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 ==============================================================================
11 ------------------------------------------------------------------------------
12 --- Functionality changes from Asterisk 13 to Asterisk 14 --------------------
13 ------------------------------------------------------------------------------
20 * A new application in Asterisk, this will join the calling channel
21 to an existing bridge containing the named channel prefix.
25 * Added the ability to pass options to MixMonitor when recording is used with
26 ConfBridge. This includes the addition of the following configuration
27 parameters for the 'bridge' object:
28 - record_file_timestamp: whether or not to append the start time to the
30 - record_options: the options to pass to the MixMonitor application
31 - record_command: a command to execute when recording is finished
32 Note that these options may also be with the CONFBRIDGE function.
36 * Added the 'n' option, which prevents the SMS from being written to the log
37 file. This is needed for those countries with privacy laws that require
38 providers to not log SMS content.
45 * Added a new configuration option, "newcdrcolumns", which enables use of the
46 post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.
51 * Added a new configuration option, "newcdrcolumns", which enables use of the
52 post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.
60 * The CALLERID(ani2) value for incoming calls is now populated in featdmf
61 signaling mode. The information was previously discarded.
62 * Added the force_restart_unavailable_chans compatibility option. When
63 enabled it causes Asterisk to restart the ISDN B channel if an outgoing
64 call receives cause 44 (Requested channel not available).
68 * The iax.conf forcejitterbuffer option has been removed. It is now always
69 forced if you set iax.conf jitterbuffer=yes. If you put a jitter buffer
70 on a channel it will be on the channel.
71 * A new configuration parameters, 'calltokenexpiration', has been added that
72 controls the duration before a call token expires. Default duration is 10
73 seconds. Setting this to a higher value may help in lagged networks or those
74 experiencing high packet loss.
78 * New 'rtpbindaddr' global setting. This allows a user to define which
79 ipaddress to bind the rtpengine to. For example, chan_sip might bind
80 to eth0 (10.0.0.2) but rtpengine to eth1 (192.168.1.10).
81 * DTLS related configuration options can now be set at a general level.
82 Enabling DTLS support, though, requires enabling it at the user
87 * New 'user_eq_phone' endpoint setting. This adds a 'user=phone' parameter
88 to the request URI and From URI if the user is determined to be a phone number.
89 * New 'moh_passthrough' endpoint setting. This will pass hold and unhold requests
90 through using SIP re-invites with sendonly and sendrecv accordingly.
91 * Added the pjsip.conf system type disable_tcp_switch option. The option
92 allows the user to disable switching from UDP to TCP transports described
93 by RFC 3261 section 18.1.1.
94 * New 'line' and 'endpoint' options added on outbound registrations. This allows some
95 identifying information to be added to the Contact of the outbound registration.
96 If this information is present on messages received from the remote server
97 the message will automatically be associated with the configured endpoint on the
98 outbound registration.
102 * The core of Asterisk uses a message bus called "Stasis" to distribute
103 information to internal components. For performance reasons, the message
104 distribution was modified to make use of a thread pool instead of a
105 dedicated thread per consumer in certain cases. The initial settings for
106 the thread pool can now be configured in 'stasis.conf'.
108 * A new core DNS API has been implemented which provides a common interface
109 for DNS functionality. Modules that use this functionality will require that
110 a DNS resolver module is loaded and available.
112 * Modified processing of command-line options to first parse only what
113 is necessary to read asterisk.conf. Once asterisk.conf is fully loaded,
114 the remaining options are processed. The -X option now applies to
115 asterisk.conf only. To enable #exec for other config files you must
116 set execincludes=yes in asterisk.conf. Any other option set on the
117 command-line will now override the equivalent setting from asterisk.conf.
119 * The TLS core in Asterisk now supports X.509 certificate subject alternative
120 names. This way one X.509 certificate can be used for hosts that can be
121 reached under multiple DNS names or for multiple hosts.
123 * The Asterisk logging system now supports JSON structured logging. Log
124 channels specified in logger.conf or added dynamically via CLI commands now
125 support an optional specifier prior to their levels that determines their
126 formatting. To set a log channel to format its entries as JSON, a formatter
127 of '[json]' can be set, e.g.,
128 full => [json]debug,verbose,notice,warning,error
135 * Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for
136 the hold status of a channel.
140 * The transferdialattempts default value has been changed from 1 to 3. The
141 transferinvalidsound has been changed from "pbx-invalid" to "privacy-incorrect".
142 These were changed to make DTMF transfers be more user-friendly by default.
150 * Added sort=randstart to the sort options. It sorts the files by name and
151 then chooses the first file to play at random.
152 * Added preferchannelclass=no option to prefer the application-passed class
153 over the channel-set musicclass. This allows separate hold-music from
154 application (e.g. Queue or Dial) specified music.
158 * Added a res_resolver_unbound module which uses the libunbound resolver library
159 to perform DNS resolution. This module requires the libunbound library to be
160 installed in order to be used.
164 * A new SIP resolver using the core DNS API has been implemented. This relies on
165 external SIP resolver support in PJSIP which is only available as of PJSIP
166 2.4. If this support is unavailable the existing built-in PJSIP SIP resolver
167 will be used instead. The new SIP resolver provides NAPTR support, improved
168 SRV support, and AAAA record support.
170 res_pjsip_outbound_registration
171 -------------------------------
172 * A new 'fatal_retry_interval' option has been added to outbound registration.
173 When set (default is zero), and upon receiving a failure response to an
174 outbound registration, registration is retried at the given interval up to
182 * Added a new option, 'usegmtime', which causes timestamps in CEL events
185 * Added support to set schema where located the table cel. This settings is
186 configurable for cel_pgsql via the 'schema' in configuration file
194 * Added the ability to set the character to quote identifiers. This
195 allows adding the character at the start and end of table and column
196 names. This setting is configurable for cdr_adaptive_odbc via the
197 quoted_identifiers in configuration file cdr_adaptive_odbc.conf.
201 * Added field ReasonPause on QueueMemberStatus if set when paused, the reason
202 the queue member was paused.
204 ------------------------------------------------------------------------------
205 --- Functionality changes from Asterisk 13.7.0 to Asterisk 13.8.0 ------------
206 ------------------------------------------------------------------------------
210 * This module is the successor of res_pjsip_log_forwarder. As well as
211 handling the log forwarding (which now displays as 'pjproject:0' instead
212 of 'pjsip:0'), it also adds a 'pjproject show buildopts' command to the CLI.
213 This displays the compiled-in options of the pjproject installation
214 Asterisk is currently running against.
219 * Added new global option (regcontext) to pjsip. When set, Asterisk will
220 dynamically create and destroy a NoOp priority 1 extension
221 for a given endpoint who registers or unregisters with us.
225 * A new module, res_pjsip_history, has been added that provides SIP history
226 viewing/filtering from the CLI. The module is intended to be used on systems
227 with busy SIP traffic, where existing forms of viewing SIP messages - such
228 as the res_pjsip_logger - may be inadequate. The module provides two new
230 - 'pjsip set history {on|off|clear}' - this enables/disables SIP history
231 capturing, as well as clears an existing history capture. Note that SIP
232 packets captured are stored in memory until cleared. As a result, the
233 history capture should only be used for debugging/viewing purposes, and
234 should *NOT* be left permanently enabled on a system.
235 - 'pjsip show history' - displays the captured SIP history. When invoked
236 with no options, the entire captured history is displayed. Two options
238 -- 'entry <num>' - display a detailed view of a single SIP message in
240 -- 'where ...' - filter the history based on some expression. For more
241 information on filtering, view the current CLI help for the
242 'pjsip show history' command.
246 * app_voicemail and res_mwi_external can now be built together. The default
247 remains to build app_voicemail and not res_mwi_external but if they are
248 both built, the load order will cause res_mwi_external to load first and
249 app_voicemail will be skipped. Use 'preload=app_voicemail.so' in
250 modules.conf to force app_voicemail to be the voicemail provider.
254 * A new option (bind_rtp_to_media_address) has been added to endpoint which
255 will cause res_pjsip_sdp_rtp to actually bind the RTP instance to the
256 media_address as well as using it in the SDP. If set, RTP packets will now
257 originate from the media address instead of the operating system's "primary"
260 ------------------------------------------------------------------------------
261 --- Functionality changes from Asterisk 13.6.0 to Asterisk 13.7.0 ------------
262 ------------------------------------------------------------------------------
266 * Added format attribute negotiation for the VP8 video codec. Format attribute
267 negotiation is provided by the res_format_attr_vp8 module.
271 * A new "timeout" user profile option has been added. This configures the number
272 of seconds that a participant may stay in the ConfBridge after joining. When
273 the time expires, the user is ejected from the conference and CONFBRIDGE_RESULT
274 is set to "TIMEOUT" on the channel.
278 * The websockets_enabled option has been added to the general section of
279 sip.conf. The option is enabled by default to match the previous behavior.
280 The option should be disabled when using res_pjsip_transport_websockets to
281 ensure chan_sip will not conflict with PJSIP websockets.
285 * The HOLD_INTERCEPT dialplan function now actually exists in the source tree.
286 While support for the events was added in Asterisk 13.4.0, the function
287 accidentally never made it in. That function is now present, and will cause
288 the 'hold' raised by a channel to be intercepted and converted into an
291 res_pjsip_outbound_registration
292 -------------------------------
293 * If res_statsd is loaded and a StatsD server is configured, basic statistics
294 regarding the state of outbound registrations will now be emitted. This
296 - A GAUGE statistic for the overall number of outbound registrations, i.e.:
297 PJSIP.registrations.count
298 - A GAUGE statistic for the overall number of outbound registrations in a
299 particular state, e.g.:
300 PJSIP.registrations.state.Registered
304 * The ability to use "like" has been added to the pjsip list and show
305 CLI commands. For instance: CLI> pjsip list endpoints like abc
307 * If res_statsd is loaded and a StatsD server is configured, basic statistics
308 regarding the state of PJSIP contacts will now be emitted. This includes:
309 - A GAUGE statistic for the overall number of contacts in a particular
311 PJSIP.contacts.states.Reachable
312 - A TIMER statistic for the RTT time for each qualified contact, e.g.:
313 PJSIP.contacts.alice@@127.0.0.1:5061.rtt
315 res_sorcery_memory_cache
316 ------------------------
317 * A new caching strategy, full_backend_cache, has been added which caches
318 all stored objects in the backend. When enabled all objects will be
319 expired or go stale according to the configuration. As well when enabled
320 all retrieval operations will be performed against the cache instead of
325 * CALLERID(pres) is now documented as a valid alternative to setting both
326 CALLERID(name-pres) and CALLERID(num-pres) at once. Some channel drivers,
327 like chan_sip, don't make a distinction between the two: they take the
328 least public value from name-pres and num-pres. By using CALLERID(pres)
329 for reading and writing, you touch the same combined value in the dialplan.
330 The same applies to CONNECTEDLINE(pres), REDIRECTING(orig-pres),
331 REDIRECTING(to-pres) and REDIRECTING(from-pres).
335 * A new module that emits StatsD statistics regarding Asterisk endpoints.
336 This includes a total count of the number of endpoints, the count of the
337 number of endpoints in the technology agnostic state of the endpoint -
338 online or offline - as well as the number of channels associated with each
339 endpoint. These are recorded as three different GAUGE statistics:
341 - endpoints.state.{unknown|offline|online}
342 - endpoints.{tech}.{resource}.channels
345 ------------------------------------------------------------------------------
346 --- Functionality changes from Asterisk 13.5.0 to Asterisk 13.6.0 ------------
347 ------------------------------------------------------------------------------
351 * The CHANNEL function, when used on a PJSIP channel, now exposes a 'call-id'
352 extraction option when using with the 'pjsip' signalling option. It will
353 return the SIP Call-ID associated with the INVITE request that established
358 * Two new endpoint related events are now available: PeerStatusChange and
359 ContactStatusChange. In particular, these events are useful when subscribing
360 to all event sources, as they provide additional endpoint related
361 information beyond the addition/removal of channels from an endpoint.
363 * Added the ability to subscribe to all ARI events in Asterisk, regardless
364 of whether the application 'controls' the resource. This is useful for
365 scenarios where an ARI application merely wants to observe the system,
366 as opposed to control it. There are two ways to accomplish this:
367 (1) Via the WebSocket connection URI. A new query paramter, 'subscribeAll',
368 has been added that, when present and True, will subscribe all
369 specified applications to all ARI event sources in Asterisk.
370 (2) Via the applications resource. An ARI client can, at any time, subscribe
371 to all resources in an event source merely by not providing an explicit
372 resource. For example, subscribing to an event source of 'channels:'
373 as opposed to 'channels:12345' will subscribe the application to all
376 ------------------------------------------------------------------------------
377 --- Functionality changes from Asterisk 13.4.0 to Asterisk 13.5.0 ------------
378 ------------------------------------------------------------------------------
382 * A new ContactStatus event has been added that reflects res_pjsip contact
383 lifecycle changes: Created, Removed, Reachable, Unreachable, Unknown.
385 * Added the Linkedid header to the common channel headers listed for each
386 channel in AMI events.
390 * A new feature has been added that enables the retrieval of modules and
391 module information through an HTTP request. Information on a single module
392 can be also be retrieved. Individual modules can be loaded to Asterisk, as
393 well as unloaded and reloaded.
395 * A new resource has been added to the 'asterisk' resource, 'config/dynamic'.
396 This resource allows for push configuration of sorcery derived objects
397 within Asterisk. The resource supports creation, retrieval, updating, and
398 deletion. Sorcery derived objects that are manipulated by this resource
399 must have a sorcery wizard that supports the desired operations.
401 * A new feature has been added that allows for the rotation of log channels
402 through HTTP requests.
407 * A new 'g726_non_standard' endpoint option has been added that, when set to
408 'yes' and g.726 audio is negotiated, forces the codec to be treated as if it
409 is AAL2 packed on the channel.
411 * A new 'rtp_keepalive' endpoint option has been added. This option specifies
412 an interval, in seconds, at which we will send RTP comfort noise packets to
413 the endpoint. This functions identically to chan_sip's "rtpkeepalive" option.
415 * New 'rtp_timeout' and 'rtp_timeout_hold' endpoint options have been added.
416 These options specify the amount of time, in seconds, that Asterisk will wait
417 before terminating the call due to lack of received RTP. These are identical
418 to chan_sip's rtptimeout and rtpholdtimeout options.
