1 ------------------------------------------------------------------------------
2 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
3 ------------------------------------------------------------------------------
7 * Added a new dialplan function, AST_CONFIG(), which allows you to access
8 variables from an Asterisk configuration file.
9 * The JACK_HOOK function now has a c() option to supply a custom client name.
10 * Added two new dialplan functions from libspeex for audio gain control and
11 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
12 rx directions of a channel from the dialplan.
13 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
14 based on other parameters. The default is still to search based on the
15 forwarding station ID. However, there are new options that allow you to search
16 based on the message desk terminal ID, or the message desk number.
18 Zaptel channel driver (chan_zap) Changes
19 ----------------------------------------
20 * Channels can now be configured using named sections in zapata.conf, just
21 like other channel drivers, including the use of templates.
22 * The default for pridialplan has changed from 'national' to 'unknown'.
26 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
27 to something that matches the pattern a hint will be created using the contents
28 and variables evaluated.
29 * Dialplan matching has been extended to allow an extension to return to the
30 PBX core to wait for more digits. This is done by using the new dialplan
31 application called "Incomplete". This will permit a whole new level of
32 extension control, by giving the administrator more control over early
33 matches employing one of the short-circuit pattern match operators. Note
34 that custom applications can trigger this same behavior by returning the
35 special value AST_PBX_INCOMPLETE.
39 * Directory now permits both first and last names to be matched at the same
40 time. In addition, the number of digits to enter of the name can be set in
41 the arguments to Directory; previously, you could enter only 3, regardless
42 of how many names are in your company. For large companies, this should be
44 * Voicemail now permits a mailbox setting to wrap around from first to last
45 messages, if the "messagewrap" option is set to a true value.
46 * Voicemail now permits an external script to be run, for password validation.
47 The script should output "VALID" or "INVALID" on stdout, depending upon the
48 wish to validate or invalidate the password given. Arguments are:
49 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
51 * Dial has a new option: F(context^extension^pri), which permits a callee to
52 continue in the dialplan, at the specified label, if the caller hangs up.
53 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
54 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
55 * The Jack application now has a c() option to supply a custom client name.
56 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
57 like the pre-existing whisper mode, except that the spy can also talk to the
58 participant on the bridged channel as well.
59 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
60 to be spoken instead of the channel name or number. For more information on the
61 use of this option, issue the command "core show application ChanSpy" from the
63 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
64 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
65 words, if using the 'd' option, it is not possible to enter a number to append to
66 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
67 change to whisper mode, and pressing 6 will change to barge mode.
71 * The ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using setvar to cause a given
72 audio file to be played upon completion of an attended transfer.
73 * Added DNS manager support to registrations for peers referencing peer entries.
74 DNS manager runs in the background which allows DNS lookups to be run asynchronously
75 as well as periodically updating the IP address. These properties allow for
76 better performance as well as recovery in the event of an IP change.
77 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
78 load/reload of large numbers of peers/users by ~40x (for large lists of peers.
79 Initially, we saw 4x improvement in call setup/destruction, but at the time
80 of merging, this gain has disappeared; further research will be done to try
81 and restore this performance improvement. Astobj2 refcounting is now used
82 for users, peers, and dialogs. Users are encouraged to assist in regression
83 testing and problem reporting!
84 * Added ability to specify registration expiry time on a per registration basis in
86 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
91 * Existing DNS manager lookups extended to check for SRV records.
95 * New CLI command, "config reload <file.conf>" which reloads any module that
96 references that particular configuration file. Also added "config list"
97 which shows which configuration files are in use.
98 * New CLI commands, "pri show version" and "ss7 show version" that will
99 display which version of libpri and libss7 are being used, respectively.
100 A new API call was added so trunk will now have to be compiled against
101 a versions of libpri and libss7 that have them or it will not know that
102 these libraries exist.
106 * Addresses managed by DNS manager now can check to see if there is a DNS
107 SRV record for a given domain and will use that hostname/port if present.
109 Dialplan function changes
110 -------------------------
111 * TIMEOUT() has been modified to be accurate down to the millisecond.
112 * ENUM*() functions now include the following new options:
113 - 'u' returns the full URI and does not strip off the URI-scheme.
114 - 's' triggers ISN specific rewriting
115 - 'i' looks for branches into an Infrastructure ENUM tree
116 - 'd' for a direct DNS lookup without any flipping of digits.
