1 ------------------------------------------------------------------------------
2 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
3 ------------------------------------------------------------------------------
7 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
8 Snom phones use this for call pickup of extensions that the phone is
10 * Added support for subscribing to a voice mailbox on a remote server and
11 making the new/old message count available to local devices.
12 * Added support for setting the domain in the URI for caller of an
13 outbound call by using the SIPFROMDOMAIN channel variable.
17 * The configuration file now holds seperate sections for devices and lines.
18 Please have a look at configs/skinny.conf.sample and change your skinny.conf
23 * Added a new dialplan function, CURLOPT, which permits setting various
24 options that may be useful with the CURL dialplan function, such as
25 cookies, proxies, connection timeouts, passwords, etc.
26 * Permit the syntax and synopsis fields of the corresponding dialplan
27 functions to be individually set from func_odbc.conf.
28 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
29 * func_odbc now may specify an insert query to execute, when the write query
30 affects 0 rows (usually indicating that no such row exists).
34 * Scheduled meetme conferences may now have their end times extended by
36 * app_authenticate now gives the ability to select a prompt other than
38 * app_directory now pays attention to the searchcontexts setting in
39 voicemail.conf and will look through all contexts, if no context is
40 specified in the initial argument.
44 * res_jabber: autoprune has been disabled by default, to avoid misconfiguration
45 that would end up being interpreted as a bug once Asterisk started removing
46 the contacts from a user list.
47 * extensions.conf now allows you to use keyword "same" to define an extension
48 without actually specifying an extension. It uses exactly the same pattern
49 as previously used on the last "exten" line. For example:
50 exten => 123,1,NoOp(something)
51 same => n,SomethingElse()
52 * musiconhold.conf classes of type 'files' can now use relative directory paths,
53 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
55 ------------------------------------------------------------------------------
56 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
57 ------------------------------------------------------------------------------
61 * The event infrastructure in Asterisk got another big update to help support
62 distributed events. It currently supports distributed device state and
63 distributed Voicemail MWI (Message Waiting Indication). A new module has
64 been merged, res_ais, which facilitates communicating events between servers.
65 It uses the SAForum AIS (Service Availability Forum Application Interface
66 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
67 a cluster of Asterisk servers, and to share events between them. For more
68 information on setting this up, see doc/distributed_devstate.txt.
72 * Added a new dialplan function, AST_CONFIG(), which allows you to access
73 variables from an Asterisk configuration file.
74 * The JACK_HOOK function now has a c() option to supply a custom client name.
75 * Added two new dialplan functions from libspeex for audio gain control and
76 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
77 rx directions of a channel from the dialplan.
78 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
79 based on other parameters. The default is still to search based on the
80 forwarding station ID. However, there are new options that allow you to search
81 based on the message desk terminal ID, or the message desk number.
82 * TIMEOUT() has been modified to be accurate down to the millisecond.
83 * ENUM*() functions now include the following new options:
84 - 'u' returns the full URI and does not strip off the URI-scheme.
85 - 's' triggers ISN specific rewriting
86 - 'i' looks for branches into an Infrastructure ENUM tree
87 - 'd' for a direct DNS lookup without any flipping of digits.
88 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
89 * CHANNEL() now has options for the maximum, minimum, and standard or normal
90 deviation of jitter, rtt, and loss for a call using chan_sip.
92 DAHDI channel driver (chan_dahdi) Changes
93 ----------------------------------------
94 * Channels can now be configured using named sections in chan_dahdi.conf, just
95 like other channel drivers, including the use of templates.
96 * The default for pridialplan has changed from 'national' to 'unknown'.
100 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
101 to something that matches the pattern a hint will be created using the contents
102 and variables evaluated.
103 * Dialplan matching has been extended to allow an extension to return to the
104 PBX core to wait for more digits. This is done by using the new dialplan
105 application called "Incomplete". This will permit a whole new level of
106 extension control, by giving the administrator more control over early
107 matches employing one of the short-circuit pattern match operators. Note
108 that custom applications can trigger this same behavior by returning the
109 special value AST_PBX_INCOMPLETE.
113 * Directory now permits both first and last names to be matched at the same
114 time. In addition, the number of digits to enter of the name can be set in
115 the arguments to Directory; previously, you could enter only 3, regardless
116 of how many names are in your company. For large companies, this should be
118 * Voicemail now permits a mailbox setting to wrap around from first to last
119 messages, if the "messagewrap" option is set to a true value.
120 * Voicemail now permits an external script to be run, for password validation.
