1 ======================================================================
3 === This file documents the new and/or enhanced functionality added in
4 === the Asterisk versions listed below. This file does NOT include
5 === changes in behavior that would not be backwards compatible with
6 === previous versions; for that information see the UPGRADE.txt file
7 === and the other UPGRADE files for older releases.
9 ======================================================================
10 ------------------------------------------------------------------------------
11 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.6.3 -------------
12 ------------------------------------------------------------------------------
18 * Added preferred_codec_only option in sip.conf. This feature limits the joint
19 codecs sent in response to an INVITE to the single most preferred codec.
20 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
21 to be used for the outgoing call. It must be one of the codecs configured
23 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
24 to be used for holding a private key. If tlsprivatekey is not specified,
25 tlscertfile is searched for both public and private key.
29 * Added progress option to the app_dial D() option. When progress DTMF is
30 present, those values are sent immediatly upon receiving a PROGRESS message
31 regardless if the call has been answered or not.
32 * Added functionality to the app_dial F() option to continue with execution
33 at the current location when no parameters are provided.
34 * Added c() option to app_chanspy. This option allows custom DTMF to be set
35 to cycle through the next avaliable channel. By default this is still '*'.
36 * Added x() option to app_chanspy. This option allows DTMF to be set to
41 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
42 setting various connected line and redirecting party information.
43 * The CHANNEL() function now supports the "name" option.
47 * A new option, 'I' has been added to both app_queue and app_dial.
48 By setting this option, Asterisk will not update the caller with
49 connected line changes or redirecting party changes when they occur.
51 mISDN channel driver (chan_misdn) changes
52 ----------------------------------------
53 * Added display_connected parameter to misdn.conf to put a display string
54 in the CONNECT message containing the connected name and/or number if
55 the presentation setting permits it.
56 * Added display_setup parameter to misdn.conf to put a display string
57 in the SETUP message containing the caller name and/or number if the
58 presentation setting permits it.
59 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
60 indicate the dialplan settings are to be obtained from the asterisk
62 * Made misdn.conf parameter callerid accept the "name" <number> format
63 used by the rest of the system.
64 * Made use the nationalprefix and internationalprefix misdn.conf
65 parameters to prefix any received number from the ISDN link if that
66 number has the corresponding Type-Of-Number.
67 * Added the following new parameters: unknownprefix, netspecificprefix,
68 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
69 received number from the ISDN link if that number has the corresponding
71 * Added new dialplan application misdn_command which permits controlling
72 the CCBS/CCNR functionality.
73 * Added new dialplan function mISDN_CC which permits retrieval of various
74 values from an active call completion record.
75 * For PTP, you should manually send the COLR of the redirected-to party
76 for an incomming redirected call if the incoming call could experience
77 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
78 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
79 if the REDIRECTING(from-num) is not empty.
80 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
81 option on all of the REDIRECTING statements before dialing the
82 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
83 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
84 redirecting-to presentation (COLR) when it becomes available.
86 thirdparty mISDN enhancements
87 -----------------------------
88 mISDN has been modified by Digium, Inc. to greatly expand facility message
90 * Enhanced COLP support for call diversion and transfer.
93 The latest modified mISDN v1.1.x based version is available at:
94 http://svn.digium.com/svn/thirdparty/mISDN/trunk
95 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
97 Taged versions of the modified mISDN code are available under:
98 http://svn.digium.com/svn/thirdparty/mISDN/tags
99 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
101 SIP channel driver (chan_sip) changes
102 -------------------------------------------
103 * The sendrpid parameter has been expanded to include the options
104 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
105 header to be sent (equivalent to setting sendrpid=yes) and setting
106 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
108 Asterisk Manager Interface
109 --------------------------
110 * The Hangup action now accepts a Cause header which may be used to
111 set the channel's hangup cause.
112 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
113 to specify a separate .pem file to hold a private key. By default sslcert
114 is used to hold both the public and private key.
115 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
116 for options containing the 'tls' prefix. For example, 'sslenable' is now
117 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
118 across all .conf files. All affected sample.conf files have been modified to
119 reflect this change. Previous options such as 'sslenable' still work,
120 but options with the 'tls' prefix are preferred.
121 ------------------------------------------------------------------------------
122 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
123 ------------------------------------------------------------------------------
127 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
128 Snom phones use this for call pickup of extensions that the phone is
130 * Added support for subscribing to a voice mailbox on a remote server and
131 making the new/old message count available to local devices.
132 * Added support for setting the domain in the URI for caller of an
133 outbound call by using the SIPFROMDOMAIN channel variable.
134 * Added a new configuration option "remotesecret" for authentication to
135 remote services. For backwards compatibility, "secret" still has the
136 same function as before, but now you can configure both a remote secret and a
137 local secret for mutual authentication.
138 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
139 option is enabled, a SIP channel will go to the fax extension (if it exists)
140 after T38 is negotiated. This option is disabled by default.
141 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
142 the sound will be played to the target of an attended transfer
143 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
144 finer control over how many peers Asterisk will qualify and the gap between them
145 when all peers need to be qualified at the same time.
146 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
147 (either globally or for a specific peer), chan_sip will treat any SDP data
148 it receives as new data and update the media stream accordingly. By
149 default, Asterisk will only modify the media stream if the SDP session
150 version received is different from the current SDP session version. This
151 option is required to interoperate with devices that have non-standard SDP
152 session version implementations (observed with Microsoft OCS). This option
153 is disabled by default.
154 * The parsing of register => lines in sip.conf has been modified to allow a port
155 to be present in the "user" portion. Please see the sip.conf.sample file for more
157 * Added support for subscribing to MWI on a remote server and making the status available
158 as a mailbox. Please see the sip.conf.sample file for more information.
159 * Added a function to remove SIP headers added in the dialplan before the
160 first INVITE is generated - SIPRemoveHeader()
161 * Channel variables set with setvar= in a device configuration is now
162 set both for inbound and outbound calls.