420 ------------------------------------------------------------------------------
421 --- Functionality changes from Asterisk 13.3.0 to Asterisk 13.4.0 ------------
422 ------------------------------------------------------------------------------
426 * New 'rpid_immediate' option to control if connected line update information
427 goes to the caller immediately or waits for another reason to send the
428 connected line information update. See the online option documentation for
429 more information. Defaults to 'no' as setting it to 'yes' can result in
430 many unnecessary messages being sent to the caller.
432 * The configuration setting 'progressinband' now defaults to 'no', which
433 matches the actual behavior of previous versions.
437 * A new CLI command has been added: "pjsip show settings", which shows
438 both the global and system configuration settings.
440 * A new aor option has been added: "qualify_timeout", which sets the timeout
441 in seconds for a qualify. The default is 3 seconds. This overrides the
442 hard coded 32 seconds in pjproject.
444 * Endpoint status will now change to "Unreachable" when all contacts are
445 unavailable. When any contact becomes available, the endpoint will status
446 will change back to "Reachable".
448 * A new global option has been added: "max_initial_qualify_time", which
449 sets the maximum amount of time from startup that qualifies should be
450 attempted on all contacts.
454 * Two new events, 'ChannelHold' and 'ChannelUnhold', have been added to the
455 events data model. These events are raised when a channel indicates a hold
456 or unhold, respectively.
460 * A new dialplan function, HOLD_INTERCEPT, has been added. This function, when
461 placed on a channel, intercepts hold/unhold indications signalled by the
462 channel and prevents them from moving on to other channels in a bridge with
463 the hold initiator. Instead, AMI or ARI events are raised indicating that
464 the channel wanted to place someone on hold. This allows external
465 applications to implement their own custom hold/unhold logic.
467 ------------------------------------------------------------------------------
468 --- Functionality changes from Asterisk 13.2.0 to Asterisk 13.3.0 ------------
469 ------------------------------------------------------------------------------
471 chan_pjsip/app_transfer
473 * The Transfer application, when used with chan_pjsip, now supports using
474 a PJSIP endpoint as the transfer destination. This is in addition to
475 explicitly specifying a SIP URI to transfer to.
479 * The ARI /channels resource now supports a new operation, 'redirect'. The
480 redirect operation will perform a technology and state specific redirection
481 on the channel to a specified endpoint or destination. In the case of SIP
482 technologies, this is either a 302 Redirect response to an on-going INVITE
483 dialog or a SIP REFER request.
487 * A new 'endpoint_identifier_order' option has been added that allows one to
488 set the order by which endpoint identifiers are processed and checked. This
489 option is specified under the 'global' type configuration section.
491 ------------------------------------------------------------------------------
492 --- Functionality changes from Asterisk 13.1.0 to Asterisk 13.2.0 ------------
493 ------------------------------------------------------------------------------
495 * New 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions have been added which
496 allow examining PJSIP AORs or contacts from the dialplan.
498 res_pjsip_outbound_registration
500 * The 'pjsip send unregister' command now stops further registrations.
502 * A new command 'pjsip send register' has been added which allows you to
503 start or restart periodic registration. It can be used after a
504 'send unregister' or after a 401 permanent error.
506 res_pjsip_config_wizard
508 * This is a new module that adds streamlined configuration capability for
509 chan_pjsip. It's targeted at users who have lots of basic configuration
510 scenarios like 'phone' or 'agent' or 'trunk'. Additional information
511 can be found in the sample configuration file at
512 config/samples/pjsip_wizard.conf.sample.
516 * The T.38 negotiation timeout was previously hard coded at 5000 milliseconds
517 and is now configurable via the 't38timeout' configuration option in
518 res_fax.conf and via the fax options dialplan function 'FAXOPT(t38timeout)'.
519 The default remains at 5000 milliseconds.
523 * The ca_list_path transport parameter has been added for TLS transports. This
524 option behaves similarly to the old sip.conf option "tlscapath". In order to
525 use this, you must be using PJProject version 2.4 or higher.
529 * The Originate operation now takes in an originator channel. The linked ID of
530 this originator channel is applied to the newly originated outgoing channel.
531 If using CEL this allows an association to be established between the two so
532 it can be recognized that the originator is dialing the originated channel.
534 * "language" (the default spoken language for the channel) is now included in
535 the standard channel state output for suitable events.
537 * The POST channels/{id} operation and the POST channels/{id}/continue operation
538 now have a new "label" parameter. This allows for origination or continuation
539 to a labeled priority in the dialplan instead of requiring a specific priority
540 number. The ARI version has been bumped to 1.7.0 as a result.
544 * "Language" (the default spoken language for the channel) is now included in
545 the standard channel state output for suitable events.
547 * AMI actions that return a list of events have been made to return consistent
548 headers for the action response event starting the list and the list complete
549 event. The AMI version has been bumped to 2.7.0 as a result.
551 ------------------------------------------------------------------------------
552 --- Functionality changes from Asterisk 13.0.0 to Asterisk 13.1.0 ------------
553 ------------------------------------------------------------------------------
557 * Event NewConnectedLine is emitted when the connected line information on
562 * Event ChannelConnectedLine is emitted when the connected line information
563 on a channel changes.
568 The features.conf general section has three new configurable options:
569 * transferdialattempts
571 * transferinvalidsound
572 For more information on what these options do, see the Asterisk wiki:
573 https://wiki.asterisk.org/wiki/x/W4fAAQ
580 * New 'media_encryption_optimistic' endpoint setting. This will use SRTP
581 when possible but does not consider lack of it a failure.
583 res_pjsip_endpoint_identifer_ip
585 * New CLI commands have been added: "pjsip show identif(y|ies)", which lists
586 all configured PJSIP identify objects
588 ------------------------------------------------------------------------------
589 --- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
590 ------------------------------------------------------------------------------
595 Asterisk 13 is the next Long Term Support (LTS) release of Asterisk. As such,
596 the focus of development for this release of Asterisk was on improving the
597 usability and features developed in the previous Standard release, Asterisk 12.
598 Beyond a general refinement of end user features, development focussed heavily
599 on the Asterisk APIs - the Asterisk Manager Interface (AMI) and the Asterisk
600 REST Interface (ARI) - and the PJSIP stack in Asterisk. Some highlights of the
601 new features include:
603 * Asterisk security events are now provided via AMI, allowing end users to
604 monitor their Asterisk system in real time for security related issues.
605 * External control of Message Waiting Indicators (MWI) through both AMI and ARI.
606 * Reception/transmission of out of call text messages using any supported
607 channel driver/protocol stack through ARI.
608 * Resource List Server support in the PJSIP stack, providing subscriptions to
609 lists of resources and batched delivery of NOTIFY requests.
610 * Inter-Asterisk distributed device state and mailbox state using the PJSIP
613 It is important to note that Asterisk 13 is built on the architecture developed
614 during the previous Standard release, Asterisk 12. Users upgrading to
615 Asterisk 13 should read about the new features in Asterisk 12 later in this file
616 (see Functionality changes from Asterisk 11 to Asterisk 12), as well as the
617 UPGRADE-12.txt delivered with this release. In particular, users upgrading to
618 Asterisk 13 from a release prior to Asterisk 12 should read the specifications
619 on AMI, CDRs, and CEL on the Asterisk wiki:
620 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
621 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
622 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
624 Many new featuers in Asterisk 13 were introduced in point releases of
625 Asterisk 12. Following this section - which documents the changes from all
626 versions of Asterisk 12 to Asterisk 13 - users should examine the new features
627 that were introduced in the point releases of Asterisk 12, as they are also
628 included in Asterisk 13.
630 Finally, all users upgrading to Asterisk 13 should read the UPGRADE.txt file
631 delivered with this release.
636 * Sample config files have been moved from configs/ to a sub-folder of that
639 * The menuselect utility has been pulled into the Asterisk repository. As a
640 result, the libxml2 development library is now a required dependency for
643 * A new Compiler Flag, REF_DEBUG, has been added. When enabled, reference
644 counted objects will emit additional debug information to the refs log file
645 located in the standard Asterisk log file directory. This log file is useful
646 in tracking down object leaks and other reference counting issues. Prior to
647 this version, this option was only available by modifying the source code
648 directly. This change also includes a new script, refcounter.py, in the
649 contrib folder that will process the refs log file. Note that this replaces
650 the refcounter utility that could be built from the utils directory.
658 * This module was deprecated and has been removed. Users of app_dahdibarge
659 should use ChanSpy instead.
663 * New options to play a beep when starting a recording and stopping a recording
664 have been added. The option "p" will play a beep to the channel that starts
665 the recording. The option "P" will play a beep to the channel that stops the
670 * Queue rules can now be stored in a database table, queue_rules. Unlike other
671 RealTime tables, the queue_rules table is only examined on module load or
672 module reload. A new general setting has been added to queuerules.conf,
673 'realtime_rules', which, when set to 'yes', will cause app_queue to look in
674 RealTime for additional queue rules to parse. Note that both the file and
675 the database can be used as a provide of queue rules when 'realtime_rules'
678 When app_queue is reloaded, all rules are re-parsed and loaded into memory.
679 There is no caching of RealTime queue rules.
683 * This module was deprecated and has been removed. Users of app_readfile
684 should use func_env's FILE function instead.
688 * The 'say' family of dialplan applications now support the Japanese
689 language. The 'language' parameter in say.conf now recognizes a setting of
690 'ja', which will enable Japanese language specific mechanisms for playing
691 back numbers, dates, and other items.
695 * This module was deprecated and has been removed. Users of app_saycountpl
696 should use the Say family of applications.
700 * The SetMusicOnHold dialplan application was deprecated and has been removed.
701 Users of the application should use the CHANNEL function's musicclass
706 * The WaitMusicOnHold dialplan application was deprecated and has been
707 removed. Users of the application should use MusicOnHold with a duration
712 * VoiceMail and VoiceMailMain now support the Japanese language. The
713 'language' parameter in voicemail.conf now recognizes a setting of 'ja',
714 which will enable prompts to be played back using a Japanese grammatical
715 structure. Additional prompts are necessary for this functionality,
717 - jb-arimasu: there is
718 - jb-arimasen: there is not
719 - jb-oshitekudasai: please press
725 * Add the ability to specify multiple email addresses in configuration,
734 * This module was deprecated and has been removed. Users of cdr_sqlite
735 should use cdr_sqlite3_custom.
739 * Added the ability to support PostgreSQL application_name on connections.
740 This allows PostgreSQL to display the configured name in the
741 pg_stat_activity view and CSV log entries. This setting is configurable
742 for cdr_pgsql via the appname configuration setting in cdr_pgsql.conf.
750 * Added the ability to support PostgreSQL application_name on connections.
751 This allows PostgreSQL to display the configured name in the
752 pg_stat_activity view and CSV log entries. This setting is configurable
753 for cel_pgsql via the appname configuration setting in cel_pgsql.conf.
761 * SS7 support now requires libss7 v2.0 or later.
763 * Added SS7 support for connected line and redirecting.
765 * Most SS7 CLI commands are reworked as well as new SS7 commands added.
768 * Added several SS7 config option parameters described in
769 chan_dahdi.conf.sample.
773 * This module was deprecated and has been removed. Users of chan_gtalk
774 should use chan_motif.
778 * This module was deprecated and has been removed. Users of chan_h323
779 should use chan_ooh323.
783 * This module was deprecated and has been removed. Users of chan_jingle
784 should use chan_motif.
788 * Added the CLI command 'pjsip list ciphers' so a user can know what
789 OpenSSL names are available on their system for the pjsip.conf cipher
794 * The SIPPEER dialplan function no longer supports using a colon as a
795 delimiter for parameters. The parameters for the function should be
796 delimited using a comma.
798 * The SIPCHANINFO dialplan function was deprecated and has been removed. Users
799 of the function should use the CHANNEL function instead.
807 * Added functional peeraccount support. Except for Queue, the
808 accountcode propagation is now consistently propagated to outgoing
809 channels before dialing. The channel accountcode can change from its
810 original non-empty value on channel creation for the following specific
811 reasons. One, dialplan sets it using CHANNEL(accountcode). Two, an
812 originate method that can specify an accountcode value. Three, the
813 calling channel propagates its peeraccount or accountcode to the
814 outgoing channel's accountcode before dialing. The change has two
815 visible effects. One, local channels now cross accountcode and
816 peeraccount across the special bridge between the ;1 and ;2 channels
817 just like channels between normal bridges. Two, the
818 CHANNEL(peeraccount) value can now be set before Dial and FollowMe to
819 set the accountcode on the outgoing channel(s).
821 For Queue, an outgoing channel's non-empty accountcode will not change
822 unless explicitly set by CHANNEL(accountcode). The change has three
823 visible effects. One, local channels now cross accountcode and
824 peeraccount across the special bridge between the ;1 and ;2 channels
825 just like channels between normal bridges. Two, the queue member will
826 get an accountcode if it doesn't have one and one is available from the
827 calling channel's peeraccount. Three, accountcode propagation includes
828 local channel members where the accountcodes are propagated early
829 enough to be available on the ;2 channel.
833 * New DeviceStateChanged and PresenceStateChanged AMI events have been added.
834 These events are emitted whenever a device state or presence state change
835 occurs. The events are controlled by res_manager_device_state.so and
836 res_manager_presence_state.so. If the high frequency of these events is
837 problematic for you, do not load these modules.
839 * Added DialplanExtensionAdd and DialplanExtensionRemove AMI commands. They
840 work in basically the same way as the 'dialplan add extension' and
841 'dialplan remove extension' CLI commands respectively.
843 * New AMI action LoggerRotate reloads and rotates logger in the same manner
844 as CLI command 'logger rotate'
846 * New AMI Actions FAXSessions, FAXSession, and FAXStats replicate the
847 functionality of CLI commands 'fax show sessions', 'fax show session',
848 and fax show stats' respectively.
850 * New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset
851 enable manager control over PRI debugging levels and file output.
853 * AMI action PJSIPNotify may now send to a URI instead of only to a PJSIP
854 endpoint as long as a default outbound endpoint is set. This also applies
855 to the equivalent CLI command (pjsip send notify)
857 * The AMI action PJSIPShowEndpoint now includes ContactStatusDetail sections
858 that give information on Asterisk's attempts to qualify the endpoint.
860 * The DialEnd event will now contain a Forward header if the dial is ending
861 due to the call being forwarded. The contents of the Forward header is the
862 extension in the number to which the call is being forwarded.
866 * The "bridge_technology" extra field key has been added to BRIDGE_ENTER
867 and BRIDGE_EXIT events.
871 * Channel variables are now substituted in arguments passed to applications
872 run by using dynamic features.
876 * The TLS core in Asterisk now supports Perfect Forward Secrecy (PFS).
877 Enabling PFS is attempted by default, and is dependent on the configuration
878 of the module using TLS.