117 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
119 AMI - The manager (TCP/TLS/HTTP)
120 --------------------------------
121 * The Status command now takes an optional list of variables to display
122 along with channel status.
124 ------------------------------------------------------------------------------
125 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
126 ------------------------------------------------------------------------------
128 AMI - The manager (TCP/TLS/HTTP)
129 --------------------------------
130 * Manager has undergone a lot of changes, all of them documented
131 in doc/manager_1_1.txt
132 * Manager version has changed to 1.1
133 * Added a new action 'CoreShowChannels' to list currently defined channels
134 and some information about them.
135 * Added a new action 'SIPshowregistry' to list SIP registrations.
136 * Added TLS support for the manager interface and HTTP server
137 * Added the URI redirect option for the built-in HTTP server
138 * The output of CallerID in Manager events is now more consistent.
139 CallerIDNum is used for number and CallerIDName for name.
140 * Enable https support for builtin web server.
141 See configs/http.conf.sample for details.
142 * Added a new action, GetConfigJSON, which can return the contents of an
143 Asterisk configuration file in JSON format. This is intended to help
144 improve the performance of AJAX applications using the manager interface
146 * SIP and IAX manager events now use "ChannelType" in all cases where we
147 indicate channel driver. Previously, we used a mixture of "Channel"
148 and "ChannelDriver" headers.
149 * Added a "Bridge" action which allows you to bridge any two channels that
150 are currently active on the system.
151 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
152 the voicemail users setup.
153 * Added 'DBDel' and 'DBDelTree' manager commands.
154 * cdr_manager now reports events via the "cdr" level, separating it from
155 the very verbose "call" level.
156 * Manager users are now stored in memory. If you change the manager account
157 list (delete or add accounts) you need to reload manager.
158 * Added Masquerade manager event for when a masquerade happens between
160 * Added "manager reload" command for the CLI
161 * Lots of commands that only provided information are now allowed under the
162 Reporting privilege, instead of only under Call or System.
163 * The IAX* commands now require either System or Reporting privilege, to
164 mirror the privileges of the SIP* commands.
165 * Added ability to retrieve list of categories in a config file.
166 * Added ability to retrieve the content of a particular category.
167 * Added ability to empty a context.
168 * Created new action to create a new file.
169 * Updated delete action to allow deletion by line number with respect to category.
170 * Added new action insert to add new variable to category at specified line.
171 * Updated action newcat to allow new category to be inserted in file above another
173 * Added new event "JitterBufStats" in the IAX2 channel
174 * Originate now requires the Originate privilege and, if you want to call out
175 to a subshell, it requires the System privilege, as well. This was done to
176 enhance manager security.
177 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
178 * New command: Atxfer. See doc/manager_1_1.txt for more details or
179 manager show command Atxfer from the CLI
183 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
184 state in the dialplan, as well as creating custom device states that are
185 controllable from the dialplan.
186 * Extend CALLERID() function with "pres" and "ton" parameters to
187 fetch string representation of calling number presentation indicator
188 and numeric representation of type of calling number value.
189 * MailboxExists converted to dialplan function
190 * A new option to Dial() for telling IP phones not to count the call
191 as "missed" when dial times out and cancels.
192 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
193 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
194 held for any given channel. Also, locks are automatically freed when a
196 * Added HINT() dialplan function that allows retrieving hint information.
197 Hints are mappings between extensions and devices for the sake of
198 determining the state of an extension. This function can retrieve the list
199 of devices or the name associated with a hint.
200 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
202 * Added SYSINFO() dialplan function which allows retrieval of system information
203 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
204 the existence of a dialplan target.
205 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
206 upper and lower case, respectively.
207 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
208 ID for the call (not the Asterisk call ID or unique ID), provided that the
209 channel driver supports this. For SIP, you get the SIP call-ID for the
210 bridged channel which you can store in the CDR with a custom field.
214 * New CLI command "core show hint" (usage: core show hint <exten>)
215 * New CLI command "core show settings"
216 * Added 'core show channels count' CLI command.
217 * Added the ability to set the core debug and verbose values on a per-file basis.
218 * Added 'queue pause member' and 'queue unpause member' CLI commands
219 * Ability to set process limits ("ulimit") without restarting Asterisk
220 * Enhanced "agi debug" to print the channel name as a prefix to the debug
221 output to make debugging on busy systems much easier.
