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.
203 * Added field LastPause on QueueMemberStatus for time when started the last
204 pause for a queue member.
205 * Show the time when started the last pause for queue member on CLI for command
209 ------------------------------------------------------------------------------
210 --- Functionality changes from Asterisk 13.7.0 to Asterisk 13.8.0 ------------
211 ------------------------------------------------------------------------------
215 * This module is the successor of res_pjsip_log_forwarder. As well as
216 handling the log forwarding (which now displays as 'pjproject:0' instead
217 of 'pjsip:0'), it also adds a 'pjproject show buildopts' command to the CLI.
218 This displays the compiled-in options of the pjproject installation
219 Asterisk is currently running against.
224 * Added new global option (regcontext) to pjsip. When set, Asterisk will
225 dynamically create and destroy a NoOp priority 1 extension
226 for a given endpoint who registers or unregisters with us.
230 * A new module, res_pjsip_history, has been added that provides SIP history
231 viewing/filtering from the CLI. The module is intended to be used on systems
232 with busy SIP traffic, where existing forms of viewing SIP messages - such
233 as the res_pjsip_logger - may be inadequate. The module provides two new
235 - 'pjsip set history {on|off|clear}' - this enables/disables SIP history
236 capturing, as well as clears an existing history capture. Note that SIP
237 packets captured are stored in memory until cleared. As a result, the
238 history capture should only be used for debugging/viewing purposes, and
239 should *NOT* be left permanently enabled on a system.
240 - 'pjsip show history' - displays the captured SIP history. When invoked
241 with no options, the entire captured history is displayed. Two options
243 -- 'entry <num>' - display a detailed view of a single SIP message in
245 -- 'where ...' - filter the history based on some expression. For more
246 information on filtering, view the current CLI help for the
247 'pjsip show history' command.
251 * app_voicemail and res_mwi_external can now be built together. The default
252 remains to build app_voicemail and not res_mwi_external but if they are
253 both built, the load order will cause res_mwi_external to load first and
254 app_voicemail will be skipped. Use 'preload=app_voicemail.so' in
255 modules.conf to force app_voicemail to be the voicemail provider.
259 * A new option (bind_rtp_to_media_address) has been added to endpoint which
260 will cause res_pjsip_sdp_rtp to actually bind the RTP instance to the
261 media_address as well as using it in the SDP. If set, RTP packets will now
262 originate from the media address instead of the operating system's "primary"
267 * A new configuration section - ice_host_candidates - has been added to
268 rtp.conf, allowing automatically discovered ICE host candidates to be
269 overriden. This allows an Asterisk server behind a 1:1 NAT to send its
270 external IP as a host candidate rather than relying on STUN to discover it.
272 ------------------------------------------------------------------------------
273 --- Functionality changes from Asterisk 13.6.0 to Asterisk 13.7.0 ------------
274 ------------------------------------------------------------------------------
278 * Added format attribute negotiation for the VP8 video codec. Format attribute
279 negotiation is provided by the res_format_attr_vp8 module.
283 * A new "timeout" user profile option has been added. This configures the number
284 of seconds that a participant may stay in the ConfBridge after joining. When
285 the time expires, the user is ejected from the conference and CONFBRIDGE_RESULT
286 is set to "TIMEOUT" on the channel.
290 * The websockets_enabled option has been added to the general section of
291 sip.conf. The option is enabled by default to match the previous behavior.
292 The option should be disabled when using res_pjsip_transport_websockets to
293 ensure chan_sip will not conflict with PJSIP websockets.
297 * The HOLD_INTERCEPT dialplan function now actually exists in the source tree.
298 While support for the events was added in Asterisk 13.4.0, the function
299 accidentally never made it in. That function is now present, and will cause
300 the 'hold' raised by a channel to be intercepted and converted into an
303 res_pjsip_outbound_registration
304 -------------------------------
305 * If res_statsd is loaded and a StatsD server is configured, basic statistics
306 regarding the state of outbound registrations will now be emitted. This
308 - A GAUGE statistic for the overall number of outbound registrations, i.e.:
309 PJSIP.registrations.count
310 - A GAUGE statistic for the overall number of outbound registrations in a
311 particular state, e.g.:
312 PJSIP.registrations.state.Registered
316 * The ability to use "like" has been added to the pjsip list and show
317 CLI commands. For instance: CLI> pjsip list endpoints like abc
319 * If res_statsd is loaded and a StatsD server is configured, basic statistics
320 regarding the state of PJSIP contacts will now be emitted. This includes:
321 - A GAUGE statistic for the overall number of contacts in a particular
323 PJSIP.contacts.states.Reachable
324 - A TIMER statistic for the RTT time for each qualified contact, e.g.:
325 PJSIP.contacts.alice@@127.0.0.1:5061.rtt
327 res_sorcery_memory_cache
328 ------------------------
329 * A new caching strategy, full_backend_cache, has been added which caches
330 all stored objects in the backend. When enabled all objects will be
331 expired or go stale according to the configuration. As well when enabled
332 all retrieval operations will be performed against the cache instead of
337 * CALLERID(pres) is now documented as a valid alternative to setting both
338 CALLERID(name-pres) and CALLERID(num-pres) at once. Some channel drivers,
339 like chan_sip, don't make a distinction between the two: they take the
340 least public value from name-pres and num-pres. By using CALLERID(pres)
341 for reading and writing, you touch the same combined value in the dialplan.
342 The same applies to CONNECTEDLINE(pres), REDIRECTING(orig-pres),
343 REDIRECTING(to-pres) and REDIRECTING(from-pres).
347 * A new module that emits StatsD statistics regarding Asterisk endpoints.
348 This includes a total count of the number of endpoints, the count of the
349 number of endpoints in the technology agnostic state of the endpoint -
350 online or offline - as well as the number of channels associated with each
351 endpoint. These are recorded as three different GAUGE statistics:
353 - endpoints.state.{unknown|offline|online}
354 - endpoints.{tech}.{resource}.channels
357 ------------------------------------------------------------------------------
358 --- Functionality changes from Asterisk 13.5.0 to Asterisk 13.6.0 ------------
359 ------------------------------------------------------------------------------
363 * The CHANNEL function, when used on a PJSIP channel, now exposes a 'call-id'
364 extraction option when using with the 'pjsip' signalling option. It will
365 return the SIP Call-ID associated with the INVITE request that established
370 * Two new endpoint related events are now available: PeerStatusChange and
371 ContactStatusChange. In particular, these events are useful when subscribing
372 to all event sources, as they provide additional endpoint related
373 information beyond the addition/removal of channels from an endpoint.
375 * Added the ability to subscribe to all ARI events in Asterisk, regardless
376 of whether the application 'controls' the resource. This is useful for
377 scenarios where an ARI application merely wants to observe the system,
378 as opposed to control it. There are two ways to accomplish this:
379 (1) Via the WebSocket connection URI. A new query paramter, 'subscribeAll',
380 has been added that, when present and True, will subscribe all
381 specified applications to all ARI event sources in Asterisk.
382 (2) Via the applications resource. An ARI client can, at any time, subscribe
383 to all resources in an event source merely by not providing an explicit
384 resource. For example, subscribing to an event source of 'channels:'
385 as opposed to 'channels:12345' will subscribe the application to all
388 ------------------------------------------------------------------------------
389 --- Functionality changes from Asterisk 13.4.0 to Asterisk 13.5.0 ------------
390 ------------------------------------------------------------------------------
394 * A new ContactStatus event has been added that reflects res_pjsip contact
395 lifecycle changes: Created, Removed, Reachable, Unreachable, Unknown.
397 * Added the Linkedid header to the common channel headers listed for each
398 channel in AMI events.
402 * A new feature has been added that enables the retrieval of modules and
403 module information through an HTTP request. Information on a single module
404 can be also be retrieved. Individual modules can be loaded to Asterisk, as
405 well as unloaded and reloaded.
407 * A new resource has been added to the 'asterisk' resource, 'config/dynamic'.
408 This resource allows for push configuration of sorcery derived objects
409 within Asterisk. The resource supports creation, retrieval, updating, and
410 deletion. Sorcery derived objects that are manipulated by this resource
411 must have a sorcery wizard that supports the desired operations.
413 * A new feature has been added that allows for the rotation of log channels
414 through HTTP requests.
419 * A new 'g726_non_standard' endpoint option has been added that, when set to
420 'yes' and g.726 audio is negotiated, forces the codec to be treated as if it
421 is AAL2 packed on the channel.
423 * A new 'rtp_keepalive' endpoint option has been added. This option specifies
424 an interval, in seconds, at which we will send RTP comfort noise packets to
425 the endpoint. This functions identically to chan_sip's "rtpkeepalive" option.
427 * New 'rtp_timeout' and 'rtp_timeout_hold' endpoint options have been added.
428 These options specify the amount of time, in seconds, that Asterisk will wait
429 before terminating the call due to lack of received RTP. These are identical
430 to chan_sip's rtptimeout and rtpholdtimeout options.
432 ------------------------------------------------------------------------------
433 --- Functionality changes from Asterisk 13.3.0 to Asterisk 13.4.0 ------------
434 ------------------------------------------------------------------------------
438 * New 'rpid_immediate' option to control if connected line update information
439 goes to the caller immediately or waits for another reason to send the
440 connected line information update. See the online option documentation for
441 more information. Defaults to 'no' as setting it to 'yes' can result in
442 many unnecessary messages being sent to the caller.
444 * The configuration setting 'progressinband' now defaults to 'no', which
445 matches the actual behavior of previous versions.
449 * A new CLI command has been added: "pjsip show settings", which shows
450 both the global and system configuration settings.
452 * A new aor option has been added: "qualify_timeout", which sets the timeout
453 in seconds for a qualify. The default is 3 seconds. This overrides the
454 hard coded 32 seconds in pjproject.
456 * Endpoint status will now change to "Unreachable" when all contacts are
457 unavailable. When any contact becomes available, the endpoint will status
458 will change back to "Reachable".
460 * A new global option has been added: "max_initial_qualify_time", which
461 sets the maximum amount of time from startup that qualifies should be
462 attempted on all contacts.
466 * Two new events, 'ChannelHold' and 'ChannelUnhold', have been added to the
467 events data model. These events are raised when a channel indicates a hold
468 or unhold, respectively.
472 * A new dialplan function, HOLD_INTERCEPT, has been added. This function, when
473 placed on a channel, intercepts hold/unhold indications signalled by the
474 channel and prevents them from moving on to other channels in a bridge with
475 the hold initiator. Instead, AMI or ARI events are raised indicating that
476 the channel wanted to place someone on hold. This allows external
477 applications to implement their own custom hold/unhold logic.
479 ------------------------------------------------------------------------------
480 --- Functionality changes from Asterisk 13.2.0 to Asterisk 13.3.0 ------------
481 ------------------------------------------------------------------------------
483 chan_pjsip/app_transfer
485 * The Transfer application, when used with chan_pjsip, now supports using
486 a PJSIP endpoint as the transfer destination. This is in addition to
487 explicitly specifying a SIP URI to transfer to.
491 * The ARI /channels resource now supports a new operation, 'redirect'. The
492 redirect operation will perform a technology and state specific redirection
493 on the channel to a specified endpoint or destination. In the case of SIP
494 technologies, this is either a 302 Redirect response to an on-going INVITE
495 dialog or a SIP REFER request.
499 * A new 'endpoint_identifier_order' option has been added that allows one to
500 set the order by which endpoint identifiers are processed and checked. This
501 option is specified under the 'global' type configuration section.