163 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
167 * Added immediate option to iax.conf
168 * Added forceencryption option to iax.conf
169 * Added Encryption and Trunk status to manager command "iaxpeers"
173 * The configuration file now holds separate sections for devices and lines.
174 Please have a look at configs/skinny.conf.sample and change your skinny.conf
179 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
180 support for LibOpenR2. http://www.libopenr2.org/
181 * The UK option waitfordialtone has been added for use with BT analog
183 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
184 is used in conjunction with the 'faxdetect' configuration option. When
185 'faxbuffers' is used and fax tones are detected, the channel will dynamically
186 switch to the configured faxbuffers policy. For example, to use 6 buffers
187 and a 'full' buffer policy for a fax transmission, add:
189 The faxbuffers configuration will be in affect until the call is torn down.
190 * Added service message support for 4ESS/5ESS switches.
194 * Added a new dialplan function, CURLOPT, which permits setting various
195 options that may be useful with the CURL dialplan function, such as
196 cookies, proxies, connection timeouts, passwords, etc.
197 * Permit the syntax and synopsis fields of the corresponding dialplan
198 functions to be individually set from func_odbc.conf.
199 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
200 * func_odbc now may specify an insert query to execute, when the write query
201 affects 0 rows (usually indicating that no such row exists).
202 * Added a new dialplan function, LISTFILTER, which permits removing elements
203 from a set list, by name. Uses the same general syntax as the existing CUT
204 and FIELDQTY dialplan functions, which also manage lists.
205 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
206 obtaining realtime data from the dialplan.
207 * Added LOCAL_PEEK, which I have no idea how to use, but Leif Madsen wanted it.
208 Russell says it's, like, a scope resolution function for LOCAL variables.
209 Totally. Hopefully, that means more to you than it does to me.
210 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
211 of "core show function AUDIOHOOK_INHERIT" from the CLI
212 * Added AES_ENCRYPT. For information on its use, please see the output
213 of "core show function AES_ENCRYPT" from the CLI
214 * Added AES_DECRYPT. For information on its use, please see the output
215 of "core show function AES_DECRYPT" from the CLI
216 * func_odbc now supports database transactions across multiple queries.
220 * DAHDISendCallreroutingFacility parameters are now comma-separated,
221 instead of the old pipe.
222 * Scheduled meetme conferences may now have their end times extended by
224 * app_authenticate now gives the ability to select a prompt other than
226 * app_directory now pays attention to the searchcontexts setting in
227 voicemail.conf and will look through all contexts, if no context is
228 specified in the initial argument.
229 * A new application, Originate, has been introduced, that allows asynchronous
230 call origination from the dialplan.
231 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
232 in addition to the setting in the "general" context.
233 * Added ConfBridge dialplan application which does conference bridges without
234 DAHDI. For information on its use, please see the output of
235 "core show application ConfBridge" from the CLI.
239 * The Asterisk CLI has a new command, "channel redirect", which is similar in
240 operation to the AMI Redirect action.
241 * res_jabber: autoprune has been disabled by default, to avoid misconfiguration
242 that would end up being interpreted as a bug once Asterisk started removing
243 the contacts from a user list.
244 * extensions.conf now allows you to use keyword "same" to define an extension
245 without actually specifying an extension. It uses exactly the same pattern
246 as previously used on the last "exten" line. For example:
247 exten => 123,1,NoOp(something)
248 same => n,SomethingElse()
249 * musiconhold.conf classes of type 'files' can now use relative directory paths,
250 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
251 * All deprecated CLI commands are removed from the sourcecode. They are now handled
252 by the new clialiases module. See cli_aliases.conf.sample file.
253 * Times within timespecs are now accurate down to the minute. This is a change
254 from historical Asterisk, which only provided timespecs rounded to the nearest
255 even (read: evenly divisible by 2) minute mark.
256 * The realtime switch now supports an option flag, 'p', which disables searches for
258 * In addition to a time range and date range, timespecs now accept a 5th optional
259 argument, timezone. This allows you to perform time checks on alternate
260 timezones, especially if those daylight savings time ranges vary from your
261 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
263 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
264 give you the correct output for an asterisk box behind nat. It will give you the
265 externhost and localnet settings.
266 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
267 can connect calls in passthrough mode, as well as record and play back files.
268 * Successful and unsuccessful call pickup can now be alerted through sounds, by
269 using pickupsound and pickupfailsound in features.conf.
270 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
271 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
272 instead of the /var/run/asterisk.pid where it used to be. This will make
273 installs as non-root easier to manage.
275 Asterisk Manager Interface
276 --------------------------
277 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
278 a non-empty value) in your request. If you do this, any pending AMI events will
279 *not* be included in the response to your request as they would normally, but
280 will be left in the event queue for the next request you make to retrieve. For
281 some applications, this will allow you to guarantee that you will only see
282 events in responses to 'WaitEvent' actions, and can better know when to expect them.
283 To know whether the Asterisk server supports this header or not, your client can
284 inspect the first response back from the server to see if it includes this header:
286 Pragma: SuppressEvents
288 If this is included, the server supports event suppression.
290 * Added 4 new Actions to list skinny device(s) and line(s)
296 ------------------------------------------------------------------------------
297 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
298 ------------------------------------------------------------------------------
300 Device State Handling
301 ---------------------
302 * The event infrastructure in Asterisk got another big update to help support
303 distributed events. It currently supports distributed device state and
304 distributed Voicemail MWI (Message Waiting Indication). A new module has
305 been merged, res_ais, which facilitates communicating events between servers.
306 It uses the SAForum AIS (Service Availability Forum Application Interface
307 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
308 a cluster of Asterisk servers, and to share events between them. For more
309 information on setting this up, see doc/distributed_devstate.txt.
313 * Added a new dialplan function, AST_CONFIG(), which allows you to access
314 variables from an Asterisk configuration file.
315 * The JACK_HOOK function now has a c() option to supply a custom client name.
316 * Added two new dialplan functions from libspeex for audio gain control and
317 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
318 rx directions of a channel from the dialplan.