222 * New CLI commands "dialplan set extenpatternmatching true/false"
223 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
224 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
225 listed in the startup_commands section of cli.conf will get executed.
226 * Added a CLI command, "devstate change", which allows you to set custom device
227 states from the func_devstate module that provides the DEVICE_STATE() function
228 and handling of the "Custom:" devices.
229 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
230 sorted into the different possible callbacks, with the number of entries
231 currently scheduled for each. Gives you a feel for how busy the sip channel
236 * Improved NAT and STUN support.
237 chan_sip now can use port numbers in bindaddr, externip and externhost
238 options, as well as contact a STUN server to detect its external address
239 for the SIP socket. See sip.conf.sample, 'NAT' section.
240 * The default SIP useragent= identifier now includes the Asterisk version
241 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
242 If set, and the incoming request carries authentication info,
243 the username to match in the users list is taken from the Digest header
244 rather than from the From: field. This feature is considered experimental.
245 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
246 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
247 * The "localmask" setting was removed in version 1.2 and the reminder about it
248 being removed is now also removed.
249 * A new option "busylevel" for setting a level of calls where asterisk reports
250 a device as busy, to separate it from call-limit. This value is also added
251 to the SIP_PEER dialplan function.
252 * A new realtime family called "sipregs" is now supported to store SIP registration
253 data. If this family is defined, "sippeers" will be used for configuration and
254 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
255 registration data, as before.
256 * The SIPPEER function have new options for port address, call and pickup groups
257 * Added support for T.140 realtime text in SIP/RTP
258 * The "checkmwi" option has been removed from sip.conf, as it is no longer
259 required due to the restructuring of how MWI is handled. See the descriptions
260 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
261 for more information.
262 * Added rtpdest option to CHANNEL() dialplan function.
263 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
264 * SIP now adds a header to the CANCEL if the call was answered by another phone
265 in the same dial command, or if the new c option in dial() is used.
266 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
267 states it is not needed. For phones, however, that do require it the "registertrying" option
268 has been added so it can be enabled.
269 * A new option called "callcounter" (global/peer/user level) enables call counters needed
270 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
271 used to enable this functionality).
272 * New settings for timer T1 and timer B on a global level or per device. This makes it
273 possible to force timeout faster on non-responsive SIP servers. These settings are
274 considered advanced, so don't use them unless you have a problem.
275 * Added a dial string option to be able to set the To: header in an INVITE to any
277 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
278 the qualify frequency.
279 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
280 were not properly torn down due to network or endpoint failures during an established
282 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
283 configs/sip.conf.sample for more information on how it is used.
284 * Added a new configuration option "authfailureevents" that enables manager events when
285 a peer can't authenticate properly.
286 * Added DNS manager support to registrations for peers not referencing a peer entry.
290 * Added the trunkmaxsize configuration option to chan_iax2.
291 * Added the srvlookup option to iax.conf
292 * Added support for OSP. The token is set and retrieved through the CHANNEL()
295 XMPP Google Talk/Jingle changes
296 -------------------------------
297 * Added the bindaddr option to gtalk.conf.
301 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
302 * Proper codec support in chan_skinny.
303 * Added settings for IP and Ethernet QoS requests
307 * Added separate settings for media QoS in mgcp.conf
309 Console Channel Driver changes
310 ------------------------------
311 * Added experimental support for video send & receive to chan_oss.
312 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
315 Phone channel changes (chan_phone)
316 ----------------------------------
317 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
319 H.323 channel Changes
320 ---------------------
321 * H323 remote hold notification support added (by NOTIFY message
322 and/or H.450 supplementary service)
324 Local channel changes
325 ---------------------
326 * The device state functionality in the Local channel driver has been updated
327 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
328 to just UNKNOWN if the extension exists.
329 * Added jitterbuffer support for chan_local. This allows you to use the
330 generic jitterbuffer on incoming calls going to Asterisk applications.
331 For example, this would allow you to use a jitterbuffer for an incoming
332 SIP call to Voicemail by putting a Local channel in the middle. This
333 feature is enabled by using the 'j' option in the Dial string to the Local
334 channel in conjunction with the existing 'n' option for local channels.