503 ------------------------------------------------------------------------------
504 --- Functionality changes from Asterisk 13.1.0 to Asterisk 13.2.0 ------------
505 ------------------------------------------------------------------------------
507 * New 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions have been added which
508 allow examining PJSIP AORs or contacts from the dialplan.
510 res_pjsip_outbound_registration
512 * The 'pjsip send unregister' command now stops further registrations.
514 * A new command 'pjsip send register' has been added which allows you to
515 start or restart periodic registration. It can be used after a
516 'send unregister' or after a 401 permanent error.
518 res_pjsip_config_wizard
520 * This is a new module that adds streamlined configuration capability for
521 chan_pjsip. It's targeted at users who have lots of basic configuration
522 scenarios like 'phone' or 'agent' or 'trunk'. Additional information
523 can be found in the sample configuration file at
524 config/samples/pjsip_wizard.conf.sample.
528 * The T.38 negotiation timeout was previously hard coded at 5000 milliseconds
529 and is now configurable via the 't38timeout' configuration option in
530 res_fax.conf and via the fax options dialplan function 'FAXOPT(t38timeout)'.
531 The default remains at 5000 milliseconds.
535 * The ca_list_path transport parameter has been added for TLS transports. This
536 option behaves similarly to the old sip.conf option "tlscapath". In order to
537 use this, you must be using PJProject version 2.4 or higher.
541 * The Originate operation now takes in an originator channel. The linked ID of
542 this originator channel is applied to the newly originated outgoing channel.
543 If using CEL this allows an association to be established between the two so
544 it can be recognized that the originator is dialing the originated channel.
546 * "language" (the default spoken language for the channel) is now included in
547 the standard channel state output for suitable events.
549 * The POST channels/{id} operation and the POST channels/{id}/continue operation
550 now have a new "label" parameter. This allows for origination or continuation
551 to a labeled priority in the dialplan instead of requiring a specific priority
552 number. The ARI version has been bumped to 1.7.0 as a result.
556 * "Language" (the default spoken language for the channel) is now included in
557 the standard channel state output for suitable events.
559 * AMI actions that return a list of events have been made to return consistent
560 headers for the action response event starting the list and the list complete
561 event. The AMI version has been bumped to 2.7.0 as a result.
563 ------------------------------------------------------------------------------
564 --- Functionality changes from Asterisk 13.0.0 to Asterisk 13.1.0 ------------
565 ------------------------------------------------------------------------------
569 * Event NewConnectedLine is emitted when the connected line information on
574 * Event ChannelConnectedLine is emitted when the connected line information
575 on a channel changes.
580 The features.conf general section has three new configurable options:
581 * transferdialattempts
583 * transferinvalidsound
584 For more information on what these options do, see the Asterisk wiki:
585 https://wiki.asterisk.org/wiki/x/W4fAAQ
592 * New 'media_encryption_optimistic' endpoint setting. This will use SRTP
593 when possible but does not consider lack of it a failure.
595 res_pjsip_endpoint_identifer_ip
597 * New CLI commands have been added: "pjsip show identif(y|ies)", which lists
598 all configured PJSIP identify objects
600 ------------------------------------------------------------------------------
601 --- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
602 ------------------------------------------------------------------------------
607 Asterisk 13 is the next Long Term Support (LTS) release of Asterisk. As such,
608 the focus of development for this release of Asterisk was on improving the
609 usability and features developed in the previous Standard release, Asterisk 12.
610 Beyond a general refinement of end user features, development focussed heavily
611 on the Asterisk APIs - the Asterisk Manager Interface (AMI) and the Asterisk
612 REST Interface (ARI) - and the PJSIP stack in Asterisk. Some highlights of the
613 new features include:
615 * Asterisk security events are now provided via AMI, allowing end users to
616 monitor their Asterisk system in real time for security related issues.
617 * External control of Message Waiting Indicators (MWI) through both AMI and ARI.
618 * Reception/transmission of out of call text messages using any supported
619 channel driver/protocol stack through ARI.
620 * Resource List Server support in the PJSIP stack, providing subscriptions to
621 lists of resources and batched delivery of NOTIFY requests.
622 * Inter-Asterisk distributed device state and mailbox state using the PJSIP
625 It is important to note that Asterisk 13 is built on the architecture developed
626 during the previous Standard release, Asterisk 12. Users upgrading to
627 Asterisk 13 should read about the new features in Asterisk 12 later in this file
628 (see Functionality changes from Asterisk 11 to Asterisk 12), as well as the
629 UPGRADE-12.txt delivered with this release. In particular, users upgrading to
630 Asterisk 13 from a release prior to Asterisk 12 should read the specifications
631 on AMI, CDRs, and CEL on the Asterisk wiki:
632 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
633 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
634 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
636 Many new featuers in Asterisk 13 were introduced in point releases of
637 Asterisk 12. Following this section - which documents the changes from all
638 versions of Asterisk 12 to Asterisk 13 - users should examine the new features
639 that were introduced in the point releases of Asterisk 12, as they are also
640 included in Asterisk 13.
642 Finally, all users upgrading to Asterisk 13 should read the UPGRADE.txt file
643 delivered with this release.
648 * Sample config files have been moved from configs/ to a sub-folder of that
651 * The menuselect utility has been pulled into the Asterisk repository. As a
652 result, the libxml2 development library is now a required dependency for
655 * A new Compiler Flag, REF_DEBUG, has been added. When enabled, reference
656 counted objects will emit additional debug information to the refs log file
657 located in the standard Asterisk log file directory. This log file is useful
658 in tracking down object leaks and other reference counting issues. Prior to
659 this version, this option was only available by modifying the source code
660 directly. This change also includes a new script, refcounter.py, in the
661 contrib folder that will process the refs log file. Note that this replaces
662 the refcounter utility that could be built from the utils directory.
670 * This module was deprecated and has been removed. Users of app_dahdibarge
671 should use ChanSpy instead.
675 * New options to play a beep when starting a recording and stopping a recording
676 have been added. The option "p" will play a beep to the channel that starts
677 the recording. The option "P" will play a beep to the channel that stops the
682 * Queue rules can now be stored in a database table, queue_rules. Unlike other
683 RealTime tables, the queue_rules table is only examined on module load or
684 module reload. A new general setting has been added to queuerules.conf,
685 'realtime_rules', which, when set to 'yes', will cause app_queue to look in
686 RealTime for additional queue rules to parse. Note that both the file and
687 the database can be used as a provide of queue rules when 'realtime_rules'
690 When app_queue is reloaded, all rules are re-parsed and loaded into memory.
691 There is no caching of RealTime queue rules.
695 * This module was deprecated and has been removed. Users of app_readfile
696 should use func_env's FILE function instead.
700 * The 'say' family of dialplan applications now support the Japanese
701 language. The 'language' parameter in say.conf now recognizes a setting of
702 'ja', which will enable Japanese language specific mechanisms for playing
703 back numbers, dates, and other items.
707 * This module was deprecated and has been removed. Users of app_saycountpl
708 should use the Say family of applications.
712 * The SetMusicOnHold dialplan application was deprecated and has been removed.
713 Users of the application should use the CHANNEL function's musicclass
718 * The WaitMusicOnHold dialplan application was deprecated and has been
719 removed. Users of the application should use MusicOnHold with a duration
724 * VoiceMail and VoiceMailMain now support the Japanese language. The
725 'language' parameter in voicemail.conf now recognizes a setting of 'ja',
726 which will enable prompts to be played back using a Japanese grammatical
727 structure. Additional prompts are necessary for this functionality,
729 - jb-arimasu: there is
730 - jb-arimasen: there is not
731 - jb-oshitekudasai: please press
737 * Add the ability to specify multiple email addresses in configuration,
746 * This module was deprecated and has been removed. Users of cdr_sqlite
747 should use cdr_sqlite3_custom.
751 * Added the ability to support PostgreSQL application_name on connections.
752 This allows PostgreSQL to display the configured name in the
753 pg_stat_activity view and CSV log entries. This setting is configurable
754 for cdr_pgsql via the appname configuration setting in cdr_pgsql.conf.
762 * Added the ability to support PostgreSQL application_name on connections.
763 This allows PostgreSQL to display the configured name in the
764 pg_stat_activity view and CSV log entries. This setting is configurable
765 for cel_pgsql via the appname configuration setting in cel_pgsql.conf.
773 * SS7 support now requires libss7 v2.0 or later.
775 * Added SS7 support for connected line and redirecting.
777 * Most SS7 CLI commands are reworked as well as new SS7 commands added.
780 * Added several SS7 config option parameters described in
781 chan_dahdi.conf.sample.
785 * This module was deprecated and has been removed. Users of chan_gtalk
786 should use chan_motif.
790 * This module was deprecated and has been removed. Users of chan_h323
791 should use chan_ooh323.
795 * This module was deprecated and has been removed. Users of chan_jingle
796 should use chan_motif.
800 * Added the CLI command 'pjsip list ciphers' so a user can know what
801 OpenSSL names are available on their system for the pjsip.conf cipher
806 * The SIPPEER dialplan function no longer supports using a colon as a
807 delimiter for parameters. The parameters for the function should be
808 delimited using a comma.
810 * The SIPCHANINFO dialplan function was deprecated and has been removed. Users
811 of the function should use the CHANNEL function instead.
819 * Added functional peeraccount support. Except for Queue, the
820 accountcode propagation is now consistently propagated to outgoing
821 channels before dialing. The channel accountcode can change from its
822 original non-empty value on channel creation for the following specific
823 reasons. One, dialplan sets it using CHANNEL(accountcode). Two, an
824 originate method that can specify an accountcode value. Three, the
825 calling channel propagates its peeraccount or accountcode to the
826 outgoing channel's accountcode before dialing. The change has two
827 visible effects. One, local channels now cross accountcode and
828 peeraccount across the special bridge between the ;1 and ;2 channels
829 just like channels between normal bridges. Two, the
830 CHANNEL(peeraccount) value can now be set before Dial and FollowMe to
831 set the accountcode on the outgoing channel(s).
833 For Queue, an outgoing channel's non-empty accountcode will not change
834 unless explicitly set by CHANNEL(accountcode). The change has three
835 visible effects. One, local channels now cross accountcode and
836 peeraccount across the special bridge between the ;1 and ;2 channels
837 just like channels between normal bridges. Two, the queue member will
838 get an accountcode if it doesn't have one and one is available from the
839 calling channel's peeraccount. Three, accountcode propagation includes
840 local channel members where the accountcodes are propagated early
841 enough to be available on the ;2 channel.
845 * New DeviceStateChanged and PresenceStateChanged AMI events have been added.
846 These events are emitted whenever a device state or presence state change
847 occurs. The events are controlled by res_manager_device_state.so and
848 res_manager_presence_state.so. If the high frequency of these events is
849 problematic for you, do not load these modules.
851 * Added DialplanExtensionAdd and DialplanExtensionRemove AMI commands. They
852 work in basically the same way as the 'dialplan add extension' and
853 'dialplan remove extension' CLI commands respectively.
855 * New AMI action LoggerRotate reloads and rotates logger in the same manner
856 as CLI command 'logger rotate'
858 * New AMI Actions FAXSessions, FAXSession, and FAXStats replicate the
859 functionality of CLI commands 'fax show sessions', 'fax show session',
860 and fax show stats' respectively.
862 * New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset
863 enable manager control over PRI debugging levels and file output.
865 * AMI action PJSIPNotify may now send to a URI instead of only to a PJSIP
866 endpoint as long as a default outbound endpoint is set. This also applies
867 to the equivalent CLI command (pjsip send notify)
869 * The AMI action PJSIPShowEndpoint now includes ContactStatusDetail sections
870 that give information on Asterisk's attempts to qualify the endpoint.