319 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
320 based on other parameters. The default is still to search based on the
321 forwarding station ID. However, there are new options that allow you to search
322 based on the message desk terminal ID, or the message desk number.
323 * TIMEOUT() has been modified to be accurate down to the millisecond.
324 * ENUM*() functions now include the following new options:
325 - 'u' returns the full URI and does not strip off the URI-scheme.
326 - 's' triggers ISN specific rewriting
327 - 'i' looks for branches into an Infrastructure ENUM tree
328 - 'd' for a direct DNS lookup without any flipping of digits.
329 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
330 * CHANNEL() now has options for the maximum, minimum, and standard or normal
331 deviation of jitter, rtt, and loss for a call using chan_sip.
333 DAHDI channel driver (chan_dahdi) Changes
334 ----------------------------------------
335 * Channels can now be configured using named sections in chan_dahdi.conf, just
336 like other channel drivers, including the use of templates.
337 * The default for pridialplan has changed from 'national' to 'unknown'.
341 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
342 to something that matches the pattern a hint will be created using the contents
343 and variables evaluated.
344 * Dialplan matching has been extended to allow an extension to return to the
345 PBX core to wait for more digits. This is done by using the new dialplan
346 application called "Incomplete". This will permit a whole new level of
347 extension control, by giving the administrator more control over early
348 matches employing one of the short-circuit pattern match operators. Note
349 that custom applications can trigger this same behavior by returning the
350 special value AST_PBX_INCOMPLETE.
354 * Directory now permits both first and last names to be matched at the same
355 time. In addition, the number of digits to enter of the name can be set in
356 the arguments to Directory; previously, you could enter only 3, regardless
357 of how many names are in your company. For large companies, this should be
359 * Voicemail now permits a mailbox setting to wrap around from first to last
360 messages, if the "messagewrap" option is set to a true value.
361 * Voicemail now permits an external script to be run, for password validation.
362 The script should output "VALID" or "INVALID" on stdout, depending upon the
363 wish to validate or invalidate the password given. Arguments are:
364 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
366 * Dial has a new option: F(context^extension^pri), which permits a callee to
367 continue in the dialplan, at the specified label, if the caller hangs up.
368 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
369 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
370 * The Jack application now has a c() option to supply a custom client name.
371 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
372 like the pre-existing whisper mode, except that the spy can also talk to the
373 participant on the bridged channel as well.
374 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
375 to be spoken instead of the channel name or number. For more information on the
376 use of this option, issue the command "core show application ChanSpy" from the
378 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
379 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
380 words, if using the 'd' option, it is not possible to enter a number to append to
381 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
382 change to whisper mode, and pressing 6 will change to barge mode.
383 * ExternalIVR now takes several options that affect the way it performs, as
384 well as having several new commands. Please see doc/externalivr.txt for the
385 complete documentation.
386 * Added ability to communicate over a TCP socket instead of forking a child process for the
387 ExternalIVR application.
388 * ChanIsAvail has a new option, 'a', which will return all available channels instead
389 of just the first one if you give the function more then one channel to check.
390 * PrivacyManager now takes an option where you can specify a context where the
391 given number will be matched. This way you have more control over who is allowed
392 and it stops the people who blindly enter 10 digits.
393 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
394 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
395 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
396 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
397 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
398 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
399 * The Dial() application no longer copies the language used by the caller to the callee's
400 channel. If you desire for the caller's channel's language to be used for file playback
401 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
402 * SendImage() no longer hangs up the channel on error; instead, it sets the
403 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
404 'UNSUPPORTED'. This change makes SendImage() more consistent with other
406 * Park has a new option, 's', which silences the announcement of the parking space number.
407 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
408 invalid input and will be assumed to mean that no timeout is desired.
412 * Added DNS manager support to registrations for peers referencing peer entries.
413 DNS manager runs in the background which allows DNS lookups to be run asynchronously
414 as well as periodically updating the IP address. These properties allow for
415 better performance as well as recovery in the event of an IP change.
416 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
417 load/reload of large numbers of peers/users by ~40x (for large lists of peers.
418 Initially, we saw 4x improvement in call setup/destruction, but at the time
419 of merging, this gain has disappeared; further research will be done to try
420 and restore this performance improvement. Astobj2 refcounting is now used
421 for users, peers, and dialogs. Users are encouraged to assist in regression
422 testing and problem reporting!
423 * Added ability to specify registration expiry time on a per registration basis in
425 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
427 * Added t38pt_usertpsource option. See sip.conf.sample for details.
428 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
429 * 'sip show peers' and 'sip show users' display their entries sorted in
430 alphabetical order, as opposed to the order they were in, in the config
432 * Videosupport now supports an additional option, "always", which always sets
433 up video RTP ports, even on clients that don't support it. This helps with
434 callfiles and certain transfers to ensure that if two video phones are
435 connected, they will always share video feeds.
439 * Existing DNS manager lookups extended to check for SRV records.
440 * IAX2 encryption support has been improved to support periodic key rotation
441 within a call for enhanced security. The option "keyrotate" has been
442 provided to disable this functionality to preserve backwards compatibility
443 with older versions of IAX2 that do not support key rotation.
447 * New CLI command, "config reload <file.conf>" which reloads any module that
448 references that particular configuration file. Also added "config list"
449 which shows which configuration files are in use.
450 * New CLI commands, "pri show version" and "ss7 show version" that will
451 display which version of libpri and libss7 are being used, respectively.
452 A new API call was added so trunk will now have to be compiled against
453 a versions of libpri and libss7 that have them or it will not know that
454 these libraries exist.
455 * The commands "core show globals", "core set global" and "core set chanvar" has
456 been deprecated in favor of the more semanticly correct "dialplan show globals",
457 "dialplan set chanvar" and "dialplan set global".
458 * New CLI command "dialplan show chanvar" to list all variables associated
459 with a given channel.