335 * A 'b' option has been added which causes chan_local to return the actual channel
336 that is behind it when queried. This is useful for transfer scenarios as the
337 actual channel will be transferred, not the Local channel.
339 Zaptel channel driver (chan_zap) Changes
340 ----------------------------------------
341 * SS7 support in chan_zap (via libss7 library)
342 * In India, some carriers transmit CID via dtmf. Some code has been added
343 that will handle some situations. The cidstart=polarity_IN choice has been added for
344 those carriers that transmit CID via dtmf after a polarity change.
345 * CID matching information is now shown when doing 'dialplan show'.
346 * Added zap show version CLI command to chan_zap.
347 * Added setvar support to zapata.conf channel entries.
348 * Added two new options: mwimonitor and mwimonitornotify. These options allow
349 you to enable MWI monitoring on FXO lines. When the MWI state changes,
350 the script specified in the mwimonitornotify option is executed. An internal
351 event indicating the new state of the mailbox is also generated, so that
352 the normal MWI facilities in Asterisk work as usual.
353 * Added signalling type 'auto', which attempts to use the same signalling type
354 for a channel as configured in Zaptel. This is primarily designed for analog
355 ports, but will also work for digital ports that are configured for FXS or FXO
356 signalling types. This mode is also the default now, so if your zapata.conf
357 does not specify signalling for a channel (which is unlikely as the sample
358 configuration file has always recommended specifying it for every channel) then
359 the 'auto' mode will be used for that channel if possible.
360 * Added a 'zap set dnd' command to allow CLI control of the Do-Not-Disturb
361 state for a channel; also ensured that the DNDState Manager event is
362 emitted no matter how the DND state is set or cleared.
366 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
367 configs/unistim.conf.sample for details. This new channel driver allows
368 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
369 * Added a new channel driver, chan_console, which uses portaudio as a cross
370 platform audio interface. It was written as a channel driver that would
371 work with Mac CoreAudio, but portaudio supports a number of other audio
372 interfaces, as well. Note that this channel driver requires v19 or higher
373 of portaudio; older versions have a different API.
377 * Added the ability to specify arguments to the Dial application when using
378 the DUNDi switch in the dialplan.
379 * Added the ability to set weights for responses dynamically. This can be
380 done using a global variable or a dialplan function. Using the SHELL()
381 function would allow you to have an external script set the weight for
383 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
384 functions will allow you to initiate a DUNDi query from the dialplan,
385 find out how many results there are, and access each one.
389 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
390 functions will allow you to initiate an ENUM lookup from the dialplan,
391 and Asterisk will cache the results. ENUMRESULT can be used to access
392 the results without doing multiple DNS queries.
396 * Added the ability to customize which sound files are used for some of the
397 prompts within the Voicemail application by changing them in voicemail.conf
398 * Added the ability for the "voicemail show users" CLI command to show users
399 configured by the dynamic realtime configuration method.
400 * MWI (Message Waiting Indication) handling has been significantly
401 restructured internally to Asterisk. It is now totally event based
402 instead of polling based. The voicemail application will notify other
403 modules that have subscribed to MWI events when something in the mailbox
405 This also means that if any other entity outside of Asterisk is changing
406 the contents of mailboxes, then the voicemail application still needs to
407 poll for changes. Examples of situations that would require this option
408 are web interfaces to voicemail or an email client in the case of using
409 IMAP storage. So, two new options have been added to voicemail.conf
410 to account for this: "pollmailboxes" and "pollfreq". See the sample
411 configuration file for details.
412 * Added "tw" language support
413 * Added support for storage of greetings using an IMAP server
414 * Added ability to customize forward, reverse, stop, and pause keys for message playback
415 * SMDI is now enabled in voicemail using the smdienable option.
416 * A "lockmode" option has been added to asterisk.conf to configure the file
417 locking method used for voicemail, and potentially other things in the
418 future. The default is the old behavior, lockfile. However, there is a
419 new method, "flock", that uses a different method for situations where the
420 lockfile will not work, such as on SMB/CIFS mounts.
421 * Added the ability to backup deleted messages, to ease recovery in the case
422 that a user accidentally deletes a message, and discovers that they need it.
423 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
424 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
425 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
426 voicemail boxes. The SMDI interface can also poll for MWI changes when some
427 outside entity is modifying the state of the mailbox (such as IMAP storage or
428 a web interface of some kind).