872 * The DialEnd event will now contain a Forward header if the dial is ending
873 due to the call being forwarded. The contents of the Forward header is the
874 extension in the number to which the call is being forwarded.
878 * The "bridge_technology" extra field key has been added to BRIDGE_ENTER
879 and BRIDGE_EXIT events.
883 * Channel variables are now substituted in arguments passed to applications
884 run by using dynamic features.
888 * The TLS core in Asterisk now supports Perfect Forward Secrecy (PFS).
889 Enabling PFS is attempted by default, and is dependent on the configuration
890 of the module using TLS.
891 - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
892 specify a ECDHE cipher suite in sip.conf, for example:
893 tlscipher=AES128-SHA:DES-CBC3-SHA
894 - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
895 into the private key file, e.g., sip.conf tlsprivatekey. For example, the
896 default dh2048.pem - see
897 http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt
898 - Because clients expect the server to prefer PFS, and because OpenSSL sorts
899 its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
900 Consider re-ordering your cipher suites in the respective configuration
902 tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
903 will use PFS when offered by the client. Clients which do not offer PFS
904 fall-back to AES-128 (or even 3DES, as recommended by RFC 3261).
912 * The JACK_HOOK function now supports audio with a sample rate higher than
921 * Added the ability to support PostgreSQL application_name on connections.
922 This allows PostgreSQL to display the configured name in the
923 pg_stat_activity view and CSV log entries. This setting is configurable
924 for res_config_pgsql via the dbappname configuration setting in
927 res_pjsip_outbound_publish
929 * A new module, res_pjsip_outbound_publish provides the mechanisms for sending
930 PUBLISH requests for specific event packages to another SIP User Agent.
934 * The publish/subscribe core module has been updated to support RFC 4662
935 Resource Lists, allowing Asterisk to act as a Resource List Server (RLS).
936 Resource lists are configured in pjsip.conf under a new object type,
937 resource_list. Resource lists can contain either message-summary or presence
938 events, and can be composed of specific resources that provide the event or
939 other resource lists.
941 * Inbound publication support is provided by a new object, inbound-publication.
942 This configures res_pjsip_pubsub to accept PUBLISH requests from a particular
943 resource. Which events are accepted is constructed dynamically; see
944 res_pjsip_publish_asterisk for more information.
946 res_pjsip_publish_asterisk
948 * A new module, res_pjsip_publish_asterisk adds support for PUBLISH requests of
949 Asterisk information to other Asterisk servers. This module is intended only
950 for Asterisk to Asterisk exchanges of information. Currently, this includes
951 both mailbox state and device state information.
953 ------------------------------------------------------------------------------
954 --- Functionality changes from Asterisk 12.4.0 to Asterisk 12.5.0 ------------
955 ------------------------------------------------------------------------------
959 * Stored recordings now support a new operation, copy. This will take an
960 existing stored recording and copy it to a new location in the recordings
963 * LiveRecording objects now have three additional fields that can be reported
964 in a RecordingFinished ARI event:
965 - total_duration: the duration of the recording
966 - talking_duration: optional. The duration of talking detected in the
967 recording. This is only available if max_silence_seconds was specified
968 when the recording was started.
969 - silence_duration: optional. The duration of silence detected in the
970 recording. This is only available if max_silence_seconds was specified
971 when the recording was started.
972 Note that all duration values are reported in seconds.
974 * Users of ARI can now send and receive out of call text messages. Messages
975 can be sent directly to a particular endpoint, or can be sent to the
976 endpoints resource directly and inferred from the URI scheme. Text
977 messages are passed to ARI clients as TextMessageReceived events. ARI
978 clients can choose to receive text messages by subscribing to the particular
979 endpoint technology or endpoints that they are interested in.
981 * The applications resource now supports subscriptions to all endpoints of
982 a particular channel technology. For example, subscribing to an eventSource
983 of 'endpoint:PJSIP' will subscribe to all PJSIP endpoints.
987 * The endpoint configuration object now supports 'accountcode'. Any channel
988 created for an endpoint with this setting will have its accountcode set
989 to the specified value.
993 * A new module, res_hep_rtcp, has been added that will forward RTCP call
994 statistics to a HEP capture server. See res_hep for more information.
998 * Function AUDIOHOOK_INHERIT has been deprecated. Audiohooks are now
999 unconditionally inhereted through masquerades. As a side benefit, more
1000 than one audiohook of a given type may persist through a masquerade now.
1002 ------------------------------------------------------------------------------
1003 --- Functionality changes from Asterisk 12.3.0 to Asterisk 12.4.0 ------------
1004 ------------------------------------------------------------------------------
1008 * Returns new AGENT_STATUS value "NOT_CONNECTED" if the agent fails to
1009 connect with an incoming caller after being alerted to the presence
1010 of the incoming caller. The most likely reason this would happen is
1011 the agent did not acknowledge the call in time.
1015 * New events have been added for the TALK_DETECT function. When the function
1016 is used on a channel, ChannelTalkingStart/ChannelTalkingStop events will be
1017 emitted to connected AMI clients indicating the start/stop of talking on
1022 * New event models have been aded for the TALK_DETECT function. When the
1023 function is used on a channel, ChannelTalkingStarted/ChannelTalkingFinished
1024 events will be emitted to connected WebSockets subscribed to the channel,
1025 indicating the start/stop of talking on the channel.
1029 * A new function, TALK_DETECT, has been added. When set on a channel, this
1030 fucntion causes events indicating the starting/stoping of talking on said
1031 channel to be emitted to both AMI and ARI clients.
1033 ------------------------------------------------------------------------------
1034 --- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------
1035 ------------------------------------------------------------------------------
1039 * A new Playback URI 'tone' has been added. Tones are specified either as
1040 an indication name (e.g. 'tone:busy') from indications.conf or as a tone
1041 pattern (e.g. 'tone:240/250,0/250'). Tones differ from normal playback
1042 URIs in that they must be stopped manually and will continue to occupy
1043 a channel's ARI control queue until they are stopped. They also can not
1044 be rewound or fastforwarded.
1046 * User events can now be generated from ARI. Events can be signalled with
1047 arbitrary json variables, and include one or more of channel, bridge, or
1048 endpoint snapshots. An application must be specified which will receive
1049 the event message (other applications can subscribe to it). The message
1050 will also be delivered via AMI provided a channel is attached. Dialplan
1051 generated user event messages are still transmitted via the channel, and
1052 will only be received by a stasis application they are attached to or if
1053 the channel is subscribed to.
1057 * SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI
1058 fields for prohibited callingpres information. Values are legacy, no, and
1059 yes. By default, legacy is used.
1060 trust_id_outbound=legacy - behavior remains the same as 1.8.26.1. When
1061 dealing with prohibited callingpres and sendrpid=pai/rpid, RPID/PAI
1062 headers are appended to outbound SIP messages just as they are with
1063 allowed callingpres values, but data about the remote party's identity is
1065 When sendrpid=rpid, only the remote party's domain is anonymized.
1066 trust_id_outbound=no - when dealing with prohibited callingpres, RPID/PAI
1067 headers are not sent.
1068 trust_id_outbound=yes - RPID/PAI headers are applied with the full remote
1069 party information in tact even for prohibited callingpres information.
1070 In the case of PAI, a Privacy: id header will be appended for prohibited
1071 calling information to communicate that the private information should
1072 not be relayed to untrusted parties.
1076 * Manager action 'Park' now takes an additional argument 'AnnounceChannel'
1077 which can be used to announce the parked call's location to an arbitrary
1078 channel in a bridge. If 'Channel' and 'TimeoutChannel' are now the two
1079 parties in a one to one bridge, 'TimeoutChannel' is treated as having
1080 parked 'Channel' like with the Park Call DTMF feature and will receive
1081 announcements prior to being hung up.
1083 ------------------------------------------------------------------------------
1084 --- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
1085 ------------------------------------------------------------------------------
1089 * Record application now has an option 'o' which allows 0 to act as an exit
1090 key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'
1093 --------------------------
1094 * ChanSpy now accepts a channel uniqueid or a fully specified channel name
1095 as the chanprefix parameter if the 'u' option is specified.
1098 --------------------------
1099 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
1100 conference user menus.
1102 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
1103 menus, bridge settings, and user settings that have been applied by the
1104 CONFBRIDGE dialplan function.
1106 * The ConfBridge dialplan application now sets a channel variable,
1107 CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
1108 how a channel exited the conference.
1110 * Added conference user option 'announce_join_leave_review'. This option
1111 implies 'announce_join_leave' with the added effect that the user will
1112 be asked if they want to confirm or re-record the recording of their
1113 name when entering the conference
1116 --------------------------
1117 * At exit, the Directory application now sets a channel variable
1118 DIRECTORY_RESULT to one of the following based on the reason for exiting:
1119 OPERATOR user requested operator by pressing '0' for operator
1120 ASSISTANT user requested assistant by pressing '*' for assistant
1121 TIMEOUT user pressed nothing and Directory stopped waiting
1122 HANGUP user's channel hung up
1123 SELECTED user selected a user from the directory and is routed
1124 USEREXIT user pressed '#' from the selection prompt to exit
1125 FAILED directory failed in a way that wasn't accounted for. Dang.
1129 * Monitor() - A new option, B(), has been added that will turn on a periodic
1130 beep while the call is being recorded.
1133 --------------------------
1134 * MusicOnHold streams (all modes other than "files") now support wide band
1138 --------------------------
1139 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
1140 and for the channel executing Page respectively.
1143 --------------------------
1144 * PickupChan now accepts channel uniqueids of channels to pickup.
1147 --------------------------
1148 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
1149 to 'true' (case insensitive), then any Say application (SayNumber,
1150 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
1151 anticipate DTMF. If DTMF is received, these applications will behave like
1152 the background application and jump to the received extension once a match
1153 is established or after a short period of inactivity.
1156 -------------------------
1157 * A new function, MIXMONITOR, has been added to allow access to individual
1158 instances of MixMonitor on a channel.
1160 * A new option, B(), has been added that will turn on a periodic beep while the
1161 call is being recorded.
1165 -------------------------
1168 -------------------------
1169 * TEL URI support for inbound INVITE requests has been added. chan_sip will
1170 now handle TEL schemes in the Request and From URIs. The phone-context in
1171 the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on
1172 the inbound channel.
1176 * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
1177 the new AST_SORCERY diaplan function.
1179 * Core Show Locks output now includes Thread/LWP ID if the platform
1180 supports this feature.
1182 * New "logger add channel" and "logger remove channel" CLI commands have
1183 been added to allow creation and deletion of dynamic logger channels
1184 without configuration changes. These dynamic logger channels will only
1185 exist until the next restart of asterisk.
1189 * The live recording object on recording events now contains a target_uri
1190 field which contains the URI of what is being recorded.
1192 * The bridge type used when creating a bridge is now a comma separated list of
1193 bridge properties. Valid options are: mixing, holding, dtmf_events, and
1196 * A channelId can now be provided when creating a channel, either in the
1197 uri (POST channels/my-channel-id) or as query parameter. A local channel
1198 will suffix the second channel id with ';2' unless provided as query
1199 parameter otherChannelId.
1201 * A bridgeId can now be provided when creating a bridge, either in the uri
1202 (POST bridges/my-bridge-id) or as a query parameter.
1204 * A playbackId can be provided when starting a playback, either in the uri
1205 (POST channels/my-channel-id/play/my-playback-id /
1206 POST bridges/my-bridge-id/play/my-playback-id) or as a query parameter.
1208 * A snoop channel can be started with a snoopId, in the uri or query.