879 - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
880 specify a ECDHE cipher suite in sip.conf, for example:
881 tlscipher=AES128-SHA:DES-CBC3-SHA
882 - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
883 into the private key file, e.g., sip.conf tlsprivatekey. For example, the
884 default dh2048.pem - see
885 http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt
886 - Because clients expect the server to prefer PFS, and because OpenSSL sorts
887 its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
888 Consider re-ordering your cipher suites in the respective configuration
890 tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
891 will use PFS when offered by the client. Clients which do not offer PFS
892 fall-back to AES-128 (or even 3DES, as recommended by RFC 3261).
900 * The JACK_HOOK function now supports audio with a sample rate higher than
909 * Added the ability to support PostgreSQL application_name on connections.
910 This allows PostgreSQL to display the configured name in the
911 pg_stat_activity view and CSV log entries. This setting is configurable
912 for res_config_pgsql via the dbappname configuration setting in
915 res_pjsip_outbound_publish
917 * A new module, res_pjsip_outbound_publish provides the mechanisms for sending
918 PUBLISH requests for specific event packages to another SIP User Agent.
922 * The publish/subscribe core module has been updated to support RFC 4662
923 Resource Lists, allowing Asterisk to act as a Resource List Server (RLS).
924 Resource lists are configured in pjsip.conf under a new object type,
925 resource_list. Resource lists can contain either message-summary or presence
926 events, and can be composed of specific resources that provide the event or
927 other resource lists.
929 * Inbound publication support is provided by a new object, inbound-publication.
930 This configures res_pjsip_pubsub to accept PUBLISH requests from a particular
931 resource. Which events are accepted is constructed dynamically; see
932 res_pjsip_publish_asterisk for more information.
934 res_pjsip_publish_asterisk
936 * A new module, res_pjsip_publish_asterisk adds support for PUBLISH requests of
937 Asterisk information to other Asterisk servers. This module is intended only
938 for Asterisk to Asterisk exchanges of information. Currently, this includes
939 both mailbox state and device state information.
941 ------------------------------------------------------------------------------
942 --- Functionality changes from Asterisk 12.4.0 to Asterisk 12.5.0 ------------
943 ------------------------------------------------------------------------------
947 * Stored recordings now support a new operation, copy. This will take an
948 existing stored recording and copy it to a new location in the recordings
951 * LiveRecording objects now have three additional fields that can be reported
952 in a RecordingFinished ARI event:
953 - total_duration: the duration of the recording
954 - talking_duration: optional. The duration of talking detected in the
955 recording. This is only available if max_silence_seconds was specified
956 when the recording was started.
957 - silence_duration: optional. The duration of silence detected in the
958 recording. This is only available if max_silence_seconds was specified
959 when the recording was started.
960 Note that all duration values are reported in seconds.
962 * Users of ARI can now send and receive out of call text messages. Messages
963 can be sent directly to a particular endpoint, or can be sent to the
964 endpoints resource directly and inferred from the URI scheme. Text
965 messages are passed to ARI clients as TextMessageReceived events. ARI
966 clients can choose to receive text messages by subscribing to the particular
967 endpoint technology or endpoints that they are interested in.
969 * The applications resource now supports subscriptions to all endpoints of
970 a particular channel technology. For example, subscribing to an eventSource
971 of 'endpoint:PJSIP' will subscribe to all PJSIP endpoints.
975 * The endpoint configuration object now supports 'accountcode'. Any channel
976 created for an endpoint with this setting will have its accountcode set
977 to the specified value.
981 * A new module, res_hep_rtcp, has been added that will forward RTCP call
982 statistics to a HEP capture server. See res_hep for more information.
986 * Function AUDIOHOOK_INHERIT has been deprecated. Audiohooks are now
987 unconditionally inhereted through masquerades. As a side benefit, more
988 than one audiohook of a given type may persist through a masquerade now.
990 ------------------------------------------------------------------------------
991 --- Functionality changes from Asterisk 12.3.0 to Asterisk 12.4.0 ------------
992 ------------------------------------------------------------------------------
996 * Returns new AGENT_STATUS value "NOT_CONNECTED" if the agent fails to
997 connect with an incoming caller after being alerted to the presence
998 of the incoming caller. The most likely reason this would happen is
999 the agent did not acknowledge the call in time.
1003 * New events have been added for the TALK_DETECT function. When the function
1004 is used on a channel, ChannelTalkingStart/ChannelTalkingStop events will be
1005 emitted to connected AMI clients indicating the start/stop of talking on
1010 * New event models have been aded for the TALK_DETECT function. When the
1011 function is used on a channel, ChannelTalkingStarted/ChannelTalkingFinished
1012 events will be emitted to connected WebSockets subscribed to the channel,
1013 indicating the start/stop of talking on the channel.
1017 * A new function, TALK_DETECT, has been added. When set on a channel, this
1018 fucntion causes events indicating the starting/stoping of talking on said
1019 channel to be emitted to both AMI and ARI clients.
1021 ------------------------------------------------------------------------------
1022 --- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------
1023 ------------------------------------------------------------------------------
1027 * A new Playback URI 'tone' has been added. Tones are specified either as
1028 an indication name (e.g. 'tone:busy') from indications.conf or as a tone
1029 pattern (e.g. 'tone:240/250,0/250'). Tones differ from normal playback
1030 URIs in that they must be stopped manually and will continue to occupy
1031 a channel's ARI control queue until they are stopped. They also can not
1032 be rewound or fastforwarded.
1034 * User events can now be generated from ARI. Events can be signalled with
1035 arbitrary json variables, and include one or more of channel, bridge, or
1036 endpoint snapshots. An application must be specified which will receive
1037 the event message (other applications can subscribe to it). The message
1038 will also be delivered via AMI provided a channel is attached. Dialplan
1039 generated user event messages are still transmitted via the channel, and
1040 will only be received by a stasis application they are attached to or if
1041 the channel is subscribed to.
1045 * SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI
1046 fields for prohibited callingpres information. Values are legacy, no, and
1047 yes. By default, legacy is used.
1048 trust_id_outbound=legacy - behavior remains the same as 1.8.26.1. When
1049 dealing with prohibited callingpres and sendrpid=pai/rpid, RPID/PAI
1050 headers are appended to outbound SIP messages just as they are with
1051 allowed callingpres values, but data about the remote party's identity is
1053 When sendrpid=rpid, only the remote party's domain is anonymized.
1054 trust_id_outbound=no - when dealing with prohibited callingpres, RPID/PAI
1055 headers are not sent.
1056 trust_id_outbound=yes - RPID/PAI headers are applied with the full remote
1057 party information in tact even for prohibited callingpres information.
1058 In the case of PAI, a Privacy: id header will be appended for prohibited
1059 calling information to communicate that the private information should
1060 not be relayed to untrusted parties.
1064 * Manager action 'Park' now takes an additional argument 'AnnounceChannel'
1065 which can be used to announce the parked call's location to an arbitrary
1066 channel in a bridge. If 'Channel' and 'TimeoutChannel' are now the two
1067 parties in a one to one bridge, 'TimeoutChannel' is treated as having
1068 parked 'Channel' like with the Park Call DTMF feature and will receive
1069 announcements prior to being hung up.
1071 ------------------------------------------------------------------------------
1072 --- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
1073 ------------------------------------------------------------------------------
1077 * Record application now has an option 'o' which allows 0 to act as an exit
1078 key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'
1081 --------------------------
1082 * ChanSpy now accepts a channel uniqueid or a fully specified channel name
1083 as the chanprefix parameter if the 'u' option is specified.
1086 --------------------------
1087 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
1088 conference user menus.
1090 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
1091 menus, bridge settings, and user settings that have been applied by the
1092 CONFBRIDGE dialplan function.
1094 * The ConfBridge dialplan application now sets a channel variable,
1095 CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
1096 how a channel exited the conference.
1098 * Added conference user option 'announce_join_leave_review'. This option
1099 implies 'announce_join_leave' with the added effect that the user will
1100 be asked if they want to confirm or re-record the recording of their
1101 name when entering the conference
1104 --------------------------
1105 * At exit, the Directory application now sets a channel variable
1106 DIRECTORY_RESULT to one of the following based on the reason for exiting:
1107 OPERATOR user requested operator by pressing '0' for operator
1108 ASSISTANT user requested assistant by pressing '*' for assistant
1109 TIMEOUT user pressed nothing and Directory stopped waiting
1110 HANGUP user's channel hung up
1111 SELECTED user selected a user from the directory and is routed
1112 USEREXIT user pressed '#' from the selection prompt to exit
1113 FAILED directory failed in a way that wasn't accounted for. Dang.
1117 * Monitor() - A new option, B(), has been added that will turn on a periodic
1118 beep while the call is being recorded.
1121 --------------------------
1122 * MusicOnHold streams (all modes other than "files") now support wide band
1126 --------------------------
1127 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
1128 and for the channel executing Page respectively.
1131 --------------------------
1132 * PickupChan now accepts channel uniqueids of channels to pickup.
1135 --------------------------
1136 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
1137 to 'true' (case insensitive), then any Say application (SayNumber,
1138 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
1139 anticipate DTMF. If DTMF is received, these applications will behave like
1140 the background application and jump to the received extension once a match
1141 is established or after a short period of inactivity.
1144 -------------------------
1145 * A new function, MIXMONITOR, has been added to allow access to individual
1146 instances of MixMonitor on a channel.
1148 * A new option, B(), has been added that will turn on a periodic beep while the
1149 call is being recorded.
1153 -------------------------
1156 -------------------------
1157 * TEL URI support for inbound INVITE requests has been added. chan_sip will
1158 now handle TEL schemes in the Request and From URIs. The phone-context in
1159 the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on
1160 the inbound channel.
1164 * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
1165 the new AST_SORCERY diaplan function.
1167 * Core Show Locks output now includes Thread/LWP ID if the platform
1168 supports this feature.
1170 * New "logger add channel" and "logger remove channel" CLI commands have
1171 been added to allow creation and deletion of dynamic logger channels
1172 without configuration changes. These dynamic logger channels will only
1173 exist until the next restart of asterisk.
1177 * The live recording object on recording events now contains a target_uri
1178 field which contains the URI of what is being recorded.
1180 * The bridge type used when creating a bridge is now a comma separated list of
1181 bridge properties. Valid options are: mixing, holding, dtmf_events, and
1184 * A channelId can now be provided when creating a channel, either in the
1185 uri (POST channels/my-channel-id) or as query parameter. A local channel
1186 will suffix the second channel id with ';2' unless provided as query
1187 parameter otherChannelId.
1189 * A bridgeId can now be provided when creating a bridge, either in the uri
1190 (POST bridges/my-bridge-id) or as a query parameter.
1192 * A playbackId can be provided when starting a playback, either in the uri
1193 (POST channels/my-channel-id/play/my-playback-id /
1194 POST bridges/my-bridge-id/play/my-playback-id) or as a query parameter.
1196 * A snoop channel can be started with a snoopId, in the uri or query.
1200 * Originate now takes optional parameters ChannelId and OtherChannelId,
1201 used to set the UniqueId on creation. The other id is assigned to the
1202 second channel when dialing LOCAL, or defaults to appending ;2 if only
1203 the single Id is given.
1205 * The Mixmonitor action now has a "Command" header that can be used to
1206 indicate a post-process command to run once recording finishes.
1210 * A new set of Alembic scripts has been added for CDR tables. This will create
1211 a 'cdr' table with the default schema that Asterisk expects.
1216 * A new function was added: PERIODIC_HOOK. This allows running a periodic
1217 dialplan hook on a channel. Any audio generated by this hook will be
1218 injected into the call.
1226 * A new module, res_hep, has been added, that acts as a generic packet
1227 capture agent for the Homer Encapsulation Protocol (HEP) version 3.
1228 It can be configured via hep.conf. Other modules can use res_hep to send
1229 message traffic to a HEP capture server.
1233 * A new module, res_hep_pjsip, has been added that will forward PJSIP
1234 message traffic to a HEP capture server. See res_hep for more
1239 * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
1240 be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
1242 * Added the following new CLI commands:
1243 - "pjsip show contacts" - list all current PJSIP contacts.
1244 - "pjsip show contact" - show specific information about a current PJSIP
1246 - "pjsip show channel" - show detailed information about a PJSIP channel.
1248 res_pjsip_multihomed
1250 * A new module, res_pjsip_multihomed handles situations where the system
1251 Asterisk is running out has multiple interfaces. res_pjsip_multihomed
1252 determines which interface should be used during message sending.
1254 res_pjsip_pidf_digium_body_supplement
1256 * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
1257 request body formatting for presence support in Digium phones.
1259 res_pjsip_send_to_voicemail
1261 * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
1262 particular headers to transfer a PJSIP channel directly to a particular
1263 extension that has VoiceMail. This is intended to be used with Digium
1264 phones that support this feature.
1266 res_pjsip_outbound_registration
1268 * A new CLI command has been added: "pjsip show registrations", which lists
1269 all configured PJSIP registrations
1272 ------------------------------------------------------------------------------
1273 --- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
1274 ------------------------------------------------------------------------------
1278 * Added a new module that provides AMI control over MWI within Asterisk,
1279 res_mwi_external_ami. Note that this module depends on res_mwi_external;
1280 for more information on enabling this module, see res_mwi_external.
1281 This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
1282 the MWIGet/MWIGetComplete events.
1284 * The DialStatus field in the DialEnd event can now contain additional
1285 statuses that convey how the dial operation terminated. This includes
1286 ABORT, CONTINUE, and GOTO.
1288 * AMI will now emit security events. A new class authorization has been
1289 added in manager.conf for the security events, 'security'. The new events
1291 - FailedACL - raised when a request violates an ACL check
1292 - InvalidAccountID - raised when a request fails an authentication
1293 check due to an invalid account ID
1294 - SessionLimit - raised when a request fails due to exceeding the
1295 number of allowed concurrent sessions for a service
1296 - MemoryLimit - raised when a request fails due to an internal memory
1298 - LoadAverageLimit - raised when a request fails because a configured
1299 load average limit has been reached
1300 - RequestNotAllowed - raised when a request is not allowed by
1302 - AuthMethodNotAllowed - raised when a request used an authentication
1303 method not allowed by the service
1304 - RequestBadFormat - raised when a request is received with bad formatting
1305 - SuccessfulAuth - raised when a request successfully authenticates
1306 - UnexpectedAddress - raised when a request has a different source address
1307 then what is expected for a session already in progress with a service
1308 - ChallengeResponseFailed - raised when a request's attempt to authenticate
1309 has been challenged, and the request failed the authentication challenge
1310 - InvalidPassword - raised when a request provides an invalid password
1311 during an authentication attempt
1312 - ChallengeSent - raised when an Asterisk service send an authentication
1313 challenge to a request
1314 - InvalidTransport - raised when a request attempts to use a transport not
1315 allowed by the Asterisk service
1317 * Bridge related events now have two additional fields: BridgeName and
1318 BridgeCreator. BridgeName is a descriptive name for the bridge;
1319 BridgeCreator is the name of the entity that created the bridge. This
1320 affects the following events: ConfbridgeStart, ConfbridgeEnd,
1321 ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
1322 ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
1323 AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
1327 * The Bridge data model now contains the additional fields 'name' and
1328 'creator'. The 'name' field conveys a descriptive name for the bridge;
1329 the 'creator' field conveys the name of the entity that created the bridge.