429 * Added the support for marking messages as "urgent." There are two methods to accomplish
430 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
431 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
432 the message as urgent after he has recorded a voicemail by following the voice instructions.
433 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
438 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
439 used across multiple queues.
440 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
441 setqueueentryvar options for each queue, see queues.conf.sample for details.
442 * Added keepstats option to queues.conf which will keep queue
443 statistics during a reload.
444 * setinterfacevar option in queues.conf also now sets a variable
445 called MEMBERNAME which contains the member's name.
446 * Added 'Strategy' field to manager event QueueParams which represents
447 the queue strategy in use.
448 * Added option to run macro when a queue member is connected to a caller,
449 see queues.conf.sample for details.
450 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
451 does not count paused queue members as unavailable.
452 * Added min-announce-frequency option to queues.conf which allows you to control the
453 minimum amount of time between queue announcements for use when the caller's queue
454 position changes frequently.
455 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
457 * Added ability for non-realtime queues to have realtime members
458 * Added the "linear" strategy to queues.
459 * Added the "wrandom" strategy to queues.
460 * Added new channel variable QUEUE_MIN_PENALTY
461 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
462 rules in queuerules.conf. See configs/queuerules.conf.sample for details
463 * Added a new parameter for member definition, called state_interface. This may be
464 used so that a member may be called via one interface but have a different interface's
465 device state reported.
466 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
467 specified by the periodic-announce option, then one will be chosen randomly when it is time
468 to play a periodic announcment
469 * New configuration options: announce-position now takes two more values in addition to "yes" and
470 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
471 announce-position-limit. By setting announce-position to "limit" callers will only have their
472 position announced if their position is less than what is specified by announce-position-limit.
473 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
474 will be told that their are more than announce-position-limit callers waiting.
478 * The 'o' option to provide an optimization has been removed and its functionality
479 has been enabled by default.
480 * When a conference is created, the UNIQUEID of the channel that caused it to be
481 created is stored. Then, every channel that joins the conference will have the
482 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
483 callers that come and go from long standing conferences.
484 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
485 except it does operations on a channel by name, instead of number in a conference.
486 This is a very useful feature in combination with the 'X' option to ChanSpy.
487 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
489 * Added new RealTime functionality to provide support for scheduled conferencing.
490 This includes optional messages to the caller if they attempt to join before
491 the schedule start time, or to allow the caller to join the conference early.
492 Also included is optional support for limiting the number of callers per
494 * Added the S() and L() options to the MeetMe application. These are pretty
495 much identical to the S() and L() options to Dial(). They let you set
496 timeouts for the conference, as well as have warning sounds played to
497 let the caller know how much time is left, and when it is running out.
498 * Added the ability to do "meetme concise" with the "meetme" CLI command.
499 This extends the concise capabilities of this CLI command to include
500 listing all conferences, instead of an addition to the other sub commands
501 for the "meetme" command.
502 * Added the ability to specify the music on hold class used to play into the
503 conference when there is only one member and the M option is used.
504 * Added MEETME_INFO dialplan function which provides a way to query
505 various properties of a Meetme conference.
507 Other Dialplan Application Changes
508 ----------------------------------
509 * Argument support for Gosub application
510 * From the to-do lists: straighten out the app timeout args:
511 Wait() app now really does 0.3 seconds- was truncating arg to an int.
512 WaitExten() same as Wait().
513 Congestion() - Now takes floating pt. argument.
514 Busy() - now takes floating pt. argument.
515 Read() - timeout now can be floating pt.
516 WaitForRing() now takes floating pt timeout arg.
517 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
518 * Added 's' option to Page application.
519 * Added 'E' and 'V' commands to ExternalIVR.
520 * Added 'o' and 'X' options to Chanspy.
521 * Added a new dialplan application, Bridge, which allows you to bridge the
522 calling channel to any other active channel on the system.
523 * Added the ability to specify a music on hold class to play instead of ringing
524 for the SLATrunk application.
525 * The Read application no longer exits the dialplan on error. Instead, it sets
526 READSTATUS to ERROR, which you can catch and handle separately.
527 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
528 of asking for verification of each name, one at a time.
529 * Privacy() no longer uses privacy.conf, as all options are specifyable as
530 direct options to the app.