463 * Addresses managed by DNS manager now can check to see if there is a DNS
464 SRV record for a given domain and will use that hostname/port if present.
466 AMI - The manager (TCP/TLS/HTTP)
467 --------------------------------
468 * The Status command now takes an optional list of variables to display
469 along with channel status.
470 * The QueueEntry event now also includes the channel's uniqueid
474 * res_odbc no longer has a limit of 1023 total possible unshared connections,
475 as some people were running into this limit. This limit has been increased
480 * The TRANSFER queue log entry now includes the the caller's original
481 position in the transferred-from queue.
482 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
483 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
484 as well as an explanation about timeout options in general
485 * Added a new option - C - for forcing the "answered elsewhere" flag on
486 cancellation of calls in to members of the queue. This is to avoid the
487 call to a member of a queue having the call listed as a "missed call".
491 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
492 adaptive capabilities. What this means in practical terms is that if your
493 realtime table lacks critical fields, Asterisk will now emit warnings to
494 that effect. Also, some of the realtime drivers have the ability (if
495 configured) to automatically add those columns to the table with the
496 correct type and length.
500 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
501 the 'setvar' option to cause a given audio file to be played upon completion
502 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
503 Skinny channels only.
504 * You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt
505 for more information.
506 * Config file variables may now be appended to, by using the '+=' append
507 operator. This is most helpful when working with long SQL queries in
508 func_odbc.conf, as the queries no longer need to be specified on a single
510 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
511 which will add a second to the billsec when the ending
512 time is set, if the number in the microseconds field of the end time is
513 greater than the number of microseconds in the answer time. This allows
514 users to count the 'initiated' seconds in their billing records.
516 ------------------------------------------------------------------------------
517 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
518 ------------------------------------------------------------------------------
520 AMI - The manager (TCP/TLS/HTTP)
521 --------------------------------
522 * Manager has undergone a lot of changes, all of them documented
523 in doc/manager_1_1.txt
524 * Manager version has changed to 1.1
525 * Added a new action 'CoreShowChannels' to list currently defined channels
526 and some information about them.
527 * Added a new action 'SIPshowregistry' to list SIP registrations.
528 * Added TLS support for the manager interface and HTTP server
529 * Added the URI redirect option for the built-in HTTP server
530 * The output of CallerID in Manager events is now more consistent.
531 CallerIDNum is used for number and CallerIDName for name.
532 * Enable https support for builtin web server.
533 See configs/http.conf.sample for details.
534 * Added a new action, GetConfigJSON, which can return the contents of an
535 Asterisk configuration file in JSON format. This is intended to help
536 improve the performance of AJAX applications using the manager interface
538 * SIP and IAX manager events now use "ChannelType" in all cases where we
539 indicate channel driver. Previously, we used a mixture of "Channel"
540 and "ChannelDriver" headers.
541 * Added a "Bridge" action which allows you to bridge any two channels that
542 are currently active on the system.
543 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
544 the voicemail users setup.
545 * Added 'DBDel' and 'DBDelTree' manager commands.
546 * cdr_manager now reports events via the "cdr" level, separating it from
547 the very verbose "call" level.
548 * Manager users are now stored in memory. If you change the manager account
549 list (delete or add accounts) you need to reload manager.
550 * Added Masquerade manager event for when a masquerade happens between
552 * Added "manager reload" command for the CLI
553 * Lots of commands that only provided information are now allowed under the
554 Reporting privilege, instead of only under Call or System.
555 * The IAX* commands now require either System or Reporting privilege, to
556 mirror the privileges of the SIP* commands.
557 * Added ability to retrieve list of categories in a config file.
558 * Added ability to retrieve the content of a particular category.
559 * Added ability to empty a context.
560 * Created new action to create a new file.
561 * Updated delete action to allow deletion by line number with respect to category.
562 * Added new action insert to add new variable to category at specified line.
563 * Updated action newcat to allow new category to be inserted in file above another
565 * Added new event "JitterBufStats" in the IAX2 channel
566 * Originate now requires the Originate privilege and, if you want to call out
567 to a subshell, it requires the System privilege, as well. This was done to
568 enhance manager security.
569 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
570 * New command: Atxfer. See doc/manager_1_1.txt for more details or
571 manager show command Atxfer from the CLI
572 * New command: IAXregistry. See doc/manager_1_1.txt for more details or
573 manager show command IAXregistry from the CLI
577 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
578 state in the dialplan, as well as creating custom device states that are
579 controllable from the dialplan.
580 * Extend CALLERID() function with "pres" and "ton" parameters to
581 fetch string representation of calling number presentation indicator
582 and numeric representation of type of calling number value.
583 * MailboxExists converted to dialplan function
584 * A new option to Dial() for telling IP phones not to count the call
585 as "missed" when dial times out and cancels.
586 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
587 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
588 held for any given channel. Also, locks are automatically freed when a
590 * Added HINT() dialplan function that allows retrieving hint information.
591 Hints are mappings between extensions and devices for the sake of
592 determining the state of an extension. This function can retrieve the list
593 of devices or the name associated with a hint.
594 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
596 * Added SYSINFO() dialplan function which allows retrieval of system information
597 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
598 the existence of a dialplan target.
599 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
600 upper and lower case, respectively.
601 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
602 ID for the call (not the Asterisk call ID or unique ID), provided that the
603 channel driver supports this. For SIP, you get the SIP call-ID for the
604 bridged channel which you can store in the CDR with a custom field.
608 * Added CLI permissions, config file: cli_permissions.conf
609 default is to allow all commands for every local user/group.
610 Also this new feature added three new CLI commands:
611 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
612 - cli reload permissions
613 - cli show permissions
614 * New CLI command "core show hint" (usage: core show hint <exten>)
615 * New CLI command "core show settings"
616 * Added 'core show channels count' CLI command.
617 * Added the ability to set the core debug and verbose values on a per-file basis.
618 * Added 'queue pause member' and 'queue unpause member' CLI commands
619 * Ability to set process limits ("ulimit") without restarting Asterisk
620 * Enhanced "agi debug" to print the channel name as a prefix to the debug
621 output to make debugging on busy systems much easier.
