; setting that to true will force the use of the new algorithm.
576 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
577 be used to switch the algorithms at run time.
578 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
579 specifying which socket to use to connect to the running Asterisk daemon
580 (-s)
581 * Added logging to 'make update' command. See update.log
582 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
583 do not come from the remote party.
584 * Added the 'n' option to the SpeechBackground application to tell it to not
585 answer the channel if it has not already been answered.
586 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
587 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
588 dialplan debugging.