531 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
533 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
534 * The ChannelRedirect application no longer exits the dialplan if the given channel
535 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
536 or NOCHANNEL if the given channel was not found.
537 * The silencethreshold setting that was previously configurable in multiple
538 applications is now settable globally via dsp.conf.
539 * Added ability to communicate over a TCP socket instead of forking a child process for the
540 ExternalIVR application.
542 Music On Hold Changes
543 ---------------------
544 * A new option, "digit", has been added for music on hold classes in
545 musiconhold.conf. If this is set for a music on hold class, a caller
546 listening to music on hold can press this digit to switch to listening
547 to this music on hold class.
548 * Support for realtime music on hold has been added.
549 * In conjunction with the realtime music on hold, a general section has
550 been added to musiconhold.conf, its sole variable is cachertclasses. If this
551 is set, then music on hold classes found in realtime will be cached in memory.
555 * AEL upgraded to use the Gosub with Arguments instead
556 of Macro application, to hopefully reduce the problems
557 seen with the artificially low stack ceiling that
558 Macro bumps into. Macros can only call other Macros
559 to a depth of 7. Tests run using gosub, show depths
560 limited only by virtual memory. A small test demonstrated
561 recursive call depths of 100,000 without problems.
562 -- in addition to this, all apps that allowed a macro
563 to be called, as in Dial, queues, etc, are now allowing
564 a gosub call in similar fashion.
565 * AEL now generates LOCAL(argname) declarations when it
566 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
567 etc. That makes the arguments local in scope. The user
568 can define their own local variables in macros, now,
569 by saying "local myvar=someval;" or using Set() in this
570 fashion: Set(LOCAL(myvar)=someval); ("local" is now
572 * utils/conf2ael introduced. Will convert an extensions.conf
573 file into extensions.ael. Very crude and unfinished, but
574 will be improved as time goes by. Should be useful for a
575 first pass at conversion.
576 * aelparse will now read extensions.conf to see if a referenced
577 macro or context is there before issueing a warning.
578 * AEL parser sets a local channel variable ~~EXTEN~~, to
579 preserve the value of ${EXTEN} thru switch statements.
580 * New operator in $[...] expressions: the ~~ operator serves
581 as a concatenation operator. AT THE MOMENT, it is really only
582 necessary and useful in AEL, especially in if() expressions.
583 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
584 any enclosing double-quotes, and evaluate to the value of a
585 concatenated with the value of b. For example if a is set to
586 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
590 Call Features (res_features) Changes
591 ------------------------------------
592 * Added the parkedcalltransfers option to features.conf
593 * The built-in method for doing attended transfers has been updated to
594 include some new options that allow you to have the transferee sent
595 back to the person that did the transfer if the transfer is not successful.
596 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
597 in features.conf.sample.
598 * Added support for configuring named groups of custom call features in
599 features.conf. This means that features can be written a single time, and
600 then mapped into groups of features for different key mappings or easier
602 * Updated the ParkedCall application to allow you to not specify a parking
603 extension. If you don't specify a parking space to pick up, it will grab
604 the first one available.
605 * Added cli command 'features reload' to reload call features from features.conf
606 * Moved into core asterisk binary.
608 Language Support Changes
609 ------------------------
610 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
611 * Added support for the Hungarian language for saying numbers, dates, and times.
615 * Added SPEECH commands for speech recognition. A complete listing can be found
620 * Added rotatestrategy option to logger.conf, along with two new options:
621 "timestamp" which will use the time to name the logger files instead of
622 sequence number; and "rotate", which rotates the names of the logfiles,
623 similar to the way syslog rotates files.
624 * Added exec_after_rotate option to logger.conf, which allows a system
625 command to be run after rotation. This is primarily useful with
626 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
627 and to ensure that the oldest log file gets deleted.
628 * Added realtime support for the queue log
632 * The cdr_manager module has a [mappings] feature, like cdr_custom,
633 to add fields to the manager event from the CDR variables.
634 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
635 backend database CDR table. Specifically, additional, non-standard
636 columns are supported, merely by setting the corresponding CDR variable in
637 your dialplan. In addition, you may alias any column to another name (for
638 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
639 simply "alias src => ANI" in the configuration file). Records may be
640 posted to more than one backend, simply by specifying multiple categories
641 in the configuration file. And finally, you may filter which CDRs get
642 posted to each backend, by specifying a filter (which the record must
643 match) for the particular category. Filters are additive (meaning all
644 rules must match to post that CDR).