121 The script should output "VALID" or "INVALID" on stdout, depending upon the
122 wish to validate or invalidate the password given. Arguments are:
123 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
125 * Dial has a new option: F(context^extension^pri), which permits a callee to
126 continue in the dialplan, at the specified label, if the caller hangs up.
127 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
128 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
129 * The Jack application now has a c() option to supply a custom client name.
130 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
131 like the pre-existing whisper mode, except that the spy can also talk to the
132 participant on the bridged channel as well.
133 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
134 to be spoken instead of the channel name or number. For more information on the
135 use of this option, issue the command "core show application ChanSpy" from the
137 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
138 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
139 words, if using the 'd' option, it is not possible to enter a number to append to
140 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
141 change to whisper mode, and pressing 6 will change to barge mode.
142 * ExternalIVR now takes several options that affect the way it performs, as
143 well as having several new commands. Please see doc/externalivr.txt for the
144 complete documentation.
145 * Added ability to communicate over a TCP socket instead of forking a child process for the
146 ExternalIVR application.
147 * ChanIsAvail has a new option, 'a', which will return all available channels instead
148 of just the first one if you give the function more then one channel to check.
149 * PrivacyManager now takes an option where you can specify a context where the
150 given number will be matched. This way you have more control over who is allowed
151 and it stops the people who blindly enter 10 digits.
152 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
153 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
154 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
155 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
156 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
157 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
158 * The Dial() application no longer copies the language used by the caller to the callee's
159 channel. If you desire for the caller's channel's language to be used for file playback
160 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
161 * SendImage() no longer hangs up the channel on error; instead, it sets the
162 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
163 'UNSUPPORTED'. This change makes SendImage() more consistent with other
165 * Park has a new option, 's', which silences the announcement of the parking space number.
166 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
167 invalid input and will be assumed to mean that no timeout is desired.
171 * Added DNS manager support to registrations for peers referencing peer entries.
172 DNS manager runs in the background which allows DNS lookups to be run asynchronously
173 as well as periodically updating the IP address. These properties allow for
174 better performance as well as recovery in the event of an IP change.
175 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
176 load/reload of large numbers of peers/users by ~40x (for large lists of peers.
177 Initially, we saw 4x improvement in call setup/destruction, but at the time
178 of merging, this gain has disappeared; further research will be done to try
179 and restore this performance improvement. Astobj2 refcounting is now used
180 for users, peers, and dialogs. Users are encouraged to assist in regression
181 testing and problem reporting!
182 * Added ability to specify registration expiry time on a per registration basis in
184 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
186 * Added t38pt_usertpsource option. See sip.conf.sample for details.
187 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
188 * 'sip show peers' and 'sip show users' display their entries sorted in
189 alphabetical order, as opposed to the order they were in, in the config
191 * Videosupport now supports an additional option, "always", which always sets
192 up video RTP ports, even on clients that don't support it. This helps with
193 callfiles and certain transfers to ensure that if two video phones are
194 connected, they will always share video feeds.
198 * Existing DNS manager lookups extended to check for SRV records.
199 * IAX2 encryption support has been improved to support periodic key rotation
200 within a call for enhanced security. The option "keyrotate" has been
201 provided to disable this functionality to preserve backwards compatibility
202 with older versions of IAX2 that do not support key rotation.
206 * New CLI command, "config reload <file.conf>" which reloads any module that
207 references that particular configuration file. Also added "config list"
208 which shows which configuration files are in use.
209 * New CLI commands, "pri show version" and "ss7 show version" that will
210 display which version of libpri and libss7 are being used, respectively.
211 A new API call was added so trunk will now have to be compiled against
212 a versions of libpri and libss7 that have them or it will not know that
213 these libraries exist.
214 * The commands "core show globals", "core set global" and "core set chanvar" has
215 been deprecated in favor of the more semanticly correct "dialplan show globals",
216 "dialplan set chanvar" and "dialplan set global".
217 * New CLI command "dialplan show chanvar" to list all variables associated
218 with a given channel.
222 * Addresses managed by DNS manager now can check to see if there is a DNS
223 SRV record for a given domain and will use that hostname/port if present.
225 AMI - The manager (TCP/TLS/HTTP)
226 --------------------------------
227 * The Status command now takes an optional list of variables to display
228 along with channel status.
229 * The QueueEntry event now also includes the channel's uniqueid
233 * res_odbc no longer has a limit of 1023 total possible unshared connections,
234 as some people were running into this limit. This limit has been increased
239 * The TRANSFER queue log entry now includes the the caller's original
240 position in the transferred-from queue.
241 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
242 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
243 as well as an explanation about timeout options in general
247 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
248 adaptive capabilities. What this means in practical terms is that if your
249 realtime table lacks critical fields, Asterisk will now emit warnings to
250 that effect. Also, some of the realtime drivers have the ability (if
251 configured) to automatically add those columns to the table with the
252 correct type and length.