622 * New CLI commands "dialplan set extenpatternmatching true/false"
623 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
624 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
625 listed in the startup_commands section of cli.conf will get executed.
626 * Added a CLI command, "devstate change", which allows you to set custom device
627 states from the func_devstate module that provides the DEVICE_STATE() function
628 and handling of the "Custom:" devices.
629 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
630 sorted into the different possible callbacks, with the number of entries
631 currently scheduled for each. Gives you a feel for how busy the sip channel
633 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
634 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
635 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
639 * Improved NAT and STUN support.
640 chan_sip now can use port numbers in bindaddr, externip and externhost
641 options, as well as contact a STUN server to detect its external address
642 for the SIP socket. See sip.conf.sample, 'NAT' section.
643 * The default SIP useragent= identifier now includes the Asterisk version
644 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
645 If set, and the incoming request carries authentication info,
646 the username to match in the users list is taken from the Digest header
647 rather than from the From: field. This feature is considered experimental.
648 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
649 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
650 * The "localmask" setting was removed in version 1.2 and the reminder about it
651 being removed is now also removed.
652 * A new option "busylevel" for setting a level of calls where asterisk reports
653 a device as busy, to separate it from call-limit. This value is also added
654 to the SIP_PEER dialplan function.
655 * A new realtime family called "sipregs" is now supported to store SIP registration
656 data. If this family is defined, "sippeers" will be used for configuration and
657 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
658 registration data, as before.
659 * The SIPPEER function have new options for port address, call and pickup groups
660 * Added support for T.140 realtime text in SIP/RTP
661 * The "checkmwi" option has been removed from sip.conf, as it is no longer
662 required due to the restructuring of how MWI is handled. See the descriptions
663 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
664 for more information.
665 * Added rtpdest option to CHANNEL() dialplan function.
666 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
667 * SIP now adds a header to the CANCEL if the call was answered by another phone
668 in the same dial command, or if the new c option in dial() is used.
669 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
670 states it is not needed. For phones, however, that do require it the "registertrying" option
671 has been added so it can be enabled.
672 * A new option called "callcounter" (global/peer/user level) enables call counters needed
673 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
674 used to enable this functionality).
675 * New settings for timer T1 and timer B on a global level or per device. This makes it
676 possible to force timeout faster on non-responsive SIP servers. These settings are
677 considered advanced, so don't use them unless you have a problem.
678 * Added a dial string option to be able to set the To: header in an INVITE to any
680 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
681 the qualify frequency.
682 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
683 were not properly torn down due to network or endpoint failures during an established
685 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
686 configs/sip.conf.sample for more information on how it is used.
687 * Added a new configuration option "authfailureevents" that enables manager events when
688 a peer can't authenticate properly.
689 * Added DNS manager support to registrations for peers not referencing a peer entry.
693 * Added the trunkmaxsize configuration option to chan_iax2.
694 * Added the srvlookup option to iax.conf
695 * Added support for OSP. The token is set and retrieved through the CHANNEL()
698 XMPP Google Talk/Jingle changes
699 -------------------------------
700 * Added the bindaddr option to gtalk.conf.
704 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
705 * Proper codec support in chan_skinny.
706 * Added settings for IP and Ethernet QoS requests
710 * Added separate settings for media QoS in mgcp.conf
712 Console Channel Driver changes
713 ------------------------------
714 * Added experimental support for video send & receive to chan_oss.
715 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
718 Phone channel changes (chan_phone)
719 ----------------------------------
720 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
722 H.323 channel Changes
723 ---------------------
724 * H323 remote hold notification support added (by NOTIFY message
725 and/or H.450 supplementary service)
727 Local channel changes
728 ---------------------
729 * The device state functionality in the Local channel driver has been updated
730 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
731 to just UNKNOWN if the extension exists.
732 * Added jitterbuffer support for chan_local. This allows you to use the
733 generic jitterbuffer on incoming calls going to Asterisk applications.
734 For example, this would allow you to use a jitterbuffer for an incoming
735 SIP call to Voicemail by putting a Local channel in the middle. This
736 feature is enabled by using the 'j' option in the Dial string to the Local
737 channel in conjunction with the existing 'n' option for local channels.
738 * A 'b' option has been added which causes chan_local to return the actual channel
739 that is behind it when queried. This is useful for transfer scenarios as the
740 actual channel will be transferred, not the Local channel.
742 Agent channel changes
743 ----------------------
744 * The ackcall and endcall options are now supplemented with options acceptdtmf
745 and enddtmf. These allow for the DTMF keypress to be configurable. The options
746 default to their old hard-coded values ('#' and '*' respectively) so this should
747 not break any existing agent installations.
749 DAHDI channel driver (chan_dahdi) Changes
750 ----------------------------------------
751 * SS7 support (via libss7 library)
752 * In India, some carriers transmit CID via dtmf. Some code has been added
753 that will handle some situations. The cidstart=polarity_IN choice has been added for
754 those carriers that transmit CID via dtmf after a polarity change.
755 * CID matching information is now shown when doing 'dialplan show'.
756 * Added dahdi show version CLI command.
757 * Added setvar support to chan_dahdi.conf channel entries.
758 * Added two new options: mwimonitor and mwimonitornotify. These options allow
759 you to enable MWI monitoring on FXO lines. When the MWI state changes,
760 the script specified in the mwimonitornotify option is executed. An internal
761 event indicating the new state of the mailbox is also generated, so that
762 the normal MWI facilities in Asterisk work as usual.
763 * Added signalling type 'auto', which attempts to use the same signalling type
764 for a channel as configured in DAHDI. This is primarily designed for analog
765 ports, but will also work for digital ports that are configured for FXS or FXO
766 signalling types. This mode is also the default now, so if your chan_dahdi.conf
767 does not specify signalling for a channel (which is unlikely as the sample
768 configuration file has always recommended specifying it for every channel) then
769 the 'auto' mode will be used for that channel if possible.