1212 * Originate now takes optional parameters ChannelId and OtherChannelId,
1213 used to set the UniqueId on creation. The other id is assigned to the
1214 second channel when dialing LOCAL, or defaults to appending ;2 if only
1215 the single Id is given.
1217 * The Mixmonitor action now has a "Command" header that can be used to
1218 indicate a post-process command to run once recording finishes.
1222 * A new set of Alembic scripts has been added for CDR tables. This will create
1223 a 'cdr' table with the default schema that Asterisk expects.
1228 * A new function was added: PERIODIC_HOOK. This allows running a periodic
1229 dialplan hook on a channel. Any audio generated by this hook will be
1230 injected into the call.
1238 * A new module, res_hep, has been added, that acts as a generic packet
1239 capture agent for the Homer Encapsulation Protocol (HEP) version 3.
1240 It can be configured via hep.conf. Other modules can use res_hep to send
1241 message traffic to a HEP capture server.
1245 * A new module, res_hep_pjsip, has been added that will forward PJSIP
1246 message traffic to a HEP capture server. See res_hep for more
1251 * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
1252 be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
1254 * Added the following new CLI commands:
1255 - "pjsip show contacts" - list all current PJSIP contacts.
1256 - "pjsip show contact" - show specific information about a current PJSIP
1258 - "pjsip show channel" - show detailed information about a PJSIP channel.
1260 res_pjsip_multihomed
1262 * A new module, res_pjsip_multihomed handles situations where the system
1263 Asterisk is running out has multiple interfaces. res_pjsip_multihomed
1264 determines which interface should be used during message sending.
1266 res_pjsip_pidf_digium_body_supplement
1268 * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
1269 request body formatting for presence support in Digium phones.
1271 res_pjsip_send_to_voicemail
1273 * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
1274 particular headers to transfer a PJSIP channel directly to a particular
1275 extension that has VoiceMail. This is intended to be used with Digium
1276 phones that support this feature.
1278 res_pjsip_outbound_registration
1280 * A new CLI command has been added: "pjsip show registrations", which lists
1281 all configured PJSIP registrations
1284 ------------------------------------------------------------------------------
1285 --- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
1286 ------------------------------------------------------------------------------
1290 * Added a new module that provides AMI control over MWI within Asterisk,
1291 res_mwi_external_ami. Note that this module depends on res_mwi_external;
1292 for more information on enabling this module, see res_mwi_external.
1293 This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
1294 the MWIGet/MWIGetComplete events.
1296 * The DialStatus field in the DialEnd event can now contain additional
1297 statuses that convey how the dial operation terminated. This includes
1298 ABORT, CONTINUE, and GOTO.
1300 * AMI will now emit security events. A new class authorization has been
1301 added in manager.conf for the security events, 'security'. The new events
1303 - FailedACL - raised when a request violates an ACL check
1304 - InvalidAccountID - raised when a request fails an authentication
1305 check due to an invalid account ID
1306 - SessionLimit - raised when a request fails due to exceeding the
1307 number of allowed concurrent sessions for a service
1308 - MemoryLimit - raised when a request fails due to an internal memory
1310 - LoadAverageLimit - raised when a request fails because a configured
1311 load average limit has been reached
1312 - RequestNotAllowed - raised when a request is not allowed by
1314 - AuthMethodNotAllowed - raised when a request used an authentication
1315 method not allowed by the service
1316 - RequestBadFormat - raised when a request is received with bad formatting
1317 - SuccessfulAuth - raised when a request successfully authenticates
1318 - UnexpectedAddress - raised when a request has a different source address
1319 then what is expected for a session already in progress with a service
1320 - ChallengeResponseFailed - raised when a request's attempt to authenticate
1321 has been challenged, and the request failed the authentication challenge
1322 - InvalidPassword - raised when a request provides an invalid password
1323 during an authentication attempt
1324 - ChallengeSent - raised when an Asterisk service send an authentication
1325 challenge to a request
1326 - InvalidTransport - raised when a request attempts to use a transport not
1327 allowed by the Asterisk service
1329 * Bridge related events now have two additional fields: BridgeName and
1330 BridgeCreator. BridgeName is a descriptive name for the bridge;
1331 BridgeCreator is the name of the entity that created the bridge. This
1332 affects the following events: ConfbridgeStart, ConfbridgeEnd,
1333 ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
1334 ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
1335 AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
1339 * The Bridge data model now contains the additional fields 'name' and
1340 'creator'. The 'name' field conveys a descriptive name for the bridge;
1341 the 'creator' field conveys the name of the entity that created the bridge.
1342 This affects all responses to HTTP requests that return a Bridge data model
1343 as well as all event derived data models that contain a Bridge data model.
1344 The POST /bridges operation may now optionally specify a name to give to
1345 the bridge being created.
1347 * Added a new ARI resource 'mailboxes' which allows the creation and
1348 modification of mailboxes managed by external MWI. Modules res_mwi_external
1349 and res_stasis_mailbox must be enabled to use this resource. For more
1350 information on external MWI control, see res_mwi_external.
1352 * Added new events for externally initiated transfers. The event
1353 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
1354 of a bridge in the ARI controlled application to the dialplan; the
1355 BridgeAttendedTransfer event is raised when a channel initiates an
1356 attended transfer of a bridge in the ARI controlled application to the
1359 * Channel variables may now be specified as a body parameter to the
1360 POST /channels operation. The 'variables' key in the JSON is interpreted
1361 as a sequence of key/value pairs that will be added to the created channel
1362 as channel variables. Other parameters in the JSON body are treated as
1363 query parameters of the same name.
1367 * Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be
1368 automatically handled by the HTTP server if a request is received with a
1369 Transfer-Encoding type of "chunked".
1373 * Path support has been added with the 'support_path' option in registration
1376 * A 'debug' option has been added to the globals section that will allow
1377 sip messages to be logged.
1379 * A 'set_var' option has been added to endpoints that will automatically
1380 set the desired variable(s) on a channel created for that endpoint.
1382 * Several new tables and columns have been added to the realtime schema for
1383 the res_pjsip related modules. See the UPGRADE.txt notes for updating
1384 the database schema.
1388 * A new module, res_mwi_external, has been added to Asterisk. This module
1389 acts as a base framework that other modules can build on top of to allow
1390 an external system to control MWI within Asterisk. For implementations
1391 that make use of res_mwi_external, see res_mwi_external_ami and
1392 res_ari_mailboxes. Note that res_mwi_external canflicts with other modules
1393 that may produce MWI themselves, such as app_voicemail. res_mwi_external
1394 and other modules that depend on it cannot be built or loaded with
1395 app_voicemail present.
1399 * DNS functionality will now automatically be enabled if the system configured
1400 nameservers can be retrieved. If the system configured nameservers can not be
1401 retrieved the functionality will resort to using system resolution. Functionalty
1402 such as SRV records and failover will not be available if system resolution
1405 ------------------------------------------------------------------------------
1406 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
1407 ------------------------------------------------------------------------------
1412 Asterisk 12 is a standard release of the Asterisk project. As such, the
1413 focus of development for this release was on core architectural changes and
1414 major new features. This includes:
1415 * A more flexible bridging core based on the Bridging API
1416 * A new internal message bus, Stasis
1417 * Major standardization and consistency improvements to AMI
1418 * Addition of the Asterisk RESTful Interface (ARI)
1419 * A new SIP channel driver, chan_pjsip
1420 In addition, as the vast majority of bridging in Asterisk was migrated to the
1421 Bridging API used by ConfBridge, major changes were made to most of the
1422 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
1424 Specifications have been written for the affected interfaces. These
1425 specifications are available on the Asterisk wiki:
1426 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
1427 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
1428 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
1430 It is *highly* recommended that anyone migrating to Asterisk 12 read the
1431 information regarding its release both in this file and in the accompanying
1432 UPGRADE.txt file. More detailed information on the major changes can be found
1433 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
1438 * Added build option DISABLE_INLINE. This option can be used to work around a
1439 bug in gcc. For more information, see
1440 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
1442 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
1443 the CHANNEL_TRACE build option were incompatible with the new bridging
1446 * Asterisk now optionally uses libxslt to improve XML documentation generation
1447 and maintainability. If libxslt is not available on the system, some XML
1448 documentation will be incomplete.
1450 * Asterisk now depends on libjansson. If a package of libjansson is not
1451 available on your distro, please see http://www.digip.org/jansson/.
1453 * Asterisk now depends on libuuid and, optionally, uriparser. It is
1454 recommended that you install uriparser, even if it is optional.
1456 * The new SIP stack and channel driver uses a particular version of PJSIP.
1457 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
1458 configuring and installing PJSIP for usage with Asterisk.
1460 * Optional API was re-implemented to be more portable, and no longer requires
1461 weak reference support from the compiler. The build option OPTIONAL_API may
1462 be disabled to disable Optional API support.
1469 * Along with AgentRequest, this application has been modified to be a
1470 replacement for chan_agent. The act of a channel calling the AgentLogin
1471 application places the channel into a pool of agents that can be
1472 requested by the AgentRequest application. Note that this application, as
1473 well as all other agent related functionality, is now provided by the
1474 app_agent_pool module. See chan_agent and AgentRequest for more information.
1476 * This application no longer performs agent authentication. If authentication
1477 is desired, the dialplan needs to perform this function using the
1478 Authenticate or VMAuthenticate application or through an AGI script before
1481 * If this application is called and the agent is already logged in, the
1482 dialplan will continue exection with the AGENT_STATUS channel variable set
1483 to ALREADY_LOGGED_IN.
1485 * The agents.conf schema has changed. Rather than specifying agents on a
1486 single line in comma delineated fashion, each agent is defined in a separate
1487 context. This allows agents to use the power of context templates in their
1490 * A number of parameters from agents.conf have been removed. This includes
1491 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
1492 urlprefix, and savecallsin. These options were obsoleted by the move from
1493 a channel driver model to the bridging/application model provided by
1498 * A new application, this will request a logged in agent from the pool and
1499 bridge the requested channel with the channel calling this application.
1500 Logged in agents are those channels that called the AgentLogin application.
1501 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
1502 application will be set with an appropriate error value.
1504 AgentMonitorOutgoing
1506 * This application has been removed. It was a holdover from when
1507 AgentCallbackLogin was removed.
1511 * Added support for additional Ademco DTMF signalling formats, including
1512 Express 4+1, Express 4+2, High Speed and Super Fast.
1514 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
1515 call time, in milliseconds, to run the application.
1517 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
1518 maximum number of times to retry the call.
1520 * Added a new configuration option answait. If set, the AlarmReceiver
1521 application will wait the number of milliseconds specified by answait
1522 after the channel has answered. Valid values range between 500
1523 milliseconds and 10000 milliseconds.
1525 * Added configuration option no_group_meta. If enabled, grouping of metadata
1526 information in the AlarmReceiver log file will be skipped.
1530 * It is now no longer possible to bypass updating the CDR on the channel
1531 when answering. CDRs reflect the state of the channel and will always
1532 reflect the time they were Answered.
1536 * A new application in Asterisk, this will place the calling channel
1537 into a holding bridge, optionally entertaining them with some form of
1538 media. Channels participating in a holding bridge do not interact with
1539 other channels in the same holding bridge. Optionally, however, a channel
1540 may join as an announcer. Any media passed from an announcer channel is
1541 played to all channels in the holding bridge. Channels leave a holding
1542 bridge either when an optional timer expires, or via the ChannelRedirect
1543 application or AMI Redirect action.