1330 This affects all responses to HTTP requests that return a Bridge data model
1331 as well as all event derived data models that contain a Bridge data model.
1332 The POST /bridges operation may now optionally specify a name to give to
1333 the bridge being created.
1335 * Added a new ARI resource 'mailboxes' which allows the creation and
1336 modification of mailboxes managed by external MWI. Modules res_mwi_external
1337 and res_stasis_mailbox must be enabled to use this resource. For more
1338 information on external MWI control, see res_mwi_external.
1340 * Added new events for externally initiated transfers. The event
1341 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
1342 of a bridge in the ARI controlled application to the dialplan; the
1343 BridgeAttendedTransfer event is raised when a channel initiates an
1344 attended transfer of a bridge in the ARI controlled application to the
1347 * Channel variables may now be specified as a body parameter to the
1348 POST /channels operation. The 'variables' key in the JSON is interpreted
1349 as a sequence of key/value pairs that will be added to the created channel
1350 as channel variables. Other parameters in the JSON body are treated as
1351 query parameters of the same name.
1355 * Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be
1356 automatically handled by the HTTP server if a request is received with a
1357 Transfer-Encoding type of "chunked".
1361 * Path support has been added with the 'support_path' option in registration
1364 * A 'debug' option has been added to the globals section that will allow
1365 sip messages to be logged.
1367 * A 'set_var' option has been added to endpoints that will automatically
1368 set the desired variable(s) on a channel created for that endpoint.
1370 * Several new tables and columns have been added to the realtime schema for
1371 the res_pjsip related modules. See the UPGRADE.txt notes for updating
1372 the database schema.
1376 * A new module, res_mwi_external, has been added to Asterisk. This module
1377 acts as a base framework that other modules can build on top of to allow
1378 an external system to control MWI within Asterisk. For implementations
1379 that make use of res_mwi_external, see res_mwi_external_ami and
1380 res_ari_mailboxes. Note that res_mwi_external canflicts with other modules
1381 that may produce MWI themselves, such as app_voicemail. res_mwi_external
1382 and other modules that depend on it cannot be built or loaded with
1383 app_voicemail present.
1387 * DNS functionality will now automatically be enabled if the system configured
1388 nameservers can be retrieved. If the system configured nameservers can not be
1389 retrieved the functionality will resort to using system resolution. Functionalty
1390 such as SRV records and failover will not be available if system resolution
1393 ------------------------------------------------------------------------------
1394 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
1395 ------------------------------------------------------------------------------
1400 Asterisk 12 is a standard release of the Asterisk project. As such, the
1401 focus of development for this release was on core architectural changes and
1402 major new features. This includes:
1403 * A more flexible bridging core based on the Bridging API
1404 * A new internal message bus, Stasis
1405 * Major standardization and consistency improvements to AMI
1406 * Addition of the Asterisk RESTful Interface (ARI)
1407 * A new SIP channel driver, chan_pjsip
1408 In addition, as the vast majority of bridging in Asterisk was migrated to the
1409 Bridging API used by ConfBridge, major changes were made to most of the
1410 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
1412 Specifications have been written for the affected interfaces. These
1413 specifications are available on the Asterisk wiki:
1414 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
1415 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
1416 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
1418 It is *highly* recommended that anyone migrating to Asterisk 12 read the
1419 information regarding its release both in this file and in the accompanying
1420 UPGRADE.txt file. More detailed information on the major changes can be found
1421 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
1426 * Added build option DISABLE_INLINE. This option can be used to work around a
1427 bug in gcc. For more information, see
1428 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
1430 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
1431 the CHANNEL_TRACE build option were incompatible with the new bridging
1434 * Asterisk now optionally uses libxslt to improve XML documentation generation
1435 and maintainability. If libxslt is not available on the system, some XML
1436 documentation will be incomplete.
1438 * Asterisk now depends on libjansson. If a package of libjansson is not
1439 available on your distro, please see http://www.digip.org/jansson/.
1441 * Asterisk now depends on libuuid and, optionally, uriparser. It is
1442 recommended that you install uriparser, even if it is optional.
1444 * The new SIP stack and channel driver uses a particular version of PJSIP.
1445 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
1446 configuring and installing PJSIP for usage with Asterisk.
1448 * Optional API was re-implemented to be more portable, and no longer requires
1449 weak reference support from the compiler. The build option OPTIONAL_API may
1450 be disabled to disable Optional API support.
1457 * Along with AgentRequest, this application has been modified to be a
1458 replacement for chan_agent. The act of a channel calling the AgentLogin
1459 application places the channel into a pool of agents that can be
1460 requested by the AgentRequest application. Note that this application, as
1461 well as all other agent related functionality, is now provided by the
1462 app_agent_pool module. See chan_agent and AgentRequest for more information.
1464 * This application no longer performs agent authentication. If authentication
1465 is desired, the dialplan needs to perform this function using the
1466 Authenticate or VMAuthenticate application or through an AGI script before
1469 * If this application is called and the agent is already logged in, the
1470 dialplan will continue exection with the AGENT_STATUS channel variable set
1471 to ALREADY_LOGGED_IN.
1473 * The agents.conf schema has changed. Rather than specifying agents on a
1474 single line in comma delineated fashion, each agent is defined in a separate
1475 context. This allows agents to use the power of context templates in their
1478 * A number of parameters from agents.conf have been removed. This includes
1479 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
1480 urlprefix, and savecallsin. These options were obsoleted by the move from
1481 a channel driver model to the bridging/application model provided by
1486 * A new application, this will request a logged in agent from the pool and
1487 bridge the requested channel with the channel calling this application.
1488 Logged in agents are those channels that called the AgentLogin application.
1489 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
1490 application will be set with an appropriate error value.
1492 AgentMonitorOutgoing
1494 * This application has been removed. It was a holdover from when
1495 AgentCallbackLogin was removed.
1499 * Added support for additional Ademco DTMF signalling formats, including
1500 Express 4+1, Express 4+2, High Speed and Super Fast.
1502 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
1503 call time, in milliseconds, to run the application.
1505 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
1506 maximum number of times to retry the call.
1508 * Added a new configuration option answait. If set, the AlarmReceiver
1509 application will wait the number of milliseconds specified by answait
1510 after the channel has answered. Valid values range between 500
1511 milliseconds and 10000 milliseconds.
1513 * Added configuration option no_group_meta. If enabled, grouping of metadata
1514 information in the AlarmReceiver log file will be skipped.
1518 * It is now no longer possible to bypass updating the CDR on the channel
1519 when answering. CDRs reflect the state of the channel and will always
1520 reflect the time they were Answered.
1524 * A new application in Asterisk, this will place the calling channel
1525 into a holding bridge, optionally entertaining them with some form of
1526 media. Channels participating in a holding bridge do not interact with
1527 other channels in the same holding bridge. Optionally, however, a channel
1528 may join as an announcer. Any media passed from an announcer channel is
1529 played to all channels in the holding bridge. Channels leave a holding
1530 bridge either when an optional timer expires, or via the ChannelRedirect
1531 application or AMI Redirect action.
1535 * All participants in a bridge can now be kicked out of a conference room
1536 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
1537 command, i.e., 'confbridge kick <conference> all'
1539 * CLI output for the 'confbridge list' command has been improved. When
1540 displaying information about a particular bridge, flags will now be shown
1541 for the participating users indicating properties of that user.
1543 * The ConfbridgeList event now contains the following fields: WaitMarked,
1544 EndMarked, and Waiting. This displays additional properties about the
1545 user's profile, as well as whether or not the user is waiting for a
1546 Marked user to enter the conference.
1548 * Added a new option for conference recording, record_file_append. If enabled,
1549 when the recording is stopped and then re-started, the existing recording
1550 will be used and appended to.
1552 * ConfBridge now has the ability to set the language of announcements to the
1553 conference. The language can be set on a bridge profile in confbridge.conf
1554 or by the dialplan function CONFBRIDGE(bridge,language)=en.
1558 * The channel variable CPLAYBACKSTATUS may now return the value
1559 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
1560 such as AMI. See the AMI action ControlPlayback for more information.
1564 * Added the 'a' option, which allows the caller to enter in an additional
1565 alias for the user in the directory. This option must be used in conjunction
1566 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
1567 specified in voicemail.conf.
1571 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
1572 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
1573 containing the unique ID of the bridge that the channel happens to be in.
1577 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
1578 for more information.
1580 * Variables are no longer purged from the original CDR. See the 'v' option for
1583 * The 'A' option has been removed. The Answer time on a CDR is never updated
1586 * The 'd' option has been removed. The disposition on a CDR is a function of
1587 the state of the channel and cannot be altered.
1589 * The 'D' option has been removed. Who the Party B is on a CDR is a function
1590 of the state of the respective channels involved in the CDR and cannot be
1593 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
1594 such that the start time and, if applicable, the answer time was updated.
1595 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
1596 'r' option now triggers the Reset, setting the start time (and answer time
1597 if applicable) to the current time. Note that the 'a' option still sets
1598 the answer time to the current time if the channel was already answered.
1600 * The 's' option has been removed. A variable can be set on the original CDR
1601 if desired using the CDR function, and removed from a forked CDR using the
1604 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
1605 longer applies in the CDR engine.
1607 * The 'v' option now prevents the copy of the variables from the original CDR
1608 to the forked CDR. Previously the variables were always copied but were
1609 removed from the original. This was changed as removing variables from a CDR
1610 can have unintended side effects - this option allows the user to prevent
1611 propagation of variables from the original to the forked without modifying
1616 * Added the 'n' option to MeetMe to prevent application of the DENOISE
1617 function to a channel joining a conference. Some channel drivers that vary
1618 the number of audio samples in a voice frame will experience significant
1619 quality problems if a denoiser is attached to the channel; this option gives
1620 them the ability to remove the denoiser without having to unload func_speex.
1624 * The 'b' option now includes conferences as well as sounds played to the
1627 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
1628 running during a transfer. If a MixMonitor is started on a channel,
1629 the MixMonitor will continue to record the audio passing through the
1630 channel even in the presence of transfers.
1634 * The NoCDR application is deprecated. Please use the CDR_PROP function to
1637 * While the NoCDR application will prevent CDRs for a channel from being
1638 propagated to registered CDR backends, it will not prevent that data from
1639 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
1640 function that enables CDRs on a channel will restore those records that have
1641 not yet been finalized.
1645 * The app_parkandannounce module has been removed. The application
1646 ParkAndAnnounce is now provided by the res_parking module. See the
1647 res_parking changes for more information.
1651 * Added queue available hint. The hint can be added to the dialplan using the
1652 following syntax: exten,hint,Queue:{queue_name}_avail
1653 For example, if the name of the queue is 'markq':
1654 exten => 8501,hint,Queue:markq_avail
1655 This will report 'InUse' if there are no logged in agents or no free agents.
1656 It will report 'Idle' when an agent is free.
1658 * Queues now support a hint for member paused state. The hint uses the form
1659 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
1660 are the name of the queue and the name of the member to subscribe to,
1661 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
1662 Members will show as In Use when paused.
1664 * The configuration options eventwhencalled and eventmemberstatus have been
1665 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
1666 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
1667 sent. The "Variable" fields will also no longer exist on the Agent* events.
1668 These events can be filtered out from a connected AMI client using the
1669 eventfilter setting in manager.conf.
1671 * The queue log now differentiates between blind and attended transfers. A
1672 blind transfer will result in a BLINDTRANSFER message with the destination
1673 context and extension. An attended transfer will result in an
1674 ATTENDEDTRANSFER message. This message will indicate the method by which
1675 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
1676 for running an application on a bridge or channel, or "LINK" for linking
1677 two bridges together with local channels. The queue log will also now detect
1678 externally initiated blind and attended transfers and record the transfer
1681 * When performing queue pause/unpause on an interface without specifying an
1682 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
1683 least one member of any queue exists for that interface.
1685 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
1686 for realtime queue log entries.
1690 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
1691 CDRs when they were previously disabled on a channel.
1693 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
1694 backends occurs on an as-needed basis in order to preserve linkedid
1695 propagation and other needed behavior.
1699 * A new application, this is similar to SayAlpha except that it supports
1700 case sensitive playback of the specified characters. For example,
1701 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
1705 * This application is deprecated in favor of CHANNEL(amaflags).
1709 * The SendDTMF application will now accept 'W' as valid input. This will cause
1710 the application to delay one second while streaming DTMF.
1714 * A new application in Asterisk 12, this hands control of the channel calling
1715 the application over to an external system. Currently, external systems
1716 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
1720 * UserEvent will now handle duplicate keys by overwriting the previous value
1721 assigned to the key.
1723 * In addition to AMI, UserEvent invocations will now be distributed to any
1724 interested Stasis applications.
1728 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1729 system as mailbox@context. The rest of the system cannot add @default
1730 to mailbox identifiers for app_voicemail that do not specify a context
1731 any longer. It is a mailbox identifier format that should only be
1732 interpreted by app_voicemail.
1734 * The voicemail.conf configuration file now has an 'alias' configuration
1735 parameter for use with the Directory application. The voicemail realtime
1736 database table schema has also been updated with an 'alias' column.
1741 * Pass through support has been added for both VP8 and Opus.
1743 * Added format attribute negotiation for the Opus codec. Format attribute
1744 negotiation is provided by the res_format_attr_opus module.
1749 * Masquerades as an operation inside Asterisk have been effectively hidden
1750 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
1751 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
1752 dropping of frame/audio hooks, and other internal implementation details
1753 that users had to deal with. This fundamental change has large implications
1754 throughout the changes documented for this version. For more information
1755 about the new core architecture of Asterisk, please see the Asterisk wiki.