256 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
257 the 'setvar' option to cause a given audio file to be played upon completion
258 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
259 Skinny channels only.
260 * You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt
261 for more information.
262 * Config file variables may now be appended to, by using the '+=' append
263 operator. This is most helpful when working with long SQL queries in
264 func_odbc.conf, as the queries no longer need to be specified on a single
266 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
267 which will add a second to the billsec when the ending
268 time is set, if the number in the microseconds field of the end time is
269 greater than the number of microseconds in the answer time. This allows
270 users to count the 'initiated' seconds in their billing records.
272 ------------------------------------------------------------------------------
273 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
274 ------------------------------------------------------------------------------
276 AMI - The manager (TCP/TLS/HTTP)
277 --------------------------------
278 * Manager has undergone a lot of changes, all of them documented
279 in doc/manager_1_1.txt
280 * Manager version has changed to 1.1
281 * Added a new action 'CoreShowChannels' to list currently defined channels
282 and some information about them.
283 * Added a new action 'SIPshowregistry' to list SIP registrations.
284 * Added TLS support for the manager interface and HTTP server
285 * Added the URI redirect option for the built-in HTTP server
286 * The output of CallerID in Manager events is now more consistent.
287 CallerIDNum is used for number and CallerIDName for name.
288 * Enable https support for builtin web server.
289 See configs/http.conf.sample for details.
290 * Added a new action, GetConfigJSON, which can return the contents of an
291 Asterisk configuration file in JSON format. This is intended to help
292 improve the performance of AJAX applications using the manager interface
294 * SIP and IAX manager events now use "ChannelType" in all cases where we
295 indicate channel driver. Previously, we used a mixture of "Channel"
296 and "ChannelDriver" headers.
297 * Added a "Bridge" action which allows you to bridge any two channels that
298 are currently active on the system.
299 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
300 the voicemail users setup.
301 * Added 'DBDel' and 'DBDelTree' manager commands.
302 * cdr_manager now reports events via the "cdr" level, separating it from
303 the very verbose "call" level.
304 * Manager users are now stored in memory. If you change the manager account
305 list (delete or add accounts) you need to reload manager.
306 * Added Masquerade manager event for when a masquerade happens between
308 * Added "manager reload" command for the CLI
309 * Lots of commands that only provided information are now allowed under the
310 Reporting privilege, instead of only under Call or System.
311 * The IAX* commands now require either System or Reporting privilege, to
312 mirror the privileges of the SIP* commands.
313 * Added ability to retrieve list of categories in a config file.
314 * Added ability to retrieve the content of a particular category.
315 * Added ability to empty a context.
316 * Created new action to create a new file.
317 * Updated delete action to allow deletion by line number with respect to category.
318 * Added new action insert to add new variable to category at specified line.
319 * Updated action newcat to allow new category to be inserted in file above another
321 * Added new event "JitterBufStats" in the IAX2 channel
322 * Originate now requires the Originate privilege and, if you want to call out
323 to a subshell, it requires the System privilege, as well. This was done to
324 enhance manager security.
325 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
326 * New command: Atxfer. See doc/manager_1_1.txt for more details or
327 manager show command Atxfer from the CLI
328 * New command: IAXregistry. See doc/manager_1_1.txt for more details or
329 manager show command IAXregistry from the CLI
333 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
334 state in the dialplan, as well as creating custom device states that are
335 controllable from the dialplan.
336 * Extend CALLERID() function with "pres" and "ton" parameters to
337 fetch string representation of calling number presentation indicator
338 and numeric representation of type of calling number value.
339 * MailboxExists converted to dialplan function
340 * A new option to Dial() for telling IP phones not to count the call
341 as "missed" when dial times out and cancels.
342 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
343 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
344 held for any given channel. Also, locks are automatically freed when a
346 * Added HINT() dialplan function that allows retrieving hint information.
347 Hints are mappings between extensions and devices for the sake of
348 determining the state of an extension. This function can retrieve the list
349 of devices or the name associated with a hint.
350 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
352 * Added SYSINFO() dialplan function which allows retrieval of system information
353 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
354 the existence of a dialplan target.
355 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
356 upper and lower case, respectively.
357 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
358 ID for the call (not the Asterisk call ID or unique ID), provided that the
359 channel driver supports this. For SIP, you get the SIP call-ID for the
360 bridged channel which you can store in the CDR with a custom field.
364 * New CLI command "core show hint" (usage: core show hint <exten>)
365 * New CLI command "core show settings"
366 * Added 'core show channels count' CLI command.