1547 * All participants in a bridge can now be kicked out of a conference room
1548 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
1549 command, i.e., 'confbridge kick <conference> all'
1551 * CLI output for the 'confbridge list' command has been improved. When
1552 displaying information about a particular bridge, flags will now be shown
1553 for the participating users indicating properties of that user.
1555 * The ConfbridgeList event now contains the following fields: WaitMarked,
1556 EndMarked, and Waiting. This displays additional properties about the
1557 user's profile, as well as whether or not the user is waiting for a
1558 Marked user to enter the conference.
1560 * Added a new option for conference recording, record_file_append. If enabled,
1561 when the recording is stopped and then re-started, the existing recording
1562 will be used and appended to.
1564 * ConfBridge now has the ability to set the language of announcements to the
1565 conference. The language can be set on a bridge profile in confbridge.conf
1566 or by the dialplan function CONFBRIDGE(bridge,language)=en.
1570 * The channel variable CPLAYBACKSTATUS may now return the value
1571 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
1572 such as AMI. See the AMI action ControlPlayback for more information.
1576 * Added the 'a' option, which allows the caller to enter in an additional
1577 alias for the user in the directory. This option must be used in conjunction
1578 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
1579 specified in voicemail.conf.
1583 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
1584 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
1585 containing the unique ID of the bridge that the channel happens to be in.
1589 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
1590 for more information.
1592 * Variables are no longer purged from the original CDR. See the 'v' option for
1595 * The 'A' option has been removed. The Answer time on a CDR is never updated
1598 * The 'd' option has been removed. The disposition on a CDR is a function of
1599 the state of the channel and cannot be altered.
1601 * The 'D' option has been removed. Who the Party B is on a CDR is a function
1602 of the state of the respective channels involved in the CDR and cannot be
1605 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
1606 such that the start time and, if applicable, the answer time was updated.
1607 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
1608 'r' option now triggers the Reset, setting the start time (and answer time
1609 if applicable) to the current time. Note that the 'a' option still sets
1610 the answer time to the current time if the channel was already answered.
1612 * The 's' option has been removed. A variable can be set on the original CDR
1613 if desired using the CDR function, and removed from a forked CDR using the
1616 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
1617 longer applies in the CDR engine.
1619 * The 'v' option now prevents the copy of the variables from the original CDR
1620 to the forked CDR. Previously the variables were always copied but were
1621 removed from the original. This was changed as removing variables from a CDR
1622 can have unintended side effects - this option allows the user to prevent
1623 propagation of variables from the original to the forked without modifying
1628 * Added the 'n' option to MeetMe to prevent application of the DENOISE
1629 function to a channel joining a conference. Some channel drivers that vary
1630 the number of audio samples in a voice frame will experience significant
1631 quality problems if a denoiser is attached to the channel; this option gives
1632 them the ability to remove the denoiser without having to unload func_speex.
1636 * The 'b' option now includes conferences as well as sounds played to the
1639 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
1640 running during a transfer. If a MixMonitor is started on a channel,
1641 the MixMonitor will continue to record the audio passing through the
1642 channel even in the presence of transfers.
1646 * The NoCDR application is deprecated. Please use the CDR_PROP function to
1649 * While the NoCDR application will prevent CDRs for a channel from being
1650 propagated to registered CDR backends, it will not prevent that data from
1651 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
1652 function that enables CDRs on a channel will restore those records that have
1653 not yet been finalized.
1657 * The app_parkandannounce module has been removed. The application
1658 ParkAndAnnounce is now provided by the res_parking module. See the
1659 res_parking changes for more information.
1663 * Added queue available hint. The hint can be added to the dialplan using the
1664 following syntax: exten,hint,Queue:{queue_name}_avail
1665 For example, if the name of the queue is 'markq':
1666 exten => 8501,hint,Queue:markq_avail
1667 This will report 'InUse' if there are no logged in agents or no free agents.
1668 It will report 'Idle' when an agent is free.
1670 * Queues now support a hint for member paused state. The hint uses the form
1671 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
1672 are the name of the queue and the name of the member to subscribe to,
1673 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
1674 Members will show as In Use when paused.
1676 * The configuration options eventwhencalled and eventmemberstatus have been
1677 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
1678 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
1679 sent. The "Variable" fields will also no longer exist on the Agent* events.
1680 These events can be filtered out from a connected AMI client using the
1681 eventfilter setting in manager.conf.
1683 * The queue log now differentiates between blind and attended transfers. A
1684 blind transfer will result in a BLINDTRANSFER message with the destination
1685 context and extension. An attended transfer will result in an
1686 ATTENDEDTRANSFER message. This message will indicate the method by which
1687 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
1688 for running an application on a bridge or channel, or "LINK" for linking
1689 two bridges together with local channels. The queue log will also now detect
1690 externally initiated blind and attended transfers and record the transfer
1693 * When performing queue pause/unpause on an interface without specifying an
1694 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
1695 least one member of any queue exists for that interface.
1697 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
1698 for realtime queue log entries.
1702 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
1703 CDRs when they were previously disabled on a channel.
1705 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
1706 backends occurs on an as-needed basis in order to preserve linkedid
1707 propagation and other needed behavior.
1711 * A new application, this is similar to SayAlpha except that it supports
1712 case sensitive playback of the specified characters. For example,
1713 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
1717 * This application is deprecated in favor of CHANNEL(amaflags).
1721 * The SendDTMF application will now accept 'W' as valid input. This will cause
1722 the application to delay one second while streaming DTMF.
1726 * A new application in Asterisk 12, this hands control of the channel calling
1727 the application over to an external system. Currently, external systems
1728 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
1732 * UserEvent will now handle duplicate keys by overwriting the previous value
1733 assigned to the key.
1735 * In addition to AMI, UserEvent invocations will now be distributed to any
1736 interested Stasis applications.
1740 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1741 system as mailbox@context. The rest of the system cannot add @default
1742 to mailbox identifiers for app_voicemail that do not specify a context
1743 any longer. It is a mailbox identifier format that should only be
1744 interpreted by app_voicemail.
1746 * The voicemail.conf configuration file now has an 'alias' configuration
1747 parameter for use with the Directory application. The voicemail realtime
1748 database table schema has also been updated with an 'alias' column.
1753 * Pass through support has been added for both VP8 and Opus.
1755 * Added format attribute negotiation for the Opus codec. Format attribute
1756 negotiation is provided by the res_format_attr_opus module.
1761 * Masquerades as an operation inside Asterisk have been effectively hidden
1762 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
1763 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
1764 dropping of frame/audio hooks, and other internal implementation details
1765 that users had to deal with. This fundamental change has large implications
1766 throughout the changes documented for this version. For more information
1767 about the new core architecture of Asterisk, please see the Asterisk wiki.
1769 * Multiple parties in a bridge may now be transferred. If a participant in a
1770 multi-party bridge initiates a blind transfer, a Local channel will be used
1771 to execute the dialplan location that the transferer sent the parties to. If
1772 a participant in a multi-party bridge initiates an attended transfer,
1773 several options are possible. If the attended transfer results in a transfer
1774 to an application, a Local channel is used. If the attended transfer results
1775 in a transfer to another channel, the resulting channels will be merged into
1778 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
1779 driver specific. If the channel variable is set on the transferrer channel,
1780 the sound will be played to the target of an attended transfer.
1782 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
1783 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
1784 listed. Any more peers in the bridge will not be included in the list.
1785 BRIDGEPEER is not valid in holding bridges like parking since those channels
1786 do not talk to each other even though they are in a bridge.
1788 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
1789 and will contain a value if the BRIDGEPEER's channel driver supports it.
1791 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
1792 was responsible for an attended transfer in a similar fashion to
1795 * Modules using the Configuration Framework or Sorcery must have XML
1796 configuration documentation. This configuration documentation is included
1797 with the rest of Asterisk's XML documentation, and is accessible via CLI
1798 commands. See the CLI changes for more information.
1800 AMI (Asterisk Manager Interface)
1802 * Major changes were made to both the syntax as well as the semantics of the
1803 AMI protocol. In particular, AMI events have been substantially improved
1804 in this version of Asterisk. For more information, please see the AMI
1805 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
1807 * AMI events that reference a particular channel or bridge will now always
1808 contain a standard set of fields. When multiple channels or bridges are
1809 referenced in an event, fields for at least some subset of the channels
1810 and bridges in the event will be prefixed with a descriptive name to avoid
1811 name collisions. See the AMI event documentation on the Asterisk wiki for
1814 * The CLI command 'manager show commands' no longer truncates command names
1815 longer than 15 characters and no longer shows authorization requirement
1816 for commands. 'manager show command' now displays the privileges needed
1817 for using a given manager command instead.
1819 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
1820 peer in its response if the peer has a subscribe context set.
1822 * The SIPqualifypeer action now acknowledges the request once it has
1823 established that the request is against a known peer. It also issues a new
1824 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
1826 * The PlayDTMF action now supports an optional 'Duration' parameter. This
1827 specifies the duration of the digit to be played, in milliseconds.
1829 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
1830 updates when changes occur instead of requiring the use of pollmailboxes.
1832 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
1833 AMI client to manipulate audio currently being played back on a channel. The
1834 supported operations depend on the application being used to send audio to
1835 the channel. When the audio playback was initiated using the ControlPlayback
1836 application or CONTROL STREAM FILE AGI command, the audio can be paused,
1837 stopped, restarted, reversed, or skipped forward. When initiated by other
1838 mechanisms (such as the Playback application), the audio can be stopped,
1839 reversed, or skipped forward.
1841 * Channel related events now contain a snapshot of channel state, adding new
1842 fields to many of these events.
1844 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
1845 in a future release. Please use the common 'Exten' field instead.
1847 * The AMI event 'UserEvent' from app_userevent now contains the channel state
1848 fields. The channel state fields will come before the body fields.
1850 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
1851 'UnParkedCall' have changed significantly in the new res_parking module.
1853 The 'Channel' and 'From' headers are gone. For the channel that was parked
1854 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
1855 has a number of fields associated with it. The old 'Channel' header relayed
1856 the same data as the new 'ParkeeChannel' header.
1858 The 'From' field was ambiguous and changed meaning depending on the event.
1859 for most of these, it was the name of the channel that parked the call
1860 (the 'Parker'). There is no longer a header that provides this channel name,
1861 however the 'ParkerDialString' will contain a dialstring to redial the
1862 device that parked the call.
1864 On UnParkedCall events, the 'From' header would instead represent the
1865 channel responsible for retrieving the parkee. It receives a channel
1866 snapshot labeled 'Retriever'. The 'from' field is is replaced with
1869 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
1871 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
1872 fashion has changed the field names 'StartExten' and 'StopExten' to
1873 'StartSpace' and 'StopSpace' respectively.
1875 * The deprecated use of | (pipe) as a separator in the channelvars setting in
1876 manager.conf has been removed.
1878 * Channel Variables conveyed with a channel no longer contain the name of the
1879 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
1880 ChanVariable: bar=baz. When multiple channels are present in a single AMI
1881 event, the various ChanVariable fields will contain a suffix that specifies
1882 which channel they correspond to.
1884 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
1885 event always conveys the AMI event for a particular channel.
1887 * All 'Reload' events have been consolidated into a single event type. This
1888 event will always contain a Module field specifying the name of the module
1889 and a Status field denoting the result of the reload. All modules now issue
1890 this event when being reloaded.
1892 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
1893 fail to receive this event due to being connected after modules have loaded.
1894 AMI connections that want to know when Asterisk is ready should listen for
1895 the 'FullyBooted' event.
1897 * app_fax now sends the same send fax/receive fax events as res_fax. The
1898 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
1899 now the 'ReceiveFAX' event.
1901 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
1902 'MusicOnHoldStop'. The sub type field has been removed.