1757 * Multiple parties in a bridge may now be transferred. If a participant in a
1758 multi-party bridge initiates a blind transfer, a Local channel will be used
1759 to execute the dialplan location that the transferer sent the parties to. If
1760 a participant in a multi-party bridge initiates an attended transfer,
1761 several options are possible. If the attended transfer results in a transfer
1762 to an application, a Local channel is used. If the attended transfer results
1763 in a transfer to another channel, the resulting channels will be merged into
1766 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
1767 driver specific. If the channel variable is set on the transferrer channel,
1768 the sound will be played to the target of an attended transfer.
1770 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
1771 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
1772 listed. Any more peers in the bridge will not be included in the list.
1773 BRIDGEPEER is not valid in holding bridges like parking since those channels
1774 do not talk to each other even though they are in a bridge.
1776 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
1777 and will contain a value if the BRIDGEPEER's channel driver supports it.
1779 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
1780 was responsible for an attended transfer in a similar fashion to
1783 * Modules using the Configuration Framework or Sorcery must have XML
1784 configuration documentation. This configuration documentation is included
1785 with the rest of Asterisk's XML documentation, and is accessible via CLI
1786 commands. See the CLI changes for more information.
1788 AMI (Asterisk Manager Interface)
1790 * Major changes were made to both the syntax as well as the semantics of the
1791 AMI protocol. In particular, AMI events have been substantially improved
1792 in this version of Asterisk. For more information, please see the AMI
1793 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
1795 * AMI events that reference a particular channel or bridge will now always
1796 contain a standard set of fields. When multiple channels or bridges are
1797 referenced in an event, fields for at least some subset of the channels
1798 and bridges in the event will be prefixed with a descriptive name to avoid
1799 name collisions. See the AMI event documentation on the Asterisk wiki for
1802 * The CLI command 'manager show commands' no longer truncates command names
1803 longer than 15 characters and no longer shows authorization requirement
1804 for commands. 'manager show command' now displays the privileges needed
1805 for using a given manager command instead.
1807 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
1808 peer in its response if the peer has a subscribe context set.
1810 * The SIPqualifypeer action now acknowledges the request once it has
1811 established that the request is against a known peer. It also issues a new
1812 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
1814 * The PlayDTMF action now supports an optional 'Duration' parameter. This
1815 specifies the duration of the digit to be played, in milliseconds.
1817 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
1818 updates when changes occur instead of requiring the use of pollmailboxes.
1820 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
1821 AMI client to manipulate audio currently being played back on a channel. The
1822 supported operations depend on the application being used to send audio to
1823 the channel. When the audio playback was initiated using the ControlPlayback
1824 application or CONTROL STREAM FILE AGI command, the audio can be paused,
1825 stopped, restarted, reversed, or skipped forward. When initiated by other
1826 mechanisms (such as the Playback application), the audio can be stopped,
1827 reversed, or skipped forward.
1829 * Channel related events now contain a snapshot of channel state, adding new
1830 fields to many of these events.
1832 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
1833 in a future release. Please use the common 'Exten' field instead.
1835 * The AMI event 'UserEvent' from app_userevent now contains the channel state
1836 fields. The channel state fields will come before the body fields.
1838 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
1839 'UnParkedCall' have changed significantly in the new res_parking module.
1841 The 'Channel' and 'From' headers are gone. For the channel that was parked
1842 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
1843 has a number of fields associated with it. The old 'Channel' header relayed
1844 the same data as the new 'ParkeeChannel' header.
1846 The 'From' field was ambiguous and changed meaning depending on the event.
1847 for most of these, it was the name of the channel that parked the call
1848 (the 'Parker'). There is no longer a header that provides this channel name,
1849 however the 'ParkerDialString' will contain a dialstring to redial the
1850 device that parked the call.
1852 On UnParkedCall events, the 'From' header would instead represent the
1853 channel responsible for retrieving the parkee. It receives a channel
1854 snapshot labeled 'Retriever'. The 'from' field is is replaced with
1857 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
1859 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
1860 fashion has changed the field names 'StartExten' and 'StopExten' to
1861 'StartSpace' and 'StopSpace' respectively.
1863 * The deprecated use of | (pipe) as a separator in the channelvars setting in
1864 manager.conf has been removed.
1866 * Channel Variables conveyed with a channel no longer contain the name of the
1867 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
1868 ChanVariable: bar=baz. When multiple channels are present in a single AMI
1869 event, the various ChanVariable fields will contain a suffix that specifies
1870 which channel they correspond to.
1872 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
1873 event always conveys the AMI event for a particular channel.
1875 * All 'Reload' events have been consolidated into a single event type. This
1876 event will always contain a Module field specifying the name of the module
1877 and a Status field denoting the result of the reload. All modules now issue
1878 this event when being reloaded.
1880 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
1881 fail to receive this event due to being connected after modules have loaded.
1882 AMI connections that want to know when Asterisk is ready should listen for
1883 the 'FullyBooted' event.
1885 * app_fax now sends the same send fax/receive fax events as res_fax. The
1886 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
1887 now the 'ReceiveFAX' event.
1889 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
1890 'MusicOnHoldStop'. The sub type field has been removed.
1892 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
1893 carrier for another protocol.
1895 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
1896 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
1897 to the specific channel. 'Both' may be specified to play a tone to both
1898 channels. The old 'yes' option is still accepted as a way of playing the
1899 tone to Channel2 only.
1901 * The AMI 'Status' response event to the AMI Status action replaces the
1902 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
1903 indicate what bridge the channel is currently in.
1905 * The AMI 'Hold' event has been moved out of individual channel drivers, into
1906 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
1909 * The AMI events in app_queue have been made more consistent with each other.
1910 Events that reference channels (QueueCaller* and Agent*) will show
1911 information about each channel. The (infamous) 'Join' and 'Leave' AMI
1912 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
1914 * The 'MCID' AMI event now publishes a channel snapshot when available and
1915 its non-channel-snapshot parameters now use either the "MCallerID" or
1916 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
1917 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
1918 parameters in the channel snapshot.
1920 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
1921 'AgentLogin' and 'AgentLogoff' respectively.
1923 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
1924 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
1926 * 'ChannelUpdate' events have been removed.
1928 * All AMI events now contain a 'SystemName' field, if available.
1930 * Local channel optimization is now conveyed in two events:
1931 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
1932 when the Local channel driver begins attempting to optimize itself out of
1933 the media path; the End event is sent after the channel halves have
1934 successfully optimized themselves out of the media path.
1936 * Local channel information in events is now prefixed with 'LocalOne' and
1937 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
1938 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
1939 and 'LocalOptimizationEnd' events.
1941 * The option 'allowmultiplelogin' can now be set or overriden in a particular
1942 account. When set in the general context, it will act as the default
1943 setting for defined accounts.
1945 * The 'BridgeAction' event was removed. It technically added no value, as the
1946 Bridge Action already receives confirmation of the bridge through a
1947 successful completion Event.
1949 * The 'BridgeExec' events were removed. These events duplicated the events that
1950 occur in the Briding API, and are conveyed now through BridgeCreate,
1951 BridgeEnter, and BridgeLeave events.
1953 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
1954 previous versions. They now report all SR/RR packets sent/received, and
1955 have been restructured to better reflect the data sent in a SR/RR. In
1956 particular, the event structure now supports multiple report blocks.
1958 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
1959 raised when a blind transfer/attended transfer completes successfully.
1960 They contain information about the transfer that just completed, including
1961 the location of the transfered channel.
1963 * Added a 'security' class to AMI which outputs the required fields for
1964 security messages similar to the log messages from res_security_log
1966 * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
1967 that describes the status value in a human readable string.
1969 CDR (Call Detail Records)
1971 * Significant changes have been made to the behavior of CDRs. The CDR engine
1972 was effectively rewritten and built on the Stasis message bus. For a full
1973 definition of CDR behavior in Asterisk 12, please read the specification
1974 on the Asterisk wiki (wiki.asterisk.org).
1976 * CDRs will now be created between all participants in a bridge. For each
1977 pair of channels in a bridge, a CDR is created to represent the path of
1978 communication between those two endpoints. This lets an end user choose who
1979 to bill for what during bridge operations with multiple parties.
1981 * The duration, billsec, start, answer, and end times now reflect the times
1982 associated with the current CDR for the channel, as opposed to a cumulative
1983 measurement of all CDRs for that channel.
1985 * When a CDR is dispatched, user defined CDR variables from both parties are
1986 included in the resulting CDR. If both parties have the same variable, only
1987 the Party A value is provided.
1989 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
1990 information regarding the CDR engine is logged as verbose messages. This
1991 option should only be used if the behavior of the CDR engine needs to be
1994 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
1995 normally configured in cdr.conf.
1997 * Added CLI command 'cdr show active {channel}'. When {channel} is not
1998 specified, this command provides a summary of the channels with CDR
1999 information and their statistics. When {channel} is specified, it shows
2000 detailed information about all records associated with {channel}.
2002 CEL (Channel Event Logging)
2004 * CEL has undergone significant rework in Asterisk 12, and is now built on the
2005 Stasis message bus. Please see the specification for CEL on the Asterisk
2006 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
2009 * The 'extra' field of all CEL events that use it now consists of a JSON blob
2010 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
2012 * BLINDTRANSFER events now report the transferee bridge unique
2013 identifier, extension, and context in a JSON blob as the extra string
2014 instead of the transferee channel name as the peer.
2016 * ATTENDEDTRANSFER events now report the peer as NULL and additional
2017 information in the 'extra' string as a JSON blob. For transfers that occur
2018 between two bridged channels, the 'extra' JSON blob contains the primary
2019 bridge unique identifier, the secondary channel name, and the secondary
2020 bridge unique identifier. For transfers that occur between a bridged channel
2021 and a channel running an app, the 'extra' JSON blob contains the primary
2022 bridge unique identifier, the secondary channel name, and the app name.
2024 * LOCAL_OPTIMIZE events have been added to convey local channel
2025 optimizations with the record occurring for the semi-one channel and
2026 the semi-two channel name in the peer field.
2028 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
2029 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
2030 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
2031 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
2032 regardless of whether or not that bridge happens to contain multiple
2037 * When compiled with '--enable-dev-mode', the astobj2 library will now add
2038 several CLI commands that allow for inspection of ao2 containers that
2039 register themselves with astobj2. The CLI commands are 'astobj2 container
2040 dump', 'astobj2 container stats', and 'astobj2 container check'.
2042 * Added specific CLI commands for bridge inspection. This includes 'bridge
2043 show all', which lists all bridges in the system, and 'bridge show {id}',
2044 which provides specific information about a bridge.
2046 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
2047 ejecting the channels currently in the bridge. If the channels cannot
2048 continue in the dialplan or application that put them in the bridge, they
2051 * Added command 'bridge kick'. This will eject a single channel from a bridge.
2053 * Added commands to inspect and manipulate the registered bridge technologies.
2054 This include 'bridge technology show', which lists the registered bridge
2055 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
2056 which controls whether or not a registered bridge technology can be used
2057 during smart bridge operations. If a technology is suspended, it will not
2058 be used when a bridge technology is picked for channels; when unsuspended,
2059 it can be used again.
2061 * The command 'config show help {module} {type} {option}' will show
2062 configuration documentation for modules with XML configuration
2063 documentation. When {module}, {type}, and {option} are omitted, a listing
2064 of all modules with registered documentation is displayed. When {module}
2065 is specified, a listing of all configuration types for that module is
2066 displayed, along with their synopsis. When {module} and {type} are
2067 specified, a listing of all configuration options for that type are
2068 displayed along with their synopsis. When {module}, {type}, and {option}
2069 are specified, detailed information for that configuration option is
2072 * Added 'core show sounds' and 'core show sound' CLI commands. These display
2073 a listing of all installed media sounds available on the system and
2074 detailed information about a sound, respectively.
2076 * 'xmldoc dump' has been added. This CLI command will dump the XML
2077 documentation DOM as a string to the specified file. The Asterisk core
2078 will populate certain XML elements pulled from the source files with
2079 additional run-time information; this command lets a user produce the
2080 XML documentation with all information.
2084 * Parking has been pulled from core and placed into a separate module called
2085 res_parking. See Parking changes below for more details. Configuration for
2086 parking should now be performed in res_parking.conf. Configuration for
2087 parking in features.conf is now unsupported.
2089 * Core attended transfers now have several new options. While performing an
2090 attended transfer, the transferer now has the following options:
2091 - *1 - cancel the attended transfer (configurable via atxferabort)
2092 - *2 - complete the attended transfer, dropping out of the call
2093 (configurable via atxfercomplete)
2094 - *3 - complete the attended transfer, but stay in the call. This will turn
2095 the call into a multi-party bridge (configurable via atxferthreeway)
2096 - *4 - swap to the other party. Once an attended transfer has begun, this
2097 options may be used multiple times (configurable via atxferswap)
2099 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
2100 must be on the channel initiating the transfer to have any effect.
2102 * The BRIDGE_FEATURES channel variable would previously only set features for
2103 the calling party and would set this feature regardless of whether the
2104 feature was in caps or in lowercase. Use of a caps feature for a letter
2105 will now apply the feature to the calling party while use of a lowercase
2106 letter will apply that feature to the called party.
2108 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
2110 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
2111 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
2112 activated the dynamic feature.
2114 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
2115 only on the channel executing the dynamic feature. Executing a dynamic
2116 feature on the bridge peer in a multi-party bridge will execute it on all
2117 peers of the activating channel.
2119 * You can now have the settings for a channel updated using the FEATURE()
2120 and FEATUREMAP() functions inherited to child channels by setting
2121 FEATURE(inherit)=yes.
2123 * automixmon now supports additional channel variables from automon including:
2124 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
2125 and TOUCH_MIXMONITOR_MESSAGE_STOP
2127 * A new general features.conf option 'recordingfailsound' has been added which
2128 allowssetting a failure sound for a user tries to invoke a recording feature
2129 such as automon or automixmon and it fails.
2131 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
2132 features.c for atxferdropcall=no to work properly. This option now just
2137 * Added log rotation strategy 'none'. If set, no log rotation strategy will
2138 be used. Given that this can cause the Asterisk log files to grow quickly,
2139 this option should only be used if an external mechanism for log management
2144 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
2145 will store the path information for that peer when it registers. Realtime
2146 tables can also use the 'supportpath' field to enable Path header support.
2148 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
2149 objectIdentifier. This maps to the supportpath option in sip.conf.
2153 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
2154 provides modules a useful abstraction on top of the many storage mechanisms
2155 in Asterisk, including the Asterisk Database, static configuration files,
2156 static Realtime, and dynamic Realtime. It also provides a caching service.