367 * Added the ability to set the core debug and verbose values on a per-file basis.
368 * Added 'queue pause member' and 'queue unpause member' CLI commands
369 * Ability to set process limits ("ulimit") without restarting Asterisk
370 * Enhanced "agi debug" to print the channel name as a prefix to the debug
371 output to make debugging on busy systems much easier.
372 * New CLI commands "dialplan set extenpatternmatching true/false"
373 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
374 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
375 listed in the startup_commands section of cli.conf will get executed.
376 * Added a CLI command, "devstate change", which allows you to set custom device
377 states from the func_devstate module that provides the DEVICE_STATE() function
378 and handling of the "Custom:" devices.
379 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
380 sorted into the different possible callbacks, with the number of entries
381 currently scheduled for each. Gives you a feel for how busy the sip channel
383 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
384 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
385 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
389 * Improved NAT and STUN support.
390 chan_sip now can use port numbers in bindaddr, externip and externhost
391 options, as well as contact a STUN server to detect its external address
392 for the SIP socket. See sip.conf.sample, 'NAT' section.
393 * The default SIP useragent= identifier now includes the Asterisk version
394 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
395 If set, and the incoming request carries authentication info,
396 the username to match in the users list is taken from the Digest header
397 rather than from the From: field. This feature is considered experimental.
398 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
399 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
400 * The "localmask" setting was removed in version 1.2 and the reminder about it
401 being removed is now also removed.
402 * A new option "busylevel" for setting a level of calls where asterisk reports
403 a device as busy, to separate it from call-limit. This value is also added
404 to the SIP_PEER dialplan function.
405 * A new realtime family called "sipregs" is now supported to store SIP registration
406 data. If this family is defined, "sippeers" will be used for configuration and
407 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
408 registration data, as before.
409 * The SIPPEER function have new options for port address, call and pickup groups
410 * Added support for T.140 realtime text in SIP/RTP
411 * The "checkmwi" option has been removed from sip.conf, as it is no longer
412 required due to the restructuring of how MWI is handled. See the descriptions
413 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
414 for more information.
415 * Added rtpdest option to CHANNEL() dialplan function.
416 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
417 * SIP now adds a header to the CANCEL if the call was answered by another phone
418 in the same dial command, or if the new c option in dial() is used.
419 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
420 states it is not needed. For phones, however, that do require it the "registertrying" option
421 has been added so it can be enabled.
422 * A new option called "callcounter" (global/peer/user level) enables call counters needed
423 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
424 used to enable this functionality).
425 * New settings for timer T1 and timer B on a global level or per device. This makes it
426 possible to force timeout faster on non-responsive SIP servers. These settings are
427 considered advanced, so don't use them unless you have a problem.
428 * Added a dial string option to be able to set the To: header in an INVITE to any
430 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
431 the qualify frequency.
432 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
433 were not properly torn down due to network or endpoint failures during an established
435 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
436 configs/sip.conf.sample for more information on how it is used.
437 * Added a new configuration option "authfailureevents" that enables manager events when
438 a peer can't authenticate properly.
439 * Added DNS manager support to registrations for peers not referencing a peer entry.
443 * Added the trunkmaxsize configuration option to chan_iax2.
444 * Added the srvlookup option to iax.conf
445 * Added support for OSP. The token is set and retrieved through the CHANNEL()
448 XMPP Google Talk/Jingle changes
449 -------------------------------
450 * Added the bindaddr option to gtalk.conf.
454 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
455 * Proper codec support in chan_skinny.
456 * Added settings for IP and Ethernet QoS requests
460 * Added separate settings for media QoS in mgcp.conf
462 Console Channel Driver changes
463 ------------------------------
464 * Added experimental support for video send & receive to chan_oss.
465 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
468 Phone channel changes (chan_phone)
469 ----------------------------------
470 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
472 H.323 channel Changes
473 ---------------------
474 * H323 remote hold notification support added (by NOTIFY message
475 and/or H.450 supplementary service)
477 Local channel changes
478 ---------------------
479 * The device state functionality in the Local channel driver has been updated
480 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
481 to just UNKNOWN if the extension exists.
482 * Added jitterbuffer support for chan_local. This allows you to use the
483 generic jitterbuffer on incoming calls going to Asterisk applications.
484 For example, this would allow you to use a jitterbuffer for an incoming
485 SIP call to Voicemail by putting a Local channel in the middle. This
486 feature is enabled by using the 'j' option in the Dial string to the Local
487 channel in conjunction with the existing 'n' option for local channels.