1904 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
1905 carrier for another protocol.
1907 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
1908 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
1909 to the specific channel. 'Both' may be specified to play a tone to both
1910 channels. The old 'yes' option is still accepted as a way of playing the
1911 tone to Channel2 only.
1913 * The AMI 'Status' response event to the AMI Status action replaces the
1914 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
1915 indicate what bridge the channel is currently in.
1917 * The AMI 'Hold' event has been moved out of individual channel drivers, into
1918 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
1921 * The AMI events in app_queue have been made more consistent with each other.
1922 Events that reference channels (QueueCaller* and Agent*) will show
1923 information about each channel. The (infamous) 'Join' and 'Leave' AMI
1924 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
1926 * The 'MCID' AMI event now publishes a channel snapshot when available and
1927 its non-channel-snapshot parameters now use either the "MCallerID" or
1928 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
1929 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
1930 parameters in the channel snapshot.
1932 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
1933 'AgentLogin' and 'AgentLogoff' respectively.
1935 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
1936 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
1938 * 'ChannelUpdate' events have been removed.
1940 * All AMI events now contain a 'SystemName' field, if available.
1942 * Local channel optimization is now conveyed in two events:
1943 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
1944 when the Local channel driver begins attempting to optimize itself out of
1945 the media path; the End event is sent after the channel halves have
1946 successfully optimized themselves out of the media path.
1948 * Local channel information in events is now prefixed with 'LocalOne' and
1949 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
1950 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
1951 and 'LocalOptimizationEnd' events.
1953 * The option 'allowmultiplelogin' can now be set or overriden in a particular
1954 account. When set in the general context, it will act as the default
1955 setting for defined accounts.
1957 * The 'BridgeAction' event was removed. It technically added no value, as the
1958 Bridge Action already receives confirmation of the bridge through a
1959 successful completion Event.
1961 * The 'BridgeExec' events were removed. These events duplicated the events that
1962 occur in the Briding API, and are conveyed now through BridgeCreate,
1963 BridgeEnter, and BridgeLeave events.
1965 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
1966 previous versions. They now report all SR/RR packets sent/received, and
1967 have been restructured to better reflect the data sent in a SR/RR. In
1968 particular, the event structure now supports multiple report blocks.
1970 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
1971 raised when a blind transfer/attended transfer completes successfully.
1972 They contain information about the transfer that just completed, including
1973 the location of the transfered channel.
1975 * Added a 'security' class to AMI which outputs the required fields for
1976 security messages similar to the log messages from res_security_log
1978 * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
1979 that describes the status value in a human readable string.
1981 CDR (Call Detail Records)
1983 * Significant changes have been made to the behavior of CDRs. The CDR engine
1984 was effectively rewritten and built on the Stasis message bus. For a full
1985 definition of CDR behavior in Asterisk 12, please read the specification
1986 on the Asterisk wiki (wiki.asterisk.org).
1988 * CDRs will now be created between all participants in a bridge. For each
1989 pair of channels in a bridge, a CDR is created to represent the path of
1990 communication between those two endpoints. This lets an end user choose who
1991 to bill for what during bridge operations with multiple parties.
1993 * The duration, billsec, start, answer, and end times now reflect the times
1994 associated with the current CDR for the channel, as opposed to a cumulative
1995 measurement of all CDRs for that channel.
1997 * When a CDR is dispatched, user defined CDR variables from both parties are
1998 included in the resulting CDR. If both parties have the same variable, only
1999 the Party A value is provided.
2001 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
2002 information regarding the CDR engine is logged as verbose messages. This
2003 option should only be used if the behavior of the CDR engine needs to be
2006 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
2007 normally configured in cdr.conf.
2009 * Added CLI command 'cdr show active {channel}'. When {channel} is not
2010 specified, this command provides a summary of the channels with CDR
2011 information and their statistics. When {channel} is specified, it shows
2012 detailed information about all records associated with {channel}.
2014 CEL (Channel Event Logging)
2016 * CEL has undergone significant rework in Asterisk 12, and is now built on the
2017 Stasis message bus. Please see the specification for CEL on the Asterisk
2018 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
2021 * The 'extra' field of all CEL events that use it now consists of a JSON blob
2022 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
2024 * BLINDTRANSFER events now report the transferee bridge unique
2025 identifier, extension, and context in a JSON blob as the extra string
2026 instead of the transferee channel name as the peer.
2028 * ATTENDEDTRANSFER events now report the peer as NULL and additional
2029 information in the 'extra' string as a JSON blob. For transfers that occur
2030 between two bridged channels, the 'extra' JSON blob contains the primary
2031 bridge unique identifier, the secondary channel name, and the secondary
2032 bridge unique identifier. For transfers that occur between a bridged channel
2033 and a channel running an app, the 'extra' JSON blob contains the primary
2034 bridge unique identifier, the secondary channel name, and the app name.
2036 * LOCAL_OPTIMIZE events have been added to convey local channel
2037 optimizations with the record occurring for the semi-one channel and
2038 the semi-two channel name in the peer field.
2040 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
2041 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
2042 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
2043 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
2044 regardless of whether or not that bridge happens to contain multiple
2049 * When compiled with '--enable-dev-mode', the astobj2 library will now add
2050 several CLI commands that allow for inspection of ao2 containers that
2051 register themselves with astobj2. The CLI commands are 'astobj2 container
2052 dump', 'astobj2 container stats', and 'astobj2 container check'.
2054 * Added specific CLI commands for bridge inspection. This includes 'bridge
2055 show all', which lists all bridges in the system, and 'bridge show {id}',
2056 which provides specific information about a bridge.
2058 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
2059 ejecting the channels currently in the bridge. If the channels cannot
2060 continue in the dialplan or application that put them in the bridge, they
2063 * Added command 'bridge kick'. This will eject a single channel from a bridge.
2065 * Added commands to inspect and manipulate the registered bridge technologies.
2066 This include 'bridge technology show', which lists the registered bridge
2067 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
2068 which controls whether or not a registered bridge technology can be used
2069 during smart bridge operations. If a technology is suspended, it will not
2070 be used when a bridge technology is picked for channels; when unsuspended,
2071 it can be used again.
2073 * The command 'config show help {module} {type} {option}' will show
2074 configuration documentation for modules with XML configuration
2075 documentation. When {module}, {type}, and {option} are omitted, a listing
2076 of all modules with registered documentation is displayed. When {module}
2077 is specified, a listing of all configuration types for that module is
2078 displayed, along with their synopsis. When {module} and {type} are
2079 specified, a listing of all configuration options for that type are
2080 displayed along with their synopsis. When {module}, {type}, and {option}
2081 are specified, detailed information for that configuration option is
2084 * Added 'core show sounds' and 'core show sound' CLI commands. These display
2085 a listing of all installed media sounds available on the system and
2086 detailed information about a sound, respectively.
2088 * 'xmldoc dump' has been added. This CLI command will dump the XML
2089 documentation DOM as a string to the specified file. The Asterisk core
2090 will populate certain XML elements pulled from the source files with
2091 additional run-time information; this command lets a user produce the
2092 XML documentation with all information.
2096 * Parking has been pulled from core and placed into a separate module called
2097 res_parking. See Parking changes below for more details. Configuration for
2098 parking should now be performed in res_parking.conf. Configuration for
2099 parking in features.conf is now unsupported.
2101 * Core attended transfers now have several new options. While performing an
2102 attended transfer, the transferer now has the following options:
2103 - *1 - cancel the attended transfer (configurable via atxferabort)
2104 - *2 - complete the attended transfer, dropping out of the call
2105 (configurable via atxfercomplete)
2106 - *3 - complete the attended transfer, but stay in the call. This will turn
2107 the call into a multi-party bridge (configurable via atxferthreeway)
2108 - *4 - swap to the other party. Once an attended transfer has begun, this
2109 options may be used multiple times (configurable via atxferswap)
2111 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
2112 must be on the channel initiating the transfer to have any effect.
2114 * The BRIDGE_FEATURES channel variable would previously only set features for
2115 the calling party and would set this feature regardless of whether the
2116 feature was in caps or in lowercase. Use of a caps feature for a letter
2117 will now apply the feature to the calling party while use of a lowercase
2118 letter will apply that feature to the called party.
2120 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
2122 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
2123 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
2124 activated the dynamic feature.
2126 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
2127 only on the channel executing the dynamic feature. Executing a dynamic
2128 feature on the bridge peer in a multi-party bridge will execute it on all
2129 peers of the activating channel.
2131 * You can now have the settings for a channel updated using the FEATURE()
2132 and FEATUREMAP() functions inherited to child channels by setting
2133 FEATURE(inherit)=yes.
2135 * automixmon now supports additional channel variables from automon including:
2136 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
2137 and TOUCH_MIXMONITOR_MESSAGE_STOP
2139 * A new general features.conf option 'recordingfailsound' has been added which
2140 allowssetting a failure sound for a user tries to invoke a recording feature
2141 such as automon or automixmon and it fails.
2143 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
2144 features.c for atxferdropcall=no to work properly. This option now just
2149 * Added log rotation strategy 'none'. If set, no log rotation strategy will
2150 be used. Given that this can cause the Asterisk log files to grow quickly,
2151 this option should only be used if an external mechanism for log management
2156 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
2157 will store the path information for that peer when it registers. Realtime
2158 tables can also use the 'supportpath' field to enable Path header support.
2160 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
2161 objectIdentifier. This maps to the supportpath option in sip.conf.
2165 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
2166 provides modules a useful abstraction on top of the many storage mechanisms
2167 in Asterisk, including the Asterisk Database, static configuration files,
2168 static Realtime, and dynamic Realtime. It also provides a caching service.
2169 Users can configure a hierarchy of data storage layers for specific modules
2172 * All future modules which utilize Sorcery for object persistence must have a
2173 column named "id" within their schema when using the Sorcery realtime module.
2174 This column must be able to contain a string of up to 128 characters in length.
2176 Security Events Framework
2178 * Security Event timestamps now use ISO 8601 formatted date/time instead of
2179 the "seconds-microseconds" format that it was using previously.
2183 * The Stasis message bus is a publish/subscribe message bus internal to
2184 Asterisk. Many services in Asterisk are built on the Stasis message bus,
2185 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
2186 Stasis can be configured in stasis.conf. Note that these parameters operate
2187 at a very low level in Asterisk, and generally will not require changes.
2191 * When a channel driver is configured to enable jiterbuffers, they are now
2192 applied unconditionally when a channel joins a bridge. If a jitterbuffer
2193 is already set for that channel when it enters, such as by the JITTERBUFFER
2194 function, then the existing jitterbuffer will be used and the one set by
2195 the channel driver will not be applied.
2199 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
2200 dialplan applications provided by the app_agent_pool module. Agents are
2201 connected with callers using the new AgentRequest dialplan application.
2202 The Agents:<agent-id> device state is available to monitor the status of an
2203 agent. See agents.conf.sample for valid configuration options.
2205 * The updatecdr option has been removed. Altering the names of channels on a
2206 CDR is not supported - the name of the channel is the name of the channel,
2207 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
2208 has also been removed, for the same reason.
2210 * The endcall and enddtmf configuration options are removed. Use the
2211 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
2212 channel before calling AgentLogin.
2216 * chan_bridge has been removed. Its functionality has been incorporated
2217 directly into the ConfBridge application itself.
2221 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
2222 of the specified span and its B-channels. Note that this command should
2223 only be used if you understand the risks it entails.
2225 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
2226 A range of channels can be specified to be destroyed. Note that this command
2227 should only be used if you understand the risks it entails.