770 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
771 state for a channel; also ensured that the DNDState Manager event is
772 emitted no matter how the DND state is set or cleared.
776 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
777 configs/unistim.conf.sample for details. This new channel driver allows
778 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
779 * Added a new channel driver, chan_console, which uses portaudio as a cross
780 platform audio interface. It was written as a channel driver that would
781 work with Mac CoreAudio, but portaudio supports a number of other audio
782 interfaces, as well. Note that this channel driver requires v19 or higher
783 of portaudio; older versions have a different API.
787 * Added the ability to specify arguments to the Dial application when using
788 the DUNDi switch in the dialplan.
789 * Added the ability to set weights for responses dynamically. This can be
790 done using a global variable or a dialplan function. Using the SHELL()
791 function would allow you to have an external script set the weight for
793 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
794 functions will allow you to initiate a DUNDi query from the dialplan,
795 find out how many results there are, and access each one.
799 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
800 functions will allow you to initiate an ENUM lookup from the dialplan,
801 and Asterisk will cache the results. ENUMRESULT can be used to access
802 the results without doing multiple DNS queries.
806 * Added the ability to customize which sound files are used for some of the
807 prompts within the Voicemail application by changing them in voicemail.conf
808 * Added the ability for the "voicemail show users" CLI command to show users
809 configured by the dynamic realtime configuration method.
810 * MWI (Message Waiting Indication) handling has been significantly
811 restructured internally to Asterisk. It is now totally event based
812 instead of polling based. The voicemail application will notify other
813 modules that have subscribed to MWI events when something in the mailbox
815 This also means that if any other entity outside of Asterisk is changing
816 the contents of mailboxes, then the voicemail application still needs to
817 poll for changes. Examples of situations that would require this option
818 are web interfaces to voicemail or an email client in the case of using
819 IMAP storage. So, two new options have been added to voicemail.conf
820 to account for this: "pollmailboxes" and "pollfreq". See the sample
821 configuration file for details.
822 * Added "tw" language support
823 * Added support for storage of greetings using an IMAP server
824 * Added ability to customize forward, reverse, stop, and pause keys for message playback
825 * SMDI is now enabled in voicemail using the smdienable option.
826 * A "lockmode" option has been added to asterisk.conf to configure the file
827 locking method used for voicemail, and potentially other things in the
828 future. The default is the old behavior, lockfile. However, there is a
829 new method, "flock", that uses a different method for situations where the
830 lockfile will not work, such as on SMB/CIFS mounts.
831 * Added the ability to backup deleted messages, to ease recovery in the case
832 that a user accidentally deletes a message, and discovers that they need it.
833 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
834 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
835 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
836 voicemail boxes. The SMDI interface can also poll for MWI changes when some
837 outside entity is modifying the state of the mailbox (such as IMAP storage or
838 a web interface of some kind).
839 * Added the support for marking messages as "urgent." There are two methods to accomplish
840 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
841 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
842 the message as urgent after he has recorded a voicemail by following the voice instructions.
843 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
848 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
849 used across multiple queues.
850 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
851 setqueueentryvar options for each queue, see queues.conf.sample for details.
852 * Added keepstats option to queues.conf which will keep queue
853 statistics during a reload.
854 * setinterfacevar option in queues.conf also now sets a variable
855 called MEMBERNAME which contains the member's name.
856 * Added 'Strategy' field to manager event QueueParams which represents
857 the queue strategy in use.
858 * Added option to run macro when a queue member is connected to a caller,
859 see queues.conf.sample for details.
860 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
861 does not count paused queue members as unavailable.
862 * Added min-announce-frequency option to queues.conf which allows you to control the
863 minimum amount of time between queue announcements for use when the caller's queue
864 position changes frequently.
865 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
867 * Added ability for non-realtime queues to have realtime members
868 * Added the "linear" strategy to queues.
869 * Added the "wrandom" strategy to queues.
870 * Added new channel variable QUEUE_MIN_PENALTY
871 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
872 rules in queuerules.conf. See configs/queuerules.conf.sample for details
873 * Added a new parameter for member definition, called state_interface. This may be
874 used so that a member may be called via one interface but have a different interface's
875 device state reported.
876 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
877 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
878 "manager show command QueueReset."
879 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
880 specified by the periodic-announce option, then one will be chosen randomly when it is time
881 to play a periodic announcment
882 * New configuration options: announce-position now takes two more values in addition to "yes" and
883 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
884 announce-position-limit. By setting announce-position to "limit" callers will only have their
885 position announced if their position is less than what is specified by announce-position-limit.
886 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
887 will be told that their are more than announce-position-limit callers waiting.
888 * Two new queue log events have been added. An ADDMEMBER event will be logged
889 when a realtime queue member is added and a REMOVEMEMBER event will be logged
890 when a realtime queue member is removed. Since there is no calling channel associated
891 with these events, the string "REALTIME" is placed where the channel's unique id
893 * The configuration method for the "joinempty" and "leavewhenempty" options has
894 changed to a comma-separated list of methods of determining member availability
895 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
896 values are still accepted for backwards-compatibility, though.
897 * The average talktime is now calculated on queues. This information is reported via the
898 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
899 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
904 * The 'o' option to provide an optimization has been removed and its functionality
905 has been enabled by default.
906 * When a conference is created, the UNIQUEID of the channel that caused it to be
907 created is stored. Then, every channel that joins the conference will have the
908 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
909 callers that come and go from long standing conferences.
910 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
911 except it does operations on a channel by name, instead of number in a conference.
912 This is a very useful feature in combination with the 'X' option to ChanSpy.
913 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
915 * Added new RealTime functionality to provide support for scheduled conferencing.
916 This includes optional messages to the caller if they attempt to join before
917 the schedule start time, or to allow the caller to join the conference early.
918 Also included is optional support for limiting the number of callers per
920 * Added the S() and L() options to the MeetMe application. These are pretty
921 much identical to the S() and L() options to Dial(). They let you set
922 timeouts for the conference, as well as have warning sounds played to
923 let the caller know how much time is left, and when it is running out.