488 * A 'b' option has been added which causes chan_local to return the actual channel
489 that is behind it when queried. This is useful for transfer scenarios as the
490 actual channel will be transferred, not the Local channel.
492 Agent channel changes
493 ----------------------
494 * The ackcall and endcall options are now supplemented with options acceptdtmf
495 and enddtmf. These allow for the DTMF keypress to be configurable. The options
496 default to their old hard-coded values ('#' and '*' respectively) so this should
497 not break any existing agent installations.
499 DAHDI channel driver (chan_dahdi) Changes
500 ----------------------------------------
501 * SS7 support (via libss7 library)
502 * In India, some carriers transmit CID via dtmf. Some code has been added
503 that will handle some situations. The cidstart=polarity_IN choice has been added for
504 those carriers that transmit CID via dtmf after a polarity change.
505 * CID matching information is now shown when doing 'dialplan show'.
506 * Added dahdi show version CLI command.
507 * Added setvar support to chan_dahdi.conf channel entries.
508 * Added two new options: mwimonitor and mwimonitornotify. These options allow
509 you to enable MWI monitoring on FXO lines. When the MWI state changes,
510 the script specified in the mwimonitornotify option is executed. An internal
511 event indicating the new state of the mailbox is also generated, so that
512 the normal MWI facilities in Asterisk work as usual.
513 * Added signalling type 'auto', which attempts to use the same signalling type
514 for a channel as configured in DAHDI. This is primarily designed for analog
515 ports, but will also work for digital ports that are configured for FXS or FXO
516 signalling types. This mode is also the default now, so if your chan_dahdi.conf
517 does not specify signalling for a channel (which is unlikely as the sample
518 configuration file has always recommended specifying it for every channel) then
519 the 'auto' mode will be used for that channel if possible.
520 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
521 state for a channel; also ensured that the DNDState Manager event is
522 emitted no matter how the DND state is set or cleared.
526 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
527 configs/unistim.conf.sample for details. This new channel driver allows
528 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
529 * Added a new channel driver, chan_console, which uses portaudio as a cross
530 platform audio interface. It was written as a channel driver that would
531 work with Mac CoreAudio, but portaudio supports a number of other audio
532 interfaces, as well. Note that this channel driver requires v19 or higher
533 of portaudio; older versions have a different API.
537 * Added the ability to specify arguments to the Dial application when using
538 the DUNDi switch in the dialplan.
539 * Added the ability to set weights for responses dynamically. This can be
540 done using a global variable or a dialplan function. Using the SHELL()
541 function would allow you to have an external script set the weight for
543 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
544 functions will allow you to initiate a DUNDi query from the dialplan,
545 find out how many results there are, and access each one.
549 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
550 functions will allow you to initiate an ENUM lookup from the dialplan,
551 and Asterisk will cache the results. ENUMRESULT can be used to access
552 the results without doing multiple DNS queries.
556 * Added the ability to customize which sound files are used for some of the
557 prompts within the Voicemail application by changing them in voicemail.conf
558 * Added the ability for the "voicemail show users" CLI command to show users
559 configured by the dynamic realtime configuration method.
560 * MWI (Message Waiting Indication) handling has been significantly
561 restructured internally to Asterisk. It is now totally event based
562 instead of polling based. The voicemail application will notify other
563 modules that have subscribed to MWI events when something in the mailbox
565 This also means that if any other entity outside of Asterisk is changing
566 the contents of mailboxes, then the voicemail application still needs to
567 poll for changes. Examples of situations that would require this option
568 are web interfaces to voicemail or an email client in the case of using
569 IMAP storage. So, two new options have been added to voicemail.conf
570 to account for this: "pollmailboxes" and "pollfreq". See the sample
571 configuration file for details.
572 * Added "tw" language support
573 * Added support for storage of greetings using an IMAP server
574 * Added ability to customize forward, reverse, stop, and pause keys for message playback
575 * SMDI is now enabled in voicemail using the smdienable option.
576 * A "lockmode" option has been added to asterisk.conf to configure the file
577 locking method used for voicemail, and potentially other things in the
578 future. The default is the old behavior, lockfile. However, there is a
579 new method, "flock", that uses a different method for situations where the
580 lockfile will not work, such as on SMB/CIFS mounts.
581 * Added the ability to backup deleted messages, to ease recovery in the case
582 that a user accidentally deletes a message, and discovers that they need it.
583 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
584 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
585 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
586 voicemail boxes. The SMDI interface can also poll for MWI changes when some
587 outside entity is modifying the state of the mailbox (such as IMAP storage or
588 a web interface of some kind).
589 * Added the support for marking messages as "urgent." There are two methods to accomplish
590 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
591 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
592 the message as urgent after he has recorded a voicemail by following the voice instructions.