2229 * Added the CLI command 'dahdi create channels'. A range of channels can be
2230 specified to be created, or the keyword 'new' can be used to add channels
2233 * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
2234 the exact configured mailbox name. For app_voicemail mailboxes this is
2237 * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
2241 * IPv6 support has been added. We are now able to bind to and
2242 communicate using IPv6 addresses.
2246 * The /b option has been removed.
2248 * chan_local moved into the system core and is no longer a loadable module.
2252 * Added general support for busy detection.
2254 * Added ECAM command support for Sony Ericsson phones.
2258 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
2259 SIP stack. A collection of resource modules provides the bulk of the SIP
2260 functionality. For more information on the new SIP channel driver, see
2261 https://wiki.asterisk.org/wiki/x/JYGLAQ
2265 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
2266 using the 'supportpath' setting, either on a global basis or on a peer basis.
2267 This setting enables Asterisk to route outgoing out-of-dialog requests via a
2268 set of proxies by using a pre-loaded route-set defined by the Path headers in
2269 the REGISTER request. See Realtime updates for more configuration information.
2271 * The SIP_CODEC family of variables may now specify more than one codec. Each
2272 codec must be separated by a comma. The first codec specified is the
2273 preferred codec for the offer. This allows a dialplan writer to specify both
2274 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
2276 * The 'callevents' parameter has been removed. Hold AMI events are now raised
2277 in the core, and can be filtered out using the 'eventfilter' parameter
2280 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
2281 codecs configured for a peer instead of the requested codec.
2283 * The option "register_retry_403" has been added to chan_sip to work around
2284 servers that are known to erroneously send 403 in response to valid
2285 REGISTER requests and allows Asterisk to continue attepmting to connect.
2289 * Added the 'immeddialkey' parameter. If set, when the user presses the
2290 configured key the already entered number will be immediately dialed. This
2291 is useful when the dialplan allows for variable length pattern matching.
2292 Valid options are '*' and '#'.
2294 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
2295 milliseconds) before a call forward is considered to not be answered.
2297 * The 'serviceurl' parameter allows Service URLs to be attached to line
2306 * The password option has been disabled, as the AgentLogin application no
2307 longer provides authentication.
2311 * Due to changes in the Asterisk core, this function is no longer needed to
2312 preserve a MixMonitor on a channel during transfer operations and dialplan
2313 execution. It is effectively obsolete.
2317 * The 'amaflags' and 'accountcode' attributes for the CDR function are
2318 deprecated. Use the CHANNEL function instead to access these attributes.
2320 * The 'l' option has been removed. When reading a CDR attribute, the most
2321 recent record is always used. When writing a CDR attribute, all non-finalized
2324 * The 'r' option has been removed, for the same reason as the 'l' option.
2326 * The 's' option has been removed, as LOCKED semantics no longer exist in the
2331 * A new function CDR_PROP has been added. This function lets you set properties
2332 on a channel's active CDRs. This function is write-only. Properties accept
2333 boolean values to set/clear them on the channel's CDRs. Valid properties
2335 - 'party_a' - make this channel the preferred Party A in any CDR between two
2336 channels. If two channels have this property set, the creation time of the
2337 channel is used to determine who is Party A. Note that dialed channels are
2338 never Party A in a CDR.
2339 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
2340 application when set to True, and analogous to the 'e' option in ResetCDR
2345 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
2346 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
2347 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
2350 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
2351 string, i.e., [[context],extension],priority. If set on a channel, if a
2352 channel leaves a bridge but is not hung up it will resume dialplan execution
2357 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
2358 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
2359 The value of this setting is ignored when disabled is used for the argument.
2363 * A new function provided by chan_pjsip, this function can be used in
2364 conjunction with the Dial application to construct a dial string that will
2365 dial all contacts on an Address of Record associated with a chan_pjsip
2370 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
2371 outbound channel prior to dialing.
2375 * Redirecting reasons can now be set to arbitrary strings. This means
2376 that the REDIRECTING dialplan function can be used to set the redirecting
2377 reason to any string. It also allows for custom strings to be read as the
2378 redirecting reason from SIP Diversion headers.
2382 * The SPEECH_ENGINE function now supports read operations. When read from, it
2383 will return the current value of the requested attribute.
2387 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
2388 system as mailbox@context. The rest of the system cannot add @default
2389 to mailbox identifiers for app_voicemail that do not specify a context
2390 any longer. It is a mailbox identifier format that should only be
2391 interpreted by app_voicemail.
2397 res_agi (Asterisk Gateway Interface)
2399 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
2401 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
2404 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
2405 will start the playback of the audio at the position specified. It will
2406 also return the final position of the file in 'endpos'.
2408 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
2409 channel variable if the user stopped the file playback or if a remote
2410 entity stopped the playback. If neither stopped the playback, it will
2411 indicate the overall success/failure of the playback. If stopped early,
2412 the final offset of the file will be set in the CPLAYBACKOFFSET channel
2415 * The SAY ALPHA command now accepts an additional parameter to control
2416 whether it specifies the case of uppercase, lowercase, or all letters to
2417 provide functionality similar to SayAlphaCase.
2419 res_ari (Asterisk RESTful Interface) (and others)
2421 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
2422 control telephony primitives in Asterisk by remote client. This includes
2423 channels, bridges, endpoints, media, and other fundamental concepts. Users
2424 of ARI can develop their own communications applications, controlling
2425 multiple channels using an HTTP RESTful interface and receiving JSON events
2426 about the objects via a WebSocket connection. ARI can be configured in
2427 Asterisk via ari.conf. For more information on ARI, see
2428 https://wiki.asterisk.org/wiki/x/0YCLAQ
2432 * Parking has been extracted from the Asterisk core as a loadable module,
2433 res_parking. Configuration for parking is now provided by res_parking.conf.
2434 Configuration through features.conf is no longer supported.
2436 * res_parking uses the configuration framework. If an invalid configuration is
2437 supplied, res_parking will fail to load or fail to reload. Previously,
2438 invalid configurations would generally be accepted, with certain errors
2439 resulting in individually disabled parking lots.
2441 * Parked calls are now placed in bridges. While this is largely an
2442 architectural change, it does have implications on how channels in a parking
2443 lot are viewed. For example, commands that display channels in bridges will
2444 now also display the channels in a parking lot.
2446 * The order of arguments for the new parking applications have been modified.
2447 Timeout and return context/exten/priority are now implemented as options,
2448 while the name of the parking lot is now the first parameter. See the
2449 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
2450 in-depth information as well as syntax.
2452 * Extensions are by default no longer automatically created in the dialplan to
2453 park calls or pickup parked calls. Generation of dialplan extensions can be
2454 enabled using the 'parkext' configuration option.
2456 * ADSI functionality for parking is no longer supported. The 'adsipark'
2457 configuration option has been removed as a result.
2459 * The PARKINGSLOT channel variable has been deprecated in favor of
2460 PARKING_SPACE to match the naming scheme of the new system.
2462 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
2463 channel even when the configuration option 'comebactoorigin' is enabled.
2465 * A new CLI command 'parking show' has been added. This allows a user to
2466 inspect the parking lots that are currently in use.
2467 'parking show <parkinglot>' will also show the parked calls in a specific
2470 * The CLI command 'parkedcalls' is now deprecated in favor of
2471 'parking show <parkinglot>'.
2473 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
2474 can be used to get a list of parked calls for a specific parking lot.
2476 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
2477 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
2478 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
2479 longer a required argument.
2481 * The ParkAndAnnounce application is now provided through res_parking instead
2482 of through the separate app_parkandannounce module.
2484 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
2485 by default. Instead, it will follow the timeout rules of the parking lot. The
2486 old behavior can be reproduced by using the 'c' option.
2488 * Dynamic parking lots will now fail to be created under the following
2490 - if the parking lot specified by PARKINGDYNAMIC does not exist
2491 - if they require exclusive park and parkedcall extensions which overlap
2492 with existing parking lots.
2494 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
2495 currently contain no calls. Dynamic parking lots containing parked calls
2496 will persist through the reloads without alteration.
2498 * If 'parkext_exclusive' is set for a parking lot and that extension is
2499 already in use when that parking lot tries to register it, this is now
2500 considered a parking system configuration error. Configurations which do
2501 this will be rejected.
2503 * Added channel variable PARKER_FLAT. This contains the name of the extension
2504 that would be used if 'comebacktoorigin' is enabled. This can be useful when
2505 comebacktoorigin is disabled, but the dialplan or an external control
2506 mechanism wants to use the extension in the park-dial context that was
2507 generated to re-dial the parker on timeout.
2509 res_pjsip (and many others)
2511 * A large number of resource modules make up the SIP stack based on pjsip.
2512 The chan_pjsip channel driver users these resource modules to provide
2513 various SIP functionality in Asterisk. The majority of configuration for
2514 these modules is performed in pjsip.conf. Other modules may use their
2515 own configuration files.
2517 * Added 'set_var' option for an endpoint. For each variable specified that
2518 variable gets set upon creation of a channel involving the endpoint.
2522 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
2523 them, an Asterisk-specific version of PJSIP needs to be installed.
2524 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
2526 res_statsd/res_chan_stats
2528 * A new resource module, res_statsd, has been added, which acts as a statsd
2529 client. This module allows Asterisk to publish statistics to a statsd
2530 server. In conjunction with res_chan_stats, it will publish statistics about
2531 channels to the statsd server. It can be configured via res_statsd.conf.
2535 * Device state for XMPP buddies is now available using the following format:
2536 XMPP/<client name>/<buddy address>
2537 If any resource is available the device state is considered to be not in use.
2538 If no resources exist or all are unavailable the device state is considered
2545 Realtime/Database Scripts
2547 * Asterisk previously included example db schemas in the contrib/realtime/
2548 directory of the source tree. This has been replaced by a set of database
2549 migrations using the Alembic framework. This allows you to use alembic to
2550 initialize the database for you. It will also serve as a database migration
2551 tool when upgrading Asterisk in the future.
2553 See contrib/ast-db-manage/README.md for more details.
2557 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
2558 This python script will convert an existing sip.conf file to a
2559 pjsip.conf file, for use with the chan_pjsip channel driver. This script
2560 is meant to be an aid in converting an existing chan_sip configuration to
2561 a chan_pjsip configuration, but it is expected that configuration beyond
2562 what the script provides will be needed.
2564 ------------------------------------------------------------------------------
2565 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
2566 ------------------------------------------------------------------------------
2570 * The Asterisk build system will now build and install a shared library
2571 (libasteriskssl.so) used to wrap various initialization and shutdown functions
2572 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
2573 that Asterisk can ensure that these functions do *not* get called by any
2574 modules that are loaded into Asterisk, since they should only be called once
2575 in any single process. If desired, this feature can be disabled by supplying
2576 the "--disable-asteriskssl" option to the configure script.
2578 * A new make target, 'full', has been added to the Makefile. This performs
2579 the same compilation actions as make all, but will also scan the entirety of
2580 each source file for documentation. This option is needed to generate AMI
2581 event documentation. Note that your system must have Python in order for
2582 this make target to succeed.
2584 * The optimization portion of the build system has been reworked to avoid
2585 broken builds on certain architectures. All architecture-specific
2586 optimization has been removed in favor of using -march=native to allow gcc
2587 to detect the environment in which it is running when possible. This can
2588 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
2590 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
2591 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
2593 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
2594 previously parsed the header file to obtain the version of Asterisk, you
2595 will now have to go through Asterisk to get the version information.
2603 * Added 'F()' option. Similar to the dial option, this can be supplied with
2604 arguments indicating where the callee should go after the caller is hung up,
2605 or without options specified, the priority after the Queue will be used.