2157 Users can configure a hierarchy of data storage layers for specific modules
2160 * All future modules which utilize Sorcery for object persistence must have a
2161 column named "id" within their schema when using the Sorcery realtime module.
2162 This column must be able to contain a string of up to 128 characters in length.
2164 Security Events Framework
2166 * Security Event timestamps now use ISO 8601 formatted date/time instead of
2167 the "seconds-microseconds" format that it was using previously.
2171 * The Stasis message bus is a publish/subscribe message bus internal to
2172 Asterisk. Many services in Asterisk are built on the Stasis message bus,
2173 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
2174 Stasis can be configured in stasis.conf. Note that these parameters operate
2175 at a very low level in Asterisk, and generally will not require changes.
2179 * When a channel driver is configured to enable jiterbuffers, they are now
2180 applied unconditionally when a channel joins a bridge. If a jitterbuffer
2181 is already set for that channel when it enters, such as by the JITTERBUFFER
2182 function, then the existing jitterbuffer will be used and the one set by
2183 the channel driver will not be applied.
2187 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
2188 dialplan applications provided by the app_agent_pool module. Agents are
2189 connected with callers using the new AgentRequest dialplan application.
2190 The Agents:<agent-id> device state is available to monitor the status of an
2191 agent. See agents.conf.sample for valid configuration options.
2193 * The updatecdr option has been removed. Altering the names of channels on a
2194 CDR is not supported - the name of the channel is the name of the channel,
2195 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
2196 has also been removed, for the same reason.
2198 * The endcall and enddtmf configuration options are removed. Use the
2199 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
2200 channel before calling AgentLogin.
2204 * chan_bridge has been removed. Its functionality has been incorporated
2205 directly into the ConfBridge application itself.
2209 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
2210 of the specified span and its B-channels. Note that this command should
2211 only be used if you understand the risks it entails.
2213 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
2214 A range of channels can be specified to be destroyed. Note that this command
2215 should only be used if you understand the risks it entails.
2217 * Added the CLI command 'dahdi create channels'. A range of channels can be
2218 specified to be created, or the keyword 'new' can be used to add channels
2221 * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
2222 the exact configured mailbox name. For app_voicemail mailboxes this is
2225 * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
2229 * IPv6 support has been added. We are now able to bind to and
2230 communicate using IPv6 addresses.
2234 * The /b option has been removed.
2236 * chan_local moved into the system core and is no longer a loadable module.
2240 * Added general support for busy detection.
2242 * Added ECAM command support for Sony Ericsson phones.
2246 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
2247 SIP stack. A collection of resource modules provides the bulk of the SIP
2248 functionality. For more information on the new SIP channel driver, see
2249 https://wiki.asterisk.org/wiki/x/JYGLAQ
2253 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
2254 using the 'supportpath' setting, either on a global basis or on a peer basis.
2255 This setting enables Asterisk to route outgoing out-of-dialog requests via a
2256 set of proxies by using a pre-loaded route-set defined by the Path headers in
2257 the REGISTER request. See Realtime updates for more configuration information.
2259 * The SIP_CODEC family of variables may now specify more than one codec. Each
2260 codec must be separated by a comma. The first codec specified is the
2261 preferred codec for the offer. This allows a dialplan writer to specify both
2262 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
2264 * The 'callevents' parameter has been removed. Hold AMI events are now raised
2265 in the core, and can be filtered out using the 'eventfilter' parameter
2268 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
2269 codecs configured for a peer instead of the requested codec.
2271 * The option "register_retry_403" has been added to chan_sip to work around
2272 servers that are known to erroneously send 403 in response to valid
2273 REGISTER requests and allows Asterisk to continue attepmting to connect.
2277 * Added the 'immeddialkey' parameter. If set, when the user presses the
2278 configured key the already entered number will be immediately dialed. This
2279 is useful when the dialplan allows for variable length pattern matching.
2280 Valid options are '*' and '#'.
2282 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
2283 milliseconds) before a call forward is considered to not be answered.
2285 * The 'serviceurl' parameter allows Service URLs to be attached to line
2294 * The password option has been disabled, as the AgentLogin application no
2295 longer provides authentication.
2299 * Due to changes in the Asterisk core, this function is no longer needed to
2300 preserve a MixMonitor on a channel during transfer operations and dialplan
2301 execution. It is effectively obsolete.
2305 * The 'amaflags' and 'accountcode' attributes for the CDR function are
2306 deprecated. Use the CHANNEL function instead to access these attributes.
2308 * The 'l' option has been removed. When reading a CDR attribute, the most
2309 recent record is always used. When writing a CDR attribute, all non-finalized
2312 * The 'r' option has been removed, for the same reason as the 'l' option.
2314 * The 's' option has been removed, as LOCKED semantics no longer exist in the
2319 * A new function CDR_PROP has been added. This function lets you set properties
2320 on a channel's active CDRs. This function is write-only. Properties accept
2321 boolean values to set/clear them on the channel's CDRs. Valid properties
2323 - 'party_a' - make this channel the preferred Party A in any CDR between two
2324 channels. If two channels have this property set, the creation time of the
2325 channel is used to determine who is Party A. Note that dialed channels are
2326 never Party A in a CDR.
2327 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
2328 application when set to True, and analogous to the 'e' option in ResetCDR
2333 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
2334 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
2335 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
2338 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
2339 string, i.e., [[context],extension],priority. If set on a channel, if a
2340 channel leaves a bridge but is not hung up it will resume dialplan execution
2345 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
2346 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
2347 The value of this setting is ignored when disabled is used for the argument.
2351 * A new function provided by chan_pjsip, this function can be used in
2352 conjunction with the Dial application to construct a dial string that will
2353 dial all contacts on an Address of Record associated with a chan_pjsip
2358 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
2359 outbound channel prior to dialing.
2363 * Redirecting reasons can now be set to arbitrary strings. This means
2364 that the REDIRECTING dialplan function can be used to set the redirecting
2365 reason to any string. It also allows for custom strings to be read as the
2366 redirecting reason from SIP Diversion headers.
2370 * The SPEECH_ENGINE function now supports read operations. When read from, it
2371 will return the current value of the requested attribute.
2375 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
2376 system as mailbox@context. The rest of the system cannot add @default
2377 to mailbox identifiers for app_voicemail that do not specify a context
2378 any longer. It is a mailbox identifier format that should only be
2379 interpreted by app_voicemail.
2385 res_agi (Asterisk Gateway Interface)
2387 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
2389 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
2392 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
2393 will start the playback of the audio at the position specified. It will
2394 also return the final position of the file in 'endpos'.
2396 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
2397 channel variable if the user stopped the file playback or if a remote
2398 entity stopped the playback. If neither stopped the playback, it will
2399 indicate the overall success/failure of the playback. If stopped early,
2400 the final offset of the file will be set in the CPLAYBACKOFFSET channel
2403 * The SAY ALPHA command now accepts an additional parameter to control
2404 whether it specifies the case of uppercase, lowercase, or all letters to
2405 provide functionality similar to SayAlphaCase.
2407 res_ari (Asterisk RESTful Interface) (and others)
2409 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
2410 control telephony primitives in Asterisk by remote client. This includes
2411 channels, bridges, endpoints, media, and other fundamental concepts. Users
2412 of ARI can develop their own communications applications, controlling
2413 multiple channels using an HTTP RESTful interface and receiving JSON events
2414 about the objects via a WebSocket connection. ARI can be configured in
2415 Asterisk via ari.conf. For more information on ARI, see
2416 https://wiki.asterisk.org/wiki/x/0YCLAQ
2420 * Parking has been extracted from the Asterisk core as a loadable module,
2421 res_parking. Configuration for parking is now provided by res_parking.conf.
2422 Configuration through features.conf is no longer supported.
2424 * res_parking uses the configuration framework. If an invalid configuration is
2425 supplied, res_parking will fail to load or fail to reload. Previously,
2426 invalid configurations would generally be accepted, with certain errors
2427 resulting in individually disabled parking lots.
2429 * Parked calls are now placed in bridges. While this is largely an
2430 architectural change, it does have implications on how channels in a parking
2431 lot are viewed. For example, commands that display channels in bridges will
2432 now also display the channels in a parking lot.
2434 * The order of arguments for the new parking applications have been modified.
2435 Timeout and return context/exten/priority are now implemented as options,
2436 while the name of the parking lot is now the first parameter. See the
2437 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
2438 in-depth information as well as syntax.
2440 * Extensions are by default no longer automatically created in the dialplan to
2441 park calls or pickup parked calls. Generation of dialplan extensions can be
2442 enabled using the 'parkext' configuration option.
2444 * ADSI functionality for parking is no longer supported. The 'adsipark'
2445 configuration option has been removed as a result.
2447 * The PARKINGSLOT channel variable has been deprecated in favor of
2448 PARKING_SPACE to match the naming scheme of the new system.
2450 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
2451 channel even when the configuration option 'comebactoorigin' is enabled.
2453 * A new CLI command 'parking show' has been added. This allows a user to
2454 inspect the parking lots that are currently in use.
2455 'parking show <parkinglot>' will also show the parked calls in a specific
2458 * The CLI command 'parkedcalls' is now deprecated in favor of
2459 'parking show <parkinglot>'.
2461 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
2462 can be used to get a list of parked calls for a specific parking lot.
2464 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
2465 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
2466 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
2467 longer a required argument.
2469 * The ParkAndAnnounce application is now provided through res_parking instead
2470 of through the separate app_parkandannounce module.
2472 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
2473 by default. Instead, it will follow the timeout rules of the parking lot. The
2474 old behavior can be reproduced by using the 'c' option.
2476 * Dynamic parking lots will now fail to be created under the following
2478 - if the parking lot specified by PARKINGDYNAMIC does not exist
2479 - if they require exclusive park and parkedcall extensions which overlap
2480 with existing parking lots.
2482 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
2483 currently contain no calls. Dynamic parking lots containing parked calls
2484 will persist through the reloads without alteration.
2486 * If 'parkext_exclusive' is set for a parking lot and that extension is
2487 already in use when that parking lot tries to register it, this is now
2488 considered a parking system configuration error. Configurations which do
2489 this will be rejected.
2491 * Added channel variable PARKER_FLAT. This contains the name of the extension
2492 that would be used if 'comebacktoorigin' is enabled. This can be useful when
2493 comebacktoorigin is disabled, but the dialplan or an external control
2494 mechanism wants to use the extension in the park-dial context that was
2495 generated to re-dial the parker on timeout.
2497 res_pjsip (and many others)
2499 * A large number of resource modules make up the SIP stack based on pjsip.
2500 The chan_pjsip channel driver users these resource modules to provide
2501 various SIP functionality in Asterisk. The majority of configuration for
2502 these modules is performed in pjsip.conf. Other modules may use their
2503 own configuration files.
2505 * Added 'set_var' option for an endpoint. For each variable specified that
2506 variable gets set upon creation of a channel involving the endpoint.
2510 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
2511 them, an Asterisk-specific version of PJSIP needs to be installed.
2512 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
2514 res_statsd/res_chan_stats
2516 * A new resource module, res_statsd, has been added, which acts as a statsd
2517 client. This module allows Asterisk to publish statistics to a statsd
2518 server. In conjunction with res_chan_stats, it will publish statistics about
2519 channels to the statsd server. It can be configured via res_statsd.conf.
2523 * Device state for XMPP buddies is now available using the following format:
2524 XMPP/<client name>/<buddy address>
2525 If any resource is available the device state is considered to be not in use.
2526 If no resources exist or all are unavailable the device state is considered
2533 Realtime/Database Scripts
2535 * Asterisk previously included example db schemas in the contrib/realtime/
2536 directory of the source tree. This has been replaced by a set of database
2537 migrations using the Alembic framework. This allows you to use alembic to
2538 initialize the database for you. It will also serve as a database migration
2539 tool when upgrading Asterisk in the future.
2541 See contrib/ast-db-manage/README.md for more details.
2545 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
2546 This python script will convert an existing sip.conf file to a
2547 pjsip.conf file, for use with the chan_pjsip channel driver. This script
2548 is meant to be an aid in converting an existing chan_sip configuration to
2549 a chan_pjsip configuration, but it is expected that configuration beyond
2550 what the script provides will be needed.
2552 ------------------------------------------------------------------------------
2553 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
2554 ------------------------------------------------------------------------------
2558 * The Asterisk build system will now build and install a shared library
2559 (libasteriskssl.so) used to wrap various initialization and shutdown functions
2560 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
2561 that Asterisk can ensure that these functions do *not* get called by any
2562 modules that are loaded into Asterisk, since they should only be called once
2563 in any single process. If desired, this feature can be disabled by supplying
2564 the "--disable-asteriskssl" option to the configure script.
2566 * A new make target, 'full', has been added to the Makefile. This performs
2567 the same compilation actions as make all, but will also scan the entirety of
2568 each source file for documentation. This option is needed to generate AMI
2569 event documentation. Note that your system must have Python in order for
2570 this make target to succeed.
2572 * The optimization portion of the build system has been reworked to avoid
2573 broken builds on certain architectures. All architecture-specific
2574 optimization has been removed in favor of using -march=native to allow gcc
2575 to detect the environment in which it is running when possible. This can
2576 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
2578 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
2579 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
2581 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
2582 previously parsed the header file to obtain the version of Asterisk, you
2583 will now have to go through Asterisk to get the version information.
2591 * Added 'F()' option. Similar to the dial option, this can be supplied with
2592 arguments indicating where the callee should go after the caller is hung up,
2593 or without options specified, the priority after the Queue will be used.
2598 * Added menu action admin_toggle_mute_participants. This will mute / unmute
2599 all non-admin participants on a conference. The confbridge configuration
2600 file also allows for the default sounds played to all conference users when
2601 this occurs to be overriden using sound_participants_unmuted and
2602 sound_participants_muted.
2604 * Added menu action participant_count. This will playback the number of
2605 current participants in a conference.
2607 * Added announcement configuration option to user profile. If set the sound
2608 file will be played to the user, and only the user, upon joining the
2611 * Added record_file_append option that defaults to "yes", but if set to no
2612 will create a new file between each start/stop recording.
2617 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
2618 channels respectively before the callee channels are called.
2623 * Added support for IPv6.
2625 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
2626 external process will cause the current playlist to be cleared, including
2627 stopping any audio file that is currently playing. This is useful when you
2628 want to interrupt audio playback only when specific DTMF is entered by the
2634 * A new option, 'I' has been added to app_followme. By setting this option,
2635 Asterisk will not update the caller with connected line changes when they
2636 occur. This is similar to app_dial and app_queue.
2638 * The 'N' option is now ignored if the call is already answered.