924 * Added the ability to do "meetme concise" with the "meetme" CLI command.
925 This extends the concise capabilities of this CLI command to include
926 listing all conferences, instead of an addition to the other sub commands
927 for the "meetme" command.
928 * Added the ability to specify the music on hold class used to play into the
929 conference when there is only one member and the M option is used.
930 * Added MEETME_INFO dialplan function which provides a way to query
931 various properties of a Meetme conference.
933 Other Dialplan Application Changes
934 ----------------------------------
935 * Argument support for Gosub application
936 * From the to-do lists: straighten out the app timeout args:
937 Wait() app now really does 0.3 seconds- was truncating arg to an int.
938 WaitExten() same as Wait().
939 Congestion() - Now takes floating pt. argument.
940 Busy() - now takes floating pt. argument.
941 Read() - timeout now can be floating pt.
942 WaitForRing() now takes floating pt timeout arg.
943 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
944 * Added 's' option to Page application.
945 * Added an optional timeout argument to the Page application.
946 * Added 'E', 'V', and 'P' commands to ExternalIVR.
947 * Added 'o' and 'X' options to Chanspy.
948 * Added a new dialplan application, Bridge, which allows you to bridge the
949 calling channel to any other active channel on the system.
950 * Added the ability to specify a music on hold class to play instead of ringing
951 for the SLATrunk application.
952 * The Read application no longer exits the dialplan on error. Instead, it sets
953 READSTATUS to ERROR, which you can catch and handle separately.
954 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
955 of asking for verification of each name, one at a time.
956 * Privacy() no longer uses privacy.conf, as all options are specifyable as
957 direct options to the app.
958 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
960 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
961 * The ChannelRedirect application no longer exits the dialplan if the given channel
962 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
963 or NOCHANNEL if the given channel was not found.
964 * The silencethreshold setting that was previously configurable in multiple
965 applications is now settable globally via dsp.conf.
967 Music On Hold Changes
968 ---------------------
969 * A new option, "digit", has been added for music on hold classes in
970 musiconhold.conf. If this is set for a music on hold class, a caller
971 listening to music on hold can press this digit to switch to listening
972 to this music on hold class.
973 * Support for realtime music on hold has been added.
974 * In conjunction with the realtime music on hold, a general section has
975 been added to musiconhold.conf, its sole variable is cachertclasses. If this
976 is set, then music on hold classes found in realtime will be cached in memory.
980 * AEL upgraded to use the Gosub with Arguments instead
981 of Macro application, to hopefully reduce the problems
982 seen with the artificially low stack ceiling that
983 Macro bumps into. Macros can only call other Macros
984 to a depth of 7. Tests run using gosub, show depths
985 limited only by virtual memory. A small test demonstrated
986 recursive call depths of 100,000 without problems.
987 -- in addition to this, all apps that allowed a macro
988 to be called, as in Dial, queues, etc, are now allowing
989 a gosub call in similar fashion.
990 * AEL now generates LOCAL(argname) declarations when it
991 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
992 etc. That makes the arguments local in scope. The user
993 can define their own local variables in macros, now,
994 by saying "local myvar=someval;" or using Set() in this
995 fashion: Set(LOCAL(myvar)=someval); ("local" is now
997 * utils/conf2ael introduced. Will convert an extensions.conf
998 file into extensions.ael. Very crude and unfinished, but
999 will be improved as time goes by. Should be useful for a
1000 first pass at conversion.
1001 * aelparse will now read extensions.conf to see if a referenced
1002 macro or context is there before issueing a warning.
1003 * AEL parser sets a local channel variable ~~EXTEN~~, to
1004 preserve the value of ${EXTEN} thru switch statements.
1005 * New operator in $[...] expressions: the ~~ operator serves
1006 as a concatenation operator. AT THE MOMENT, it is really only
1007 necessary and useful in AEL, especially in if() expressions.
1008 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
1009 any enclosing double-quotes, and evaluate to the value of a
1010 concatenated with the value of b. For example if a is set to
1011 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
1012 evaluate to xyzabc .
1015 Call Features (res_features) Changes
1016 ------------------------------------
1017 * Added the parkedcalltransfers option to features.conf
1018 * Added parkedcallparking option to control one touch parking w/ parking
1020 * Added parkedcallhangup option to control disconnect feature w/ parking
1022 * Added parkedcallrecording option to control one-touch record w/ parking
1024 * Added BRIDGE_FEATURES variable to set available features for a channel
1025 * The built-in method for doing attended transfers has been updated to
1026 include some new options that allow you to have the transferee sent
1027 back to the person that did the transfer if the transfer is not successful.
1028 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
1029 in features.conf.sample.
1030 * Added support for configuring named groups of custom call features in
1031 features.conf. This means that features can be written a single time, and
1032 then mapped into groups of features for different key mappings or easier
1034 * Updated the ParkedCall application to allow you to not specify a parking
1035 extension. If you don't specify a parking space to pick up, it will grab
1036 the first one available.
1037 * Added cli command 'features reload' to reload call features from features.conf
1038 * Moved into core asterisk binary.
1039 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
1041 Language Support Changes
1042 ------------------------
1043 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
1044 * Added support for the Hungarian language for saying numbers, dates, and times.
1048 * Added SPEECH commands for speech recognition. A complete listing can be found
1050 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
1051 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
1052 does not behave as expected; the native command needs to be used, instead.
1056 * Added rotatestrategy option to logger.conf, along with two new options:
1057 "timestamp" which will use the time to name the logger files instead of
1058 sequence number; and "rotate", which rotates the names of the logfiles,
1059 similar to the way syslog rotates files.
1060 * Added exec_after_rotate option to logger.conf, which allows a system
1061 command to be run after rotation. This is primarily useful with
1062 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
1063 and to ensure that the oldest log file gets deleted.