593 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
598 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
599 used across multiple queues.
600 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
601 setqueueentryvar options for each queue, see queues.conf.sample for details.
602 * Added keepstats option to queues.conf which will keep queue
603 statistics during a reload.
604 * setinterfacevar option in queues.conf also now sets a variable
605 called MEMBERNAME which contains the member's name.
606 * Added 'Strategy' field to manager event QueueParams which represents
607 the queue strategy in use.
608 * Added option to run macro when a queue member is connected to a caller,
609 see queues.conf.sample for details.
610 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
611 does not count paused queue members as unavailable.
612 * Added min-announce-frequency option to queues.conf which allows you to control the
613 minimum amount of time between queue announcements for use when the caller's queue
614 position changes frequently.
615 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
617 * Added ability for non-realtime queues to have realtime members
618 * Added the "linear" strategy to queues.
619 * Added the "wrandom" strategy to queues.
620 * Added new channel variable QUEUE_MIN_PENALTY
621 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
622 rules in queuerules.conf. See configs/queuerules.conf.sample for details
623 * Added a new parameter for member definition, called state_interface. This may be
624 used so that a member may be called via one interface but have a different interface's
625 device state reported.
626 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
627 specified by the periodic-announce option, then one will be chosen randomly when it is time
628 to play a periodic announcment
629 * New configuration options: announce-position now takes two more values in addition to "yes" and
630 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
631 announce-position-limit. By setting announce-position to "limit" callers will only have their
632 position announced if their position is less than what is specified by announce-position-limit.
633 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
634 will be told that their are more than announce-position-limit callers waiting.
635 * Two new queue log events have been added. An ADDMEMBER event will be logged
636 when a realtime queue member is added and a REMOVEMEMBER event will be logged
637 when a realtime queue member is removed. Since there is no calling channel associated
638 with these events, the string "REALTIME" is placed where the channel's unique id
640 * The configuration method for the "joinempty" and "leavewhenempty" options has
641 changed to a comma-separated list of methods of determining member availability
642 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
643 values are still accepted for backwards-compatibility, though.
647 * The 'o' option to provide an optimization has been removed and its functionality
648 has been enabled by default.
649 * When a conference is created, the UNIQUEID of the channel that caused it to be
650 created is stored. Then, every channel that joins the conference will have the
651 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
652 callers that come and go from long standing conferences.
653 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
654 except it does operations on a channel by name, instead of number in a conference.
655 This is a very useful feature in combination with the 'X' option to ChanSpy.
656 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
658 * Added new RealTime functionality to provide support for scheduled conferencing.
659 This includes optional messages to the caller if they attempt to join before
660 the schedule start time, or to allow the caller to join the conference early.
661 Also included is optional support for limiting the number of callers per
663 * Added the S() and L() options to the MeetMe application. These are pretty
664 much identical to the S() and L() options to Dial(). They let you set
665 timeouts for the conference, as well as have warning sounds played to
666 let the caller know how much time is left, and when it is running out.
667 * Added the ability to do "meetme concise" with the "meetme" CLI command.
668 This extends the concise capabilities of this CLI command to include
669 listing all conferences, instead of an addition to the other sub commands
670 for the "meetme" command.
671 * Added the ability to specify the music on hold class used to play into the
672 conference when there is only one member and the M option is used.
673 * Added MEETME_INFO dialplan function which provides a way to query
674 various properties of a Meetme conference.
676 Other Dialplan Application Changes
677 ----------------------------------
678 * Argument support for Gosub application
679 * From the to-do lists: straighten out the app timeout args:
680 Wait() app now really does 0.3 seconds- was truncating arg to an int.
681 WaitExten() same as Wait().
682 Congestion() - Now takes floating pt. argument.
683 Busy() - now takes floating pt. argument.
684 Read() - timeout now can be floating pt.
685 WaitForRing() now takes floating pt timeout arg.
686 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
687 * Added 's' option to Page application.
688 * Added 'E', 'V', and 'P' commands to ExternalIVR.
689 * Added 'o' and 'X' options to Chanspy.
690 * Added a new dialplan application, Bridge, which allows you to bridge the
691 calling channel to any other active channel on the system.
692 * Added the ability to specify a music on hold class to play instead of ringing
693 for the SLATrunk application.
694 * The Read application no longer exits the dialplan on error. Instead, it sets
695 READSTATUS to ERROR, which you can catch and handle separately.
696 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
697 of asking for verification of each name, one at a time.
698 * Privacy() no longer uses privacy.conf, as all options are specifyable as
699 direct options to the app.