2640 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
2641 and caller channels respectively before the callee channels are called.
2643 * The winning FollowMe outgoing call is now put on hold if the caller put it on
2649 * MixMonitor hooks now have IDs associated with them which can be used to
2650 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
2651 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
2652 now accepts that ID as an argument.
2654 * Added 'm' option, which stores a copy of the recording as a voicemail in the
2655 indicated mailboxes.
2660 * The connect action in app_mysql now allows you to specify a port number to
2661 connect to. This is useful if you run a MySQL server on a non-standard
2667 * Increased the default number of allowed destinations from 5 to 12.
2672 * The app_page application now no longer depends on DAHDI or app_meetme. It
2673 has been re-architected to use app_confbridge internally.
2678 * Added queue options autopausebusy and autopauseunavail for automatically
2679 pausing a queue member when their device reports busy or congestion.
2681 * The 'ignorebusy' option for queue members has been deprecated in favor of
2682 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
2683 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
2684 per interface basis. Individual ringinuse values can now be set in
2685 queues.conf via an argument to member definitions. Lastly, the queue
2686 'ringinuse' setting now only determines defaults for the per member
2687 'ringinuse' setting and does not override per member settings like it does
2688 in earlier versions.
2690 * Added 'F()' option. Similar to the dial option, this can be supplied with
2691 arguments indicating where the callee should go after the caller is hung up,
2692 or without options specified, the priority after the Queue will be used.
2694 * Added new option log_member_name_as_agent, which will cause the membername to
2695 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
2696 state_interface has been set.
2698 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
2700 * App_queue will now play periodic announcements for the caller that
2701 holds the first position in the queue while waiting for answer.
2705 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
2706 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
2707 changed arguments to SayUnixTime so that every option is truly optional even
2708 when using multiple options (so that j option could be used without having to
2709 manually specify timezone and format) There are other benefits, e.g., format
2710 can now be used without specifying time zone as well.
2715 * Addition of the VM_INFO function - see Function changes.
2717 * The imapserver, imapport, and imapflags configuration options can now be
2718 overriden on a user by user basis.
2720 * When voicemail plays a message's envelope with saycid set to yes, when
2721 reaching the caller id field it will play a recording of a file with the same
2722 base name as the sender's callerid if there is a similarly named file in
2723 <astspooldir>/recordings/callerids/
2725 * Voicemails now contains a unique message identifier "msg_id", which is stored
2726 in the message envelope with the sound files. IMAP backends will now store
2727 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
2728 backends will store the message identifier in a "msg_id" column. See
2729 UPGRADE.txt for more information.
2731 * Added VoiceMailPlayMsg application. This application will play a single
2732 voicemail message from a mailbox. The result of the application, SUCCESS or
2733 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
2738 * Hangup handlers can be attached to channels using the CHANNEL() function.
2739 Hangup handlers will run when the channel is hung up similar to the h
2740 extension. The hangup_handler_push option will push a GoSub compatible
2741 location in the dialplan onto the channel's hangup handler stack. The
2742 hangup_handler_pop option will remove the last added location, and optionally
2743 replace it with a new GoSub compatible location. The hangup_handler_wipe
2744 option will remove all locations on the stack, and optionally add a new
2747 * The expression parser now recognizes the ABS() absolute value function,
2748 which will convert negative floating point values to positive values.
2750 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
2751 control of faxdetect.
2753 * Addition of the VM_INFO function that can be used to retrieve voicemail
2754 user information, such as the email address and full name.
2755 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
2758 * The REDIRECTING function now supports the redirecting original party id
2761 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
2762 lets you set some of the configuration options from the [general] section
2763 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
2764 the key sequence used to activate built-in features, such as blindxfer,
2765 and automon. See the built-in documentation for details.
2767 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
2768 instead of simply the uri. This is the format that MessageSend() can use
2769 in the from parameter for outgoing SIP messages.
2771 * Added the PRESENCE_STATE function. This allows retrieving presence state
2772 information from any presence state provider. It also allows setting
2773 presence state information from a CustomPresence presence state provider.
2774 See AMI/CLI changes for related commands.
2776 * Added the AMI_CLIENT function to make manager account attributes available
2777 to the dialplan. It currently supports returning the current number of
2778 active sessions for a given account.
2780 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
2781 and the REDIRECTING functions.
2789 * Added a manager event "LocalBridge" for local channel call bridges between
2790 the two pseudo-channels created.
2795 * Added dialtone_detect option for analog ports to disconnect incoming
2796 calls when dialtone is detected.
2798 * Added option colp_send to send ISDN connected line information. Allowed
2799 settings are block, to not send any connected line information; connect, to
2800 send connected line information on initial connect; and update, to send
2801 information on any update during a call. Default is update.
2803 * Add options namedcallgroup and namedpickupgroup to support installations
2804 where a higher number of groups (>64) is required.
2806 * Added support to use private party ID information with PRI calls.
2811 * A new channel driver named chan_motif has been added which provides support for
2812 Google Talk and Jingle in a single channel driver. This new channel driver includes
2813 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
2814 hold, unhold, and ringing notification. It is also compliant with the current Jingle
2815 specification, current Google Jingle specification, and the original Google Talk
2821 * Added NAT support for RTP. Setting in config is 'nat', which can be set
2822 globally and overriden on a peer by peer basis.
2824 * Direct media functionality has been added. Options in config are:
2825 directmedia (directrtp) and directrtpsetup (earlydirect)
2827 * ChannelUpdate events now contain a CallRef header.
2832 * Asterisk will no longer substitute CID number for CID name in the display
2833 name field if CID number exists without a CID name. This change improves
2834 compatibility with certain device features such as Avaya IP500's directory
2837 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
2838 created using that setting to not be removed during SIP reload.
2840 * Added settings recordonfeature and recordofffeature. When receiving an INFO
2841 request with a "Record:" header, this will turn the requested feature on/off.
2842 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
2843 dynamic features must be enabled and configured properly on the requesting
2844 channel for this to function properly.
2846 * Add support to realtime for the 'callbackextension' option.
2848 * When multiple peers exist with the same address, but differing
2849 callbackextension options, incoming requests that are matched by address
2850 will be matched to the peer with the matching callbackextension if it is
2853 * Two new NAT options, auto_force_rport and auto_comedia, have been added
2854 which set the force_rport and comedia options automatically if Asterisk
2855 detects that an incoming SIP request crossed a NAT after being sent by
2856 the remote endpoint.
2858 * The default global nat setting in sip.conf has been changed from force_rport
2859 to auto_force_rport.
2861 * NAT settings are now a combinable list of options. The equivalent of the
2862 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
2864 * Adds an option send_diversion which can be disabled to prevent
2865 diversion headers from automatically being added to INVITE requests.
2867 * Add support for lightweight NAT keepalive. If enabled a blank packet will
2868 be sent to the remote host at a given interval to keep the NAT mapping open.
2869 This can be enabled using the keepalive configuration option.
2871 * Add option 'tonezone' to specify country code for indications. This option
2872 can be set both globally and overridden for specific peers.
2874 * The SIP Security Events Framework now supports IPv6.
2876 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
2877 between multiple user agents. When set, for directmedia reinvites,
2878 Asterisk will not send an immediate reinvite on an incoming call leg. This
2879 option is useful when peered with another SIP user agent that is known to
2880 send immediate direct media reinvites upon call establishment.
2882 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
2885 * Add options subminexpiry and submaxexpiry to set limits of subscription
2886 timer independently from registration timer settings. The setting of the
2887 registration timer limits still is done by options minexpiry, maxexpiry
2888 and defaultexpiry. For backwards compatibility the setting of minexpiry
2889 and maxexpiry also is used to configure the subscription timer limits if
2890 subminexpiry and submaxexpiry are not set in sip.conf.
2892 * Set registration timer limits to default values when reloading sip
2893 configuration and values are not set by configuration.
2895 * Add options namedcallgroup and namedpickupgroup to support installations
2896 where a higher number of groups (>64) is required.
2898 * When a MESSAGE request is received, the address the request was received from
2899 is now saved in the SIP_RECVADDR variable.
2901 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
2902 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
2903 the ANI2/OLI information is set on the channel, which can be retrieved using
2904 the CALLERID function.
2906 * Peers can now be configured to support negotiation of ICE candidates using
2907 the setting icesupport. See res_rtp_asterisk changes for more information.
2909 * Added support for format attribute negotiation. See the Codecs changes for
2912 * Extra headers specified with SIPAddHeader are sent with the REFER message
2913 when using Transfer application. See refer_addheaders in sip.conf.sample.
2915 * Added support to use private party ID information with calls.
2917 * Adds an option discard_remote_hold_retrieval that when set stops telling
2918 the peer to start music on hold.
2923 * Added skinny version 17 protocol support.
2927 --------------------
2928 * Added option 'dtmf_duration' allowing playback time of DTMF tones to be set
2930 * Modified option 'date_format' to allow options to display date in 31Jan and Jan31
2931 formats as options 0 and 1. The previous options 0 and 1 now map to options 2 and 3
2932 as per the UNISTIM protocol.
2934 * Fixed issues with dialtone not matching indications.conf and mute stopping rx
2935 as well as tx. Also fixed issue with call "Timer" displaying as French "Dur\E9e"
2937 * Added ability to use multiple lines for a single phone. This allows multiple
2938 calls to occur on a single phone, using callwaiting and switching between calls.
2940 * Added option 'sharpdial' allowing end dialing by pressing # key
2942 * Added option 'interdigit_timer' to control phone dial timeout
2944 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
2946 * Added global 'debug' option, that enables debug in channel driver
2948 * Added ability to translate on-screen menu in multiple languages. Tested on
2949 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
2950 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
2953 * In addition to English added French and Russian languages for on-screen menus
2955 * Reworked dialing number input: added dialing by timeout, immediate dial on
2956 on dialplan compare, phone number length now not limited by screen size
2958 * Added ability to pickup a call using features.conf defined value and
2964 * Add options namedcallgroup and namedpickupgroup to support installations
2965 where a higher number of groups (>64) is required.
2967 * Added support to use private party ID information with calls.
2972 * The minimum DTMF duration can now be configured in asterisk.conf
2973 as "mindtmfduration". The default value is (as before) set to 80 ms.
2974 (previously it was only available in source code)
2976 * Named ACLs can now be specified in acl.conf and used in configurations that
2977 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
2978 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
2979 working ACL. In addition, some CLI commands have been added to provide
2980 show information and allow for module reloading - see CLI Changes.
2982 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
2983 items (separated by commas), and items in the rule can be negated by prefixing
2984 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
2985 longer necessray to control the order that the 'permit' and 'deny' columns are
2986 returned from queries.
2988 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
2989 be used within the dynamic weight attribute when specifying a mapping.
2991 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
2992 header, instead of putting the user defined event name there. When enabled
2993 the UserDefType header is added for user defined events. This feature is
2994 enabled with the setting show_user_defined.
2996 * Macro has been deprecated in favor of GoSub. For redirecting and connected
2997 line purposes use the following variables instead of their macro equivalents:
2998 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
2999 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
3000 cc_callback_macro in channel configurations.
3002 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
3005 * Call files now support the "early_media" option to connect with an outgoing
3006 extension when early media is received.
3008 * Added support to use private party ID information with calls.
3013 * A new channel variable, AGIEXITONHANGUP, has been added which allows
3014 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
3015 AGI application would exit immediately after a channel hangup is detected.
3017 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
3018 are resolved and each address is attempted in turn until one succeeds or
3022 AMI (Asterisk Manager Interface)
3024 * The originate action now has an option "EarlyMedia" that enables the
3025 call to bridge when we get early media in the call. Previously,
3026 early media was disregarded always when originating calls using AMI.
3028 * Added setvar= option to manager accounts (much like sip.conf)
3030 * Originate now generates an error response if the extension given is not found
3033 * MixMonitor will now show IDs associated with the mixmonitor upon creating
3034 them if the i(variable) option is used. StopMixMonitor will accept
3035 MixMonitorID as an option to close specific MixMonitors.
3037 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
3038 updated to include information about peers configured with
3039 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
3040 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
3041 returned if auto_force_rport is not enabled.
3043 * Added SIPpeerstatus manager command which will generate PeerStatus events
3044 similar to the existing PeerStatus events found in chan_sip on demand.
3046 * Hangup now can take a regular expression as the Channel option. If you want
3047 to hangup multiple channels, use /regex/ as the Channel option. Existing
3048 behavior to hanging up a single channel is unchanged, but if you pass a regex,
3049 the manager will send you a list of channels back that were hung up.
3051 * Support for IPv6 addresses has been added.
3053 * AMI Events can now be documented in the Asterisk source. Note that AMI event
3054 documentation is only generated when Asterisk is compiled using 'make full'.
3055 See the CLI section for commands to display AMI event information.
3057 * The AMI Hangup event now includes the AccountCode header so you can easily
3058 correlate with AMI Newchannel events.
3060 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
3061 the StateInterface of the queue member.
3063 * Added AMI event SessionTimeout in the Call category that is issued when a
3064 call is terminated due to either RTP stream inactivity or SIP session timer
3067 * CEL events can now contain a user defined header UserDefType. See core
3068 changes for more information.
3070 * OOH323 ChannelUpdate events now contain a CallRef header.
3072 * Added PresenceState command. This command will report the presence state for
3073 the given presence provider.
3075 * Added Parkinglots command. This will list all parking lots as a series of
3076 AMI Parkinglot events.
3078 * Added MessageSend command. This behaves in the same manner as the
3079 MessageSend application, and is a technolgoy agnostic mechanism to send out
3080 of call text messages.
3082 * Added "message" class authorization. This grants an account permission to
3083 send out of call messages. Write-only.
3088 * The "dialplan add include" command has been modified to create context a context
3089 if one does not already exist. For instance, "dialplan add include foo into bar"
3090 will create context "bar" if it does not already exist.
3092 * A "dialplan remove context" command has been added to remove a context from
3095 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
3096 filenames of all running mixmonitors on a channel.
3098 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
3099 numeric instead of 0, 1, or 2.
3101 * "stun show status" will show a table describing how the STUN client is
3104 * "acl show [named acl]" will show information regarding a Named ACL. The
3105 acl module can be reloaded with "reload acl".
3107 * Added CLI command to display AMI event information - "manager show events",
3108 which shows a list of all known and documented AMI events, and "manager show
3109 event [event name]", which shows detail information about a specific AMI
3112 * The result of the CLI command "queue show" now includes the state interface
3113 information of the queue member.
3115 * The command "core set verbose" will now set a separate level of logging for
3116 each remote console without affecting any other console.