2610 * Added menu action admin_toggle_mute_participants. This will mute / unmute
2611 all non-admin participants on a conference. The confbridge configuration
2612 file also allows for the default sounds played to all conference users when
2613 this occurs to be overriden using sound_participants_unmuted and
2614 sound_participants_muted.
2616 * Added menu action participant_count. This will playback the number of
2617 current participants in a conference.
2619 * Added announcement configuration option to user profile. If set the sound
2620 file will be played to the user, and only the user, upon joining the
2623 * Added record_file_append option that defaults to "yes", but if set to no
2624 will create a new file between each start/stop recording.
2629 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
2630 channels respectively before the callee channels are called.
2635 * Added support for IPv6.
2637 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
2638 external process will cause the current playlist to be cleared, including
2639 stopping any audio file that is currently playing. This is useful when you
2640 want to interrupt audio playback only when specific DTMF is entered by the
2646 * A new option, 'I' has been added to app_followme. By setting this option,
2647 Asterisk will not update the caller with connected line changes when they
2648 occur. This is similar to app_dial and app_queue.
2650 * The 'N' option is now ignored if the call is already answered.
2652 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
2653 and caller channels respectively before the callee channels are called.
2655 * The winning FollowMe outgoing call is now put on hold if the caller put it on
2661 * MixMonitor hooks now have IDs associated with them which can be used to
2662 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
2663 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
2664 now accepts that ID as an argument.
2666 * Added 'm' option, which stores a copy of the recording as a voicemail in the
2667 indicated mailboxes.
2672 * The connect action in app_mysql now allows you to specify a port number to
2673 connect to. This is useful if you run a MySQL server on a non-standard
2679 * Increased the default number of allowed destinations from 5 to 12.
2684 * The app_page application now no longer depends on DAHDI or app_meetme. It
2685 has been re-architected to use app_confbridge internally.
2690 * Added queue options autopausebusy and autopauseunavail for automatically
2691 pausing a queue member when their device reports busy or congestion.
2693 * The 'ignorebusy' option for queue members has been deprecated in favor of
2694 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
2695 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
2696 per interface basis. Individual ringinuse values can now be set in
2697 queues.conf via an argument to member definitions. Lastly, the queue
2698 'ringinuse' setting now only determines defaults for the per member
2699 'ringinuse' setting and does not override per member settings like it does
2700 in earlier versions.
2702 * Added 'F()' option. Similar to the dial option, this can be supplied with
2703 arguments indicating where the callee should go after the caller is hung up,
2704 or without options specified, the priority after the Queue will be used.
2706 * Added new option log_member_name_as_agent, which will cause the membername to
2707 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
2708 state_interface has been set.
2710 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
2712 * App_queue will now play periodic announcements for the caller that
2713 holds the first position in the queue while waiting for answer.
2717 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
2718 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
2719 changed arguments to SayUnixTime so that every option is truly optional even
2720 when using multiple options (so that j option could be used without having to
2721 manually specify timezone and format) There are other benefits, e.g., format
2722 can now be used without specifying time zone as well.
2727 * Addition of the VM_INFO function - see Function changes.
2729 * The imapserver, imapport, and imapflags configuration options can now be
2730 overriden on a user by user basis.
2732 * When voicemail plays a message's envelope with saycid set to yes, when
2733 reaching the caller id field it will play a recording of a file with the same
2734 base name as the sender's callerid if there is a similarly named file in
2735 <astspooldir>/recordings/callerids/
2737 * Voicemails now contains a unique message identifier "msg_id", which is stored
2738 in the message envelope with the sound files. IMAP backends will now store
2739 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
2740 backends will store the message identifier in a "msg_id" column. See
2741 UPGRADE.txt for more information.
2743 * Added VoiceMailPlayMsg application. This application will play a single
2744 voicemail message from a mailbox. The result of the application, SUCCESS or
2745 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
2750 * Hangup handlers can be attached to channels using the CHANNEL() function.
2751 Hangup handlers will run when the channel is hung up similar to the h
2752 extension. The hangup_handler_push option will push a GoSub compatible
2753 location in the dialplan onto the channel's hangup handler stack. The
2754 hangup_handler_pop option will remove the last added location, and optionally
2755 replace it with a new GoSub compatible location. The hangup_handler_wipe
2756 option will remove all locations on the stack, and optionally add a new
2759 * The expression parser now recognizes the ABS() absolute value function,
2760 which will convert negative floating point values to positive values.
2762 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
2763 control of faxdetect.
2765 * Addition of the VM_INFO function that can be used to retrieve voicemail
2766 user information, such as the email address and full name.
2767 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
2770 * The REDIRECTING function now supports the redirecting original party id
2773 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
2774 lets you set some of the configuration options from the [general] section
2775 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
2776 the key sequence used to activate built-in features, such as blindxfer,
2777 and automon. See the built-in documentation for details.
2779 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
2780 instead of simply the uri. This is the format that MessageSend() can use
2781 in the from parameter for outgoing SIP messages.
2783 * Added the PRESENCE_STATE function. This allows retrieving presence state
2784 information from any presence state provider. It also allows setting
2785 presence state information from a CustomPresence presence state provider.
2786 See AMI/CLI changes for related commands.
2788 * Added the AMI_CLIENT function to make manager account attributes available
2789 to the dialplan. It currently supports returning the current number of
2790 active sessions for a given account.
2792 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
2793 and the REDIRECTING functions.
2801 * Added a manager event "LocalBridge" for local channel call bridges between
2802 the two pseudo-channels created.
2807 * Added dialtone_detect option for analog ports to disconnect incoming
2808 calls when dialtone is detected.
2810 * Added option colp_send to send ISDN connected line information. Allowed
2811 settings are block, to not send any connected line information; connect, to
2812 send connected line information on initial connect; and update, to send
2813 information on any update during a call. Default is update.
2815 * Add options namedcallgroup and namedpickupgroup to support installations
2816 where a higher number of groups (>64) is required.
2818 * Added support to use private party ID information with PRI calls.
2823 * A new channel driver named chan_motif has been added which provides support for
2824 Google Talk and Jingle in a single channel driver. This new channel driver includes
2825 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
2826 hold, unhold, and ringing notification. It is also compliant with the current Jingle
2827 specification, current Google Jingle specification, and the original Google Talk
2833 * Added NAT support for RTP. Setting in config is 'nat', which can be set
2834 globally and overriden on a peer by peer basis.
2836 * Direct media functionality has been added. Options in config are:
2837 directmedia (directrtp) and directrtpsetup (earlydirect)
2839 * ChannelUpdate events now contain a CallRef header.
2844 * Asterisk will no longer substitute CID number for CID name in the display
2845 name field if CID number exists without a CID name. This change improves
2846 compatibility with certain device features such as Avaya IP500's directory
2849 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
2850 created using that setting to not be removed during SIP reload.
2852 * Added settings recordonfeature and recordofffeature. When receiving an INFO
2853 request with a "Record:" header, this will turn the requested feature on/off.
2854 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
2855 dynamic features must be enabled and configured properly on the requesting
2856 channel for this to function properly.
2858 * Add support to realtime for the 'callbackextension' option.
2860 * When multiple peers exist with the same address, but differing
2861 callbackextension options, incoming requests that are matched by address
2862 will be matched to the peer with the matching callbackextension if it is
2865 * Two new NAT options, auto_force_rport and auto_comedia, have been added
2866 which set the force_rport and comedia options automatically if Asterisk
2867 detects that an incoming SIP request crossed a NAT after being sent by
2868 the remote endpoint.
2870 * The default global nat setting in sip.conf has been changed from force_rport
2871 to auto_force_rport.
2873 * NAT settings are now a combinable list of options. The equivalent of the
2874 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
2876 * Adds an option send_diversion which can be disabled to prevent
2877 diversion headers from automatically being added to INVITE requests.
2879 * Add support for lightweight NAT keepalive. If enabled a blank packet will
2880 be sent to the remote host at a given interval to keep the NAT mapping open.
2881 This can be enabled using the keepalive configuration option.
2883 * Add option 'tonezone' to specify country code for indications. This option
2884 can be set both globally and overridden for specific peers.
2886 * The SIP Security Events Framework now supports IPv6.
2888 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
2889 between multiple user agents. When set, for directmedia reinvites,
2890 Asterisk will not send an immediate reinvite on an incoming call leg. This
2891 option is useful when peered with another SIP user agent that is known to
2892 send immediate direct media reinvites upon call establishment.
2894 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
2897 * Add options subminexpiry and submaxexpiry to set limits of subscription
2898 timer independently from registration timer settings. The setting of the
2899 registration timer limits still is done by options minexpiry, maxexpiry
2900 and defaultexpiry. For backwards compatibility the setting of minexpiry
2901 and maxexpiry also is used to configure the subscription timer limits if
2902 subminexpiry and submaxexpiry are not set in sip.conf.
2904 * Set registration timer limits to default values when reloading sip
2905 configuration and values are not set by configuration.
2907 * Add options namedcallgroup and namedpickupgroup to support installations
2908 where a higher number of groups (>64) is required.
2910 * When a MESSAGE request is received, the address the request was received from
2911 is now saved in the SIP_RECVADDR variable.
2913 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
2914 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
2915 the ANI2/OLI information is set on the channel, which can be retrieved using
2916 the CALLERID function.
2918 * Peers can now be configured to support negotiation of ICE candidates using
2919 the setting icesupport. See res_rtp_asterisk changes for more information.
2921 * Added support for format attribute negotiation. See the Codecs changes for
2924 * Extra headers specified with SIPAddHeader are sent with the REFER message
2925 when using Transfer application. See refer_addheaders in sip.conf.sample.
2927 * Added support to use private party ID information with calls.
2929 * Adds an option discard_remote_hold_retrieval that when set stops telling
2930 the peer to start music on hold.
2935 * Added skinny version 17 protocol support.
2939 --------------------
2940 * Added option 'dtmf_duration' allowing playback time of DTMF tones to be set
2942 * Modified option 'date_format' to allow options to display date in 31Jan and Jan31
2943 formats as options 0 and 1. The previous options 0 and 1 now map to options 2 and 3
2944 as per the UNISTIM protocol.
2946 * Fixed issues with dialtone not matching indications.conf and mute stopping rx
2947 as well as tx. Also fixed issue with call "Timer" displaying as French "Dur\E9e"
2949 * Added ability to use multiple lines for a single phone. This allows multiple
2950 calls to occur on a single phone, using callwaiting and switching between calls.
2952 * Added option 'sharpdial' allowing end dialing by pressing # key
2954 * Added option 'interdigit_timer' to control phone dial timeout
2956 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
2958 * Added global 'debug' option, that enables debug in channel driver
2960 * Added ability to translate on-screen menu in multiple languages. Tested on
2961 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
2962 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
2965 * In addition to English added French and Russian languages for on-screen menus
2967 * Reworked dialing number input: added dialing by timeout, immediate dial on
2968 on dialplan compare, phone number length now not limited by screen size
2970 * Added ability to pickup a call using features.conf defined value and
2976 * Add options namedcallgroup and namedpickupgroup to support installations
2977 where a higher number of groups (>64) is required.
2979 * Added support to use private party ID information with calls.
2984 * The minimum DTMF duration can now be configured in asterisk.conf
2985 as "mindtmfduration". The default value is (as before) set to 80 ms.
2986 (previously it was only available in source code)
2988 * Named ACLs can now be specified in acl.conf and used in configurations that
2989 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
2990 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
2991 working ACL. In addition, some CLI commands have been added to provide
2992 show information and allow for module reloading - see CLI Changes.