645 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
646 module. Specifically, you may add additional columns into the table and
647 they will be set, if you set the corresponding CDR variable name. Also,
648 if you omit columns in your database table, they will be silently skipped
649 (but a record will still be inserted, based on what columns remain). Note
650 that the other two features from cdr_adaptive_odbc (alias and filter) are
651 not currently supported.
652 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
653 has been disabled using the NoCDR application.
655 Miscellaneous New Modules
656 -------------------------
657 * Added a new CDR module, cdr_sqlite3_custom.
658 * Added a new realtime configuration module, res_config_sqlite
659 * Added a new codec translation module, codec_resample, which re-samples
660 signed linear audio between 8 kHz and 16 kHz to help support wideband
662 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
663 based on configuration templates that use Asterisk dialplan function and
664 variable substitution. It should be possible to create phone profiles and
665 templates that work for the majority of phones provisioned over http. It
666 is currently only intended to provision a single user account per phone.
667 An example profile and set of templates for Polycom phones is provided.
668 NOTE: Polycom firmware is not included, but should be placed in
669 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
670 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
671 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
672 provided; there is a JACK() application, and a JACK_HOOK() function. Both
673 interfaces create an input and output JACK port. The application makes
674 these ports the endpoint of the call. The audio coming from the channel
675 goes out the output port and whatever comes back in on the input port is
676 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
677 audiohook on the channel. This lets you run the audio coming from a
678 channel through JACK, and whatever comes back in is what gets forwarded
679 on as the channel's audio. This is very useful for building custom
680 vocoders or doing recording or analysis of the channel's audio in another
682 * Added a new module, res_config_curl, which permits using a HTTP POST url
683 to retrieve, create, update, and delete realtime information from a remote
684 web server. Note that this module requires func_curl.so to be loaded for
685 backend functionality.
686 * Added a new module, res_config_ldap, which permits the use of an LDAP
687 server for realtime data access.
688 * Added support for writing and running your dialplan in lua using the pbx_lua
689 module. See configs/extensions.lua.sample for examples of how to do this.
693 * Ability to use libcap to set high ToS bits when non-root
694 on Linux. If configure is unable to find libcap then you
695 can use --with-cap to specify the path.
696 * Added maxfiles option to options section of asterisk.conf which allows you to specify
697 what Asterisk should set as the maximum number of open files when it loads.
698 * Added the jittertargetextra configuration option.
699 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
700 configuration files for the IP channel drivers. The new option is "cos".
701 This information is also documented in doc/qos.tex, or the IP Quality of Service
702 section of asterisk.pdf.
703 * When originating a call using AMI or pbx_spool that fails the reason for failure
704 will now be available in the failed extension using the REASON dialplan variable.
705 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
706 It allows you to configure a prefix for auto-monitor recordings.
707 * A new extension pattern matching algorithm, based on a trie, is introduced
708 here, that could noticeably speed up mid-sized to large dialplans.
709 It is NOT used by default, as duplicating the behaviour of the old pattern
710 matcher is still under development. A config file option, in extensions.conf,
711 in the [general] section, called "extenpatternmatchingnew", is by default
712 set to false; setting that to true will force the use of the new algorithm.
713 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
714 be used to switch the algorithms at run time.
715 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
716 specifying which socket to use to connect to the running Asterisk daemon
718 * Performance enhancements to the sched facility, which is used in
719 the channel drivers, etc. Added hashtabs and doubly-linked lists
720 to speed up deletion; start at the beginning or end of list to
722 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
723 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
724 Added regression tests to the tests/ dir, also.
725 * Added a refcount trace feature to astobj2 for those trying to balance
726 object creation, deletion; work, play; space and time. See the
727 notes in astobj2.h. Also, see utils/refcounter as well, as a
728 quick way to find unbalanced refcounts in what could be a sea
729 of objects that were balanced.
730 * Added logging to 'make update' command. See update.log
731 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
732 do not come from the remote party.
733 * Added the 'n' option to the SpeechBackground application to tell it to not
734 answer the channel if it has not already been answered.
735 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
736 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
738 * iLBC source code no longer included (see UPGRADE.txt for details)
projects / asterisk/asterisk.git / blob