700 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
702 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
703 * The ChannelRedirect application no longer exits the dialplan if the given channel
704 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
705 or NOCHANNEL if the given channel was not found.
706 * The silencethreshold setting that was previously configurable in multiple
707 applications is now settable globally via dsp.conf.
709 Music On Hold Changes
710 ---------------------
711 * A new option, "digit", has been added for music on hold classes in
712 musiconhold.conf. If this is set for a music on hold class, a caller
713 listening to music on hold can press this digit to switch to listening
714 to this music on hold class.
715 * Support for realtime music on hold has been added.
716 * In conjunction with the realtime music on hold, a general section has
717 been added to musiconhold.conf, its sole variable is cachertclasses. If this
718 is set, then music on hold classes found in realtime will be cached in memory.
722 * AEL upgraded to use the Gosub with Arguments instead
723 of Macro application, to hopefully reduce the problems
724 seen with the artificially low stack ceiling that
725 Macro bumps into. Macros can only call other Macros
726 to a depth of 7. Tests run using gosub, show depths
727 limited only by virtual memory. A small test demonstrated
728 recursive call depths of 100,000 without problems.
729 -- in addition to this, all apps that allowed a macro
730 to be called, as in Dial, queues, etc, are now allowing
731 a gosub call in similar fashion.
732 * AEL now generates LOCAL(argname) declarations when it
733 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
734 etc. That makes the arguments local in scope. The user
735 can define their own local variables in macros, now,
736 by saying "local myvar=someval;" or using Set() in this
737 fashion: Set(LOCAL(myvar)=someval); ("local" is now
739 * utils/conf2ael introduced. Will convert an extensions.conf
740 file into extensions.ael. Very crude and unfinished, but
741 will be improved as time goes by. Should be useful for a
742 first pass at conversion.
743 * aelparse will now read extensions.conf to see if a referenced
744 macro or context is there before issueing a warning.
745 * AEL parser sets a local channel variable ~~EXTEN~~, to
746 preserve the value of ${EXTEN} thru switch statements.
747 * New operator in $[...] expressions: the ~~ operator serves
748 as a concatenation operator. AT THE MOMENT, it is really only
749 necessary and useful in AEL, especially in if() expressions.
750 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
751 any enclosing double-quotes, and evaluate to the value of a
752 concatenated with the value of b. For example if a is set to
753 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
757 Call Features (res_features) Changes
758 ------------------------------------
759 * Added the parkedcalltransfers option to features.conf
760 * The built-in method for doing attended transfers has been updated to
761 include some new options that allow you to have the transferee sent
762 back to the person that did the transfer if the transfer is not successful.
763 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
764 in features.conf.sample.
765 * Added support for configuring named groups of custom call features in
766 features.conf. This means that features can be written a single time, and
767 then mapped into groups of features for different key mappings or easier
769 * Updated the ParkedCall application to allow you to not specify a parking
770 extension. If you don't specify a parking space to pick up, it will grab
771 the first one available.
772 * Added cli command 'features reload' to reload call features from features.conf
773 * Moved into core asterisk binary.
774 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
776 Language Support Changes
777 ------------------------
778 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
779 * Added support for the Hungarian language for saying numbers, dates, and times.
783 * Added SPEECH commands for speech recognition. A complete listing can be found
785 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
786 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
787 does not behave as expected; the native command needs to be used, instead.
791 * Added rotatestrategy option to logger.conf, along with two new options:
792 "timestamp" which will use the time to name the logger files instead of
793 sequence number; and "rotate", which rotates the names of the logfiles,
794 similar to the way syslog rotates files.
795 * Added exec_after_rotate option to logger.conf, which allows a system
796 command to be run after rotation. This is primarily useful with
797 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
798 and to ensure that the oldest log file gets deleted.
799 * Added realtime support for the queue log
803 * The cdr_manager module has a [mappings] feature, like cdr_custom,
804 to add fields to the manager event from the CDR variables.
805 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
806 backend database CDR table. Specifically, additional, non-standard
807 columns are supported, merely by setting the corresponding CDR variable in
808 your dialplan. In addition, you may alias any column to another name (for
809 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
810 simply "alias src => ANI" in the configuration file). Records may be
811 posted to more than one backend, simply by specifying multiple categories
812 in the configuration file. And finally, you may filter which CDRs get
813 posted to each backend, by specifying a filter (which the record must
814 match) for the particular category. Filters are additive (meaning all
815 rules must match to post that CDR).