3118 * Added command "cdr show pgsql status" to check connection status
3120 * "sip show channel" will now display the complete route set.
3122 * Added "presencestate list" command. This command will list all custom
3123 presence states that have been set by using the PRESENCE_STATE dialplan
3126 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
3127 command. This changes a custom presence to a new state.
3132 * Codec lists may now be modified by the '!' character, to allow succinct
3133 specification of a list of codecs allowed and disallowed, without the
3134 requirement to use two different keywords. For example, to specify all
3135 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
3137 * Add support for parsing SDP attributes, generating SDP attributes, and
3138 passing it through. This support includes codecs such as H.263, H.264, SILK,
3139 and CELT. You are able to set up a call and have attribute information pass.
3140 This should help considerably with video calls.
3142 * The iLBC codec can now use a system-provided iLBC library if one is installed,
3143 just like the GSM codec.
3147 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
3148 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
3152 * Asterisk version and build information is now logged at the beginning of a
3155 * Threads belonging to a particular call are now linked with callids which get
3156 added to any log messages produced by those threads. Log messages can now be
3157 easily identified as involved with a certain call by looking at their call id.
3158 Call ids may also be attached to log messages for just about any case where
3159 it can be determined to be related to a particular call.
3161 * Each logging destination and console now have an independent notion of the
3162 current verbosity level. Logger.conf now allows an optional argument to
3163 the 'verbose' specifier, indicating the level of verbosity sent to that
3164 particular logging destination. Additionally, remote consoles now each
3165 have their own verbosity level. The command 'core set verbose' will now set
3166 a separate level for each remote console without affecting any other
3172 * Added 'announcement' option which will play at the start of MOH and between
3173 songs in modes of MOH that can detect transitions between songs (eg.
3179 * New per parking lot options: comebackcontext and comebackdialtime. See
3180 configs/features.conf.sample for more details.
3182 * Channel variable PARKER is now set when comebacktoorigin is disabled in
3185 * Channel variable PARKEDCALL is now set with the name of the parking lot
3186 when a timeout occurs.
3192 CDR Postgresql Driver
3194 * Added command "cdr show pgsql status" to check connection status
3197 CDR Adaptive ODBC Driver
3199 * Added schema option for databases that support specifying a schema.
3207 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
3208 CALENDAR_WRITE has completed successfully.
3213 * A new option, 'probation' has been added to rtp.conf
3214 RTP in strictrtp mode can now require more than 1 packet to exit learning
3215 mode with a new source (and by default requires 4). The probation option
3216 allows the user to change the required number of packets in sequence to any
3217 desired value. Use a value of 1 to essentially restore the old behavior.
3218 Also, with strictrtp on, Asterisk will now drop all packets until learning
3219 mode has successfully exited. These changes are based on how pjmedia handles
3220 media sources and source changes.
3222 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
3223 enabled or disabled using the icesupport setting. A variety of other
3224 settings have been introduced to configure STUN/TURN connections.
3229 * A new module, res_corosync, has been introduced. This module uses the
3230 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
3231 of Asterisk servers to both Message Waiting Indication (MWI) and/or
3232 Device State (presence) information. This module is very similar to, and
3233 is a replacement for the res_ais module that was in previous releases of
3239 * This module adds a cleaned up, drop-in replacement for res_jabber called
3240 res_xmpp. This provides the same externally facing functionality but is
3241 implemented differently internally. res_jabber has been deprecated in favor
3242 of res_xmpp; please see the UPGRADE.txt file for more information.
3247 * The safe_asterisk script has been updated to allow several of its parameters
3248 to be set from environment variables. This also enables a custom run
3249 directory of Asterisk to be specified, instead of defaulting to /tmp.
3251 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
3252 its value to determine the directory to assume is the top-level directory of
3253 the source tree. If the variable is not set, it defaults to the current
3254 behavior and uses the current working directory.
3256 ------------------------------------------------------------------------------
3257 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
3258 ------------------------------------------------------------------------------
3262 * Asterisk now has protocol independent support for processing text messages
3263 outside of a call. Messages are routed through the Asterisk dialplan.
3264 SIP MESSAGE and XMPP are currently supported. There are options in
3265 jabber.conf and sip.conf to allow enabling these features.
3266 -> jabber.conf: see the "sendtodialplan" and "context" options.
3267 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
3268 and "outofcall_message_context" options.
3269 The MESSAGE() dialplan function and MessageSend() application have been
3270 added to go along with this functionality. More detailed usage information
3271 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
3272 * If real-time text support (T.140) is negotiated, it will be preferred for
3273 sending text via the SendText application. For example, via SIP, messages
3274 that were once sent via the SIP MESSAGE request would be sent via RTP if
3275 T.140 text is negotiated for a call.
3279 * parkedmusicclass can now be set for non-default parking lots.
3281 Asterisk Manager Interface
3282 --------------------------
3283 * PeerStatus now includes Address and Port.
3284 * Added Hold events for when the remote party puts the call on and off hold
3285 for chan_dahdi ISDN channels.
3286 * Added new action MeetmeListRooms to list active conferences (shows same
3287 data as "meetme list" at the CLI).
3288 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
3289 Description field that is set by 'description' in the channel configuration
3291 * Added Uniqueid header to UserEvent.
3292 * Added new action FilterAdd to control event filters for the current session.
3293 This requires the system permission and uses the same filter syntax as
3294 filters that can be defined in manager.conf
3295 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
3296 versions had some instances of the event converted, but others were left
3297 as-is. All Unlink events should now be converted to Bridge events. The AMI
3298 protocol version number was incremented to 1.2 as a result of this change.
3300 Asterisk HTTP Server
3301 --------------------------
3302 * The HTTP Server can bind to IPv6 addresses.
3305 --------------------------
3306 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
3307 with busydetect. usage example: busypattern=200,200,200,600
3310 --------------------------
3311 * New 'gtalk show settings' command showing the current settings loaded from
3313 * The 'logger reload' command now supports an optional argument, specifying an
3314 alternate configuration file to use.
3315 * 'dialplan add extension' command will now automatically create a context if
3316 the specified context does not exist with a message indicated it did so.
3317 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
3318 Description field which can be populated with 'description' in the channel
3319 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
3322 --------------------------
3323 * The filter option in cdr_adaptive_odbc now supports negating the argument,
3324 thus allowing records which do NOT match the specified filter.
3325 * Added ability to log CONGESTION calls to CDR
3328 --------------------------
3329 * Ability to define custom SILK formats in codecs.conf.
3330 * Addition of speex32 audio format with translation.
3331 * CELT codec pass-through support and ability to define
3332 custom CELT formats in codecs.conf.
3333 * Ability to read raw signed linear files with sample rates
3334 ranging from 8khz - 192khz. The new file extensions introduced
3335 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
3336 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
3337 Skinny, H.323, etc) can still only support the following codecs:
3338 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
3339 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
3340 Video: h261, h263, h263p, h264, mpeg4
3345 --------------------------
3346 * New highly optimized and customizable ConfBridge application capable of
3347 mixing audio at sample rates ranging from 8khz-96khz.
3348 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
3349 and bridge profiles on a channel.
3350 * CONFBRIDGE_INFO dialplan function capable of retrieving information
3351 about a conference such as locked status and number of parties, admins,
3353 * Addition of video_mode option in confbridge.conf for adding video support
3354 into a bridge profile.
3355 * Addition of the follow_talker video_mode in confbridge.conf. This video
3356 mode dynamically switches the video feed to always display the loudest talker
3357 supplying video in the conference.
3361 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
3362 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
3363 variables from asterisk.conf.
3367 * Addition of the JITTERBUFFER dialplan function. This function allows
3368 for jitterbuffering to occur on the read side of a channel. By using
3369 this function conference applications such as ConfBridge and MeetMe can
3370 have the rx streams jitterbuffered before conference mixing occurs.
3371 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
3373 * Added STRREPLACE function. This function let's the user search a variable
3374 for a given string to replace with another string as many times as the
3375 user specifies or just throughout the whole string.
3376 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
3377 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
3378 * Added extensions to chan_ooh323 in function CHANNEL()
3380 libpri channel driver (chan_dahdi) DAHDI changes
3381 --------------------------
3382 * Added moh_signaling option to specify what to do when the channel's bridged
3383 peer puts the ISDN channel on hold.
3384 * Added display_send and display_receive options to control how the display ie
3385 is handled. To send display text from the dialplan use the SendText()
3386 application when the option is enabled.
3387 * Added mcid_send option to allow sending a MCID request on a span.
3390 --------------------------
3391 * Added setvar option to calendar.conf to allow setting channel variables on
3392 notification channels.
3393 * Added "calendar show types" CLI command to list registered calendar
3397 --------------------------
3398 * Added two new options, r and t with file name arguments to record
3399 single direction (unmixed) audio recording separate from the bidirectional
3400 (mixed) recording. The mixed file name argument is optional now as long
3401 as at least one recording option is used.
3404 --------------------------
3405 * Added a new option, l, which will disable local call optimization for
3406 channels involved with the FollowMe thread. Use this option to improve
3407 compatability for a FollowMe call with certain dialplan apps, options, and
3411 --------------------------
3412 * Added option "k" that will automatically close the conference when there's
3413 only one person left when a user exits the conference.
3416 --------------------------
3417 * cel_pgsql now supports the 'extra' column for data added using the
3418 CELGenUserEvent() application.
3421 --------------------------
3422 * Support for defining hints has been added to pbx_lua. See the 'hints' table
3423 in the sample extensions.lua file for syntax details.
3424 * Applications that perform jumps in the dialplan such as Goto will now
3425 execute properly. When pbx_lua detects that the context, extension, or
3426 priority we are executing on has changed it will immediately return control
3427 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
3428 the priority after the currently executing priority.
3429 * An autoservice is now started by default for pbx_lua channels. It can be
3430 stopped and restarted using the autoservice_stop() and autoservice_start()
3434 --------------------------
3435 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
3436 into a FAXStatus event with an 'Operation' header that will be either
3437 'send', 'receive', and 'gateway'.
3438 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
3439 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
3440 feature will handle converting a fax call between an audio T.30 fax terminal
3441 and an IFP T.38 fax terminal.
3445 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
3446 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
3447 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
3451 * Added general option negative_penalty_invalid default off. when set
3452 members are seen as invalid/logged out when there penalty is negative.
3453 for realtime members when set remove from queue will set penalty to -1.
3454 * Added queue option autopausedelay when autopause is enabled it will be
3455 delayed for this number of seconds since last successful call if there
3456 was no prior call the agent will be autopaused immediately.
3457 * Added member option ignorebusy this when set and ringinuse is not
3458 will allow per member control of multiple calls as ringinuse does for
3463 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
3465 * Added 'k' option to MeetMe to automatically kill the conference when there's only
3466 one participant left (much like a normal call bridge)
3467 * Added extra argument to Originate to set timeout.
3471 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
3472 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
3473 utility in the UTILS section of menuselect. If an existing astdb is found and no
3474 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
3475 convert an existing astdb to the SQLite3 version automatically at runtime.
3479 * Modules marked as deprecated are no longer marked as building by default. Enabling
3480 these modules is still available via menuselect.
3484 * authdebug is now disabled by default. To enable this functionaility again
3485 set authdebug = yes in iax.conf.
3489 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
3490 releases it was disabled.
3494 * The PBX core previously made a call with a non-existing extension test for
3495 extension s@default and jump there if the extension existed.
3496 This was a bad default behaviour and violated the principle of least surprise.
3497 It has therefore been changed in this release. It may affect some
3498 applications and configurations that rely on this behaviour. Most channel
3499 drivers have avoided this for many releases by testing whether the extension
3500 called exists before starting the PBX and generating a local error.
3501 This behaviour still exists and works as before.
3503 Extension "s" is used when no extension is given in a channel driver,
3504 like immediate answer in DAHDI or calling to a domain with no user part
3507 ------------------------------------------------------------------------------
3508 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
3509 ------------------------------------------------------------------------------
3513 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
3514 now defaults to force_rport. It is very important that phones requiring nat=no be
3515 specifically set as such instead of relying on the default setting. If at all
3516 possible, all devices should have nat settings configured in the general section as
3517 opposed to configuring nat per-device.
3518 * Added preferred_codec_only option in sip.conf. This feature limits the joint
3519 codecs sent in response to an INVITE to the single most preferred codec.
3520 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
3521 to be used for the outgoing call. It must be one of the codecs configured
3523 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
3524 to be used for holding a private key. If tlsprivatekey is not specified,
3525 tlscertfile is searched for both public and private key.
3526 * Added tlsclientmethod option to sip.conf. This allows the protocol for
3527 outbound client connections to be specified.
3528 * The sendrpid parameter has been expanded to include the options
3529 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
3530 header to be sent (equivalent to setting sendrpid=yes) and setting
3531 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
3532 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
3533 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
3534 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
3535 will accept the SDP even if the SDP version number is not properly incremented,
3536 but will generate a warning in the log indicating that the SIP peer that sent
3537 the SDP should have the 'ignoresdpversion' option set.
3538 * The 'nat' option has now been been changed to have yes, no, force_rport, and
3539 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
3540 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
3541 remote side requests it and disables symmetric RTP support. Setting it to
3542 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
3543 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
3544 and enables symmetric RTP support.
3545 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
3546 response. This permits the master channel to know how each channel dialled
3547 in a multi-channel setup resolved in an individual way. This carries a
3548 performance penalty and can be disabled in sip.conf using the
3549 'storesipcause' option.
3550 * Added 'externtcpport' and 'externtlsport' options to allow custom port
3551 configuration for the externip and externhost options when tcp or tls is used.
3552 * Added support for message body (stored in content variable) to SIP NOTIFY message
3553 accessible via AMI and CLI.
3554 * Added 'media_address' configuration option which can be used to explicitly specify
3555 the IP address to use in the SDP for media (audio, video, and text) streams.
3556 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
3557 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
3559 * Added 'use_q850_reason' configuration option for generating and parsing
3560 if available Reason: Q.850;cause=<cause code> header. It is implemented
3561 in some gateways for better passing PRI/SS7 cause codes via SIP.
3562 * When dialing SIP peers, a new component may be added to the end of the dialstring
3563 to indicate that a specific remote IP address or host should be used when dialing
3564 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
3565 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
3566 ability to selectively force bridged channels to also be encrypted is also
3567 implemented. Branching in the dialplan can be done based on whether or not
3568 a channel has secure media and/or signaling.
3569 * Added directmediapermit/directmediadeny to limit which peers can send direct media
3571 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
3572 Charge messages to snom phones.
3573 * Added support for G.719 media streams.
projects / asterisk/asterisk.git / blob