1064 * Added realtime support for the queue log
1068 * The cdr_manager module has a [mappings] feature, like cdr_custom,
1069 to add fields to the manager event from the CDR variables.
1070 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
1071 backend database CDR table. Specifically, additional, non-standard
1072 columns are supported, merely by setting the corresponding CDR variable in
1073 your dialplan. In addition, you may alias any column to another name (for
1074 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
1075 simply "alias src => ANI" in the configuration file). Records may be
1076 posted to more than one backend, simply by specifying multiple categories
1077 in the configuration file. And finally, you may filter which CDRs get
1078 posted to each backend, by specifying a filter (which the record must
1079 match) for the particular category. Filters are additive (meaning all
1080 rules must match to post that CDR).
1081 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
1082 module. Specifically, you may add additional columns into the table and
1083 they will be set, if you set the corresponding CDR variable name. Also,
1084 if you omit columns in your database table, they will be silently skipped
1085 (but a record will still be inserted, based on what columns remain). Note
1086 that the other two features from cdr_adaptive_odbc (alias and filter) are
1087 not currently supported.
1088 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
1089 has been disabled using the NoCDR application.
1091 Miscellaneous New Modules
1092 -------------------------
1093 * Added a new CDR module, cdr_sqlite3_custom.
1094 * Added a new realtime configuration module, res_config_sqlite
1095 * Added a new codec translation module, codec_resample, which re-samples
1096 signed linear audio between 8 kHz and 16 kHz to help support wideband
1098 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
1099 based on configuration templates that use Asterisk dialplan function and
1100 variable substitution. It should be possible to create phone profiles and
1101 templates that work for the majority of phones provisioned over http. It
1102 is currently only intended to provision a single user account per phone.
1103 An example profile and set of templates for Polycom phones is provided.
1104 NOTE: Polycom firmware is not included, but should be placed in
1105 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
1106 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
1107 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
1108 provided; there is a JACK() application, and a JACK_HOOK() function. Both
1109 interfaces create an input and output JACK port. The application makes
1110 these ports the endpoint of the call. The audio coming from the channel
1111 goes out the output port and whatever comes back in on the input port is
1112 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
1113 audiohook on the channel. This lets you run the audio coming from a
1114 channel through JACK, and whatever comes back in is what gets forwarded
1115 on as the channel's audio. This is very useful for building custom
1116 vocoders or doing recording or analysis of the channel's audio in another
1118 * Added a new module, res_config_curl, which permits using a HTTP POST url
1119 to retrieve, create, update, and delete realtime information from a remote
1120 web server. Note that this module requires func_curl.so to be loaded for
1121 backend functionality.
1122 * Added a new module, res_config_ldap, which permits the use of an LDAP
1123 server for realtime data access.
1124 * Added support for writing and running your dialplan in lua using the pbx_lua
1125 module. See configs/extensions.lua.sample for examples of how to do this.
1129 * Ability to use libcap to set high ToS bits when non-root
1130 on Linux. If configure is unable to find libcap then you
1131 can use --with-cap to specify the path.
1132 * Added maxfiles option to options section of asterisk.conf which allows you to specify
1133 what Asterisk should set as the maximum number of open files when it loads.
1134 * Added the jittertargetextra configuration option.
1135 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
1136 configuration files for the IP channel drivers. The new option is "cos".
1137 This information is also documented in doc/qos.tex, or the IP Quality of Service
1138 section of asterisk.pdf.
1139 * When originating a call using AMI or pbx_spool that fails the reason for failure
1140 will now be available in the failed extension using the REASON dialplan variable.
1141 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
1142 It allows you to configure a prefix for auto-monitor recordings.
1143 * A new extension pattern matching algorithm, based on a trie, is introduced
1144 here, that could noticeably speed up mid-sized to large dialplans.
1145 It is NOT used by default, as duplicating the behaviour of the old pattern
1146 matcher is still under development. A config file option, in extensions.conf,
1147 in the [general] section, called "extenpatternmatchingnew", is by default
1148 set to false; setting that to true will force the use of the new algorithm.
1149 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
1150 be used to switch the algorithms at run time.
1151 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
1152 specifying which socket to use to connect to the running Asterisk daemon
1154 * Performance enhancements to the sched facility, which is used in
1155 the channel drivers, etc. Added hashtabs and doubly-linked lists
1156 to speed up deletion; start at the beginning or end of list to
1158 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
1159 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
1160 Added regression tests to the tests/ dir, also.
1161 * Added a refcount trace feature to astobj2 for those trying to balance
1162 object creation, deletion; work, play; space and time. See the
1163 notes in astobj2.h. Also, see utils/refcounter as well, as a
1164 quick way to find unbalanced refcounts in what could be a sea
1165 of objects that were balanced.
1166 * Added logging to 'make update' command. See update.log
1167 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
1168 do not come from the remote party.
1169 * Added the 'n' option to the SpeechBackground application to tell it to not
1170 answer the channel if it has not already been answered.
1171 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
1172 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
1174 * iLBC source code no longer included (see UPGRADE.txt for details)
1175 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
1176 deadlock is detected, a backtrace of the stack which led to the lock calls
1177 will be output to the CLI.
1178 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
1179 the "core show locks" CLI command will give lock information output as well
1180 as a backtrace of the stack which led to the lock calls.
1181 * users.conf now sports an optional alternateexts property, which permits
1182 allocation of additional extensions which will reach the specified user.
1183 * A new option for the configure script, --enable-internal-poll, has been added
1184 for use with systems which may have a buggy implementation of the poll system
1185 call. If you notice odd behavior such as the CLI being unresponsive on remote
1186 consoles, you may want to try using this option. This option is enabled by default
1187 on Darwin systems since it is known that the Darwin poll() implementation has
1191 --------------------
1192 * In addition to timing from DAHDI, there is a new timing module called
1193 res_timing_timerfd. In order to use this, you must be running Linux with
1194 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
1195 script will be able to tell if you have the requirements. From menuselect, select
1196 res_timing_timerfd from the Resource Modules menu.
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