816 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
817 module. Specifically, you may add additional columns into the table and
818 they will be set, if you set the corresponding CDR variable name. Also,
819 if you omit columns in your database table, they will be silently skipped
820 (but a record will still be inserted, based on what columns remain). Note
821 that the other two features from cdr_adaptive_odbc (alias and filter) are
822 not currently supported.
823 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
824 has been disabled using the NoCDR application.
826 Miscellaneous New Modules
827 -------------------------
828 * Added a new CDR module, cdr_sqlite3_custom.
829 * Added a new realtime configuration module, res_config_sqlite
830 * Added a new codec translation module, codec_resample, which re-samples
831 signed linear audio between 8 kHz and 16 kHz to help support wideband
833 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
834 based on configuration templates that use Asterisk dialplan function and
835 variable substitution. It should be possible to create phone profiles and
836 templates that work for the majority of phones provisioned over http. It
837 is currently only intended to provision a single user account per phone.
838 An example profile and set of templates for Polycom phones is provided.
839 NOTE: Polycom firmware is not included, but should be placed in
840 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
841 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
842 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
843 provided; there is a JACK() application, and a JACK_HOOK() function. Both
844 interfaces create an input and output JACK port. The application makes
845 these ports the endpoint of the call. The audio coming from the channel
846 goes out the output port and whatever comes back in on the input port is
847 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
848 audiohook on the channel. This lets you run the audio coming from a
849 channel through JACK, and whatever comes back in is what gets forwarded
850 on as the channel's audio. This is very useful for building custom
851 vocoders or doing recording or analysis of the channel's audio in another
853 * Added a new module, res_config_curl, which permits using a HTTP POST url
854 to retrieve, create, update, and delete realtime information from a remote
855 web server. Note that this module requires func_curl.so to be loaded for
856 backend functionality.
857 * Added a new module, res_config_ldap, which permits the use of an LDAP
858 server for realtime data access.
859 * Added support for writing and running your dialplan in lua using the pbx_lua
860 module. See configs/extensions.lua.sample for examples of how to do this.
864 * Ability to use libcap to set high ToS bits when non-root
865 on Linux. If configure is unable to find libcap then you
866 can use --with-cap to specify the path.
867 * Added maxfiles option to options section of asterisk.conf which allows you to specify
868 what Asterisk should set as the maximum number of open files when it loads.
869 * Added the jittertargetextra configuration option.
870 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
871 configuration files for the IP channel drivers. The new option is "cos".
872 This information is also documented in doc/qos.tex, or the IP Quality of Service
873 section of asterisk.pdf.
874 * When originating a call using AMI or pbx_spool that fails the reason for failure
875 will now be available in the failed extension using the REASON dialplan variable.
876 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
877 It allows you to configure a prefix for auto-monitor recordings.
878 * A new extension pattern matching algorithm, based on a trie, is introduced
879 here, that could noticeably speed up mid-sized to large dialplans.
880 It is NOT used by default, as duplicating the behaviour of the old pattern
881 matcher is still under development. A config file option, in extensions.conf,
882 in the [general] section, called "extenpatternmatchingnew", is by default
883 set to false; setting that to true will force the use of the new algorithm.
884 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
885 be used to switch the algorithms at run time.
886 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
887 specifying which socket to use to connect to the running Asterisk daemon
889 * Performance enhancements to the sched facility, which is used in
890 the channel drivers, etc. Added hashtabs and doubly-linked lists
891 to speed up deletion; start at the beginning or end of list to
893 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
894 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
895 Added regression tests to the tests/ dir, also.
896 * Added a refcount trace feature to astobj2 for those trying to balance
897 object creation, deletion; work, play; space and time. See the
898 notes in astobj2.h. Also, see utils/refcounter as well, as a
899 quick way to find unbalanced refcounts in what could be a sea
900 of objects that were balanced.
901 * Added logging to 'make update' command. See update.log
902 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
903 do not come from the remote party.
904 * Added the 'n' option to the SpeechBackground application to tell it to not
905 answer the channel if it has not already been answered.
906 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
907 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
909 * iLBC source code no longer included (see UPGRADE.txt for details)
910 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
911 deadlock is detected, a backtrace of the stack which led to the lock calls
912 will be output to the CLI.
913 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
914 the "core show locks" CLI command will give lock information output as well
915 as a backtrace of the stack which led to the lock calls.
916 * users.conf now sports an optional alternateexts property, which permits
917 allocation of additional extensions which will reach the specified user.
918 * A new option for the configure script, --enable-internal-poll, has been added
919 for use with systems which may have a buggy implementation of the poll system
920 call. If you notice odd behavior such as the CLI being unresponsive on remote
921 consoles, you may want to try using this option. This option is enabled by default
922 on Darwin systems since it is known that the Darwin poll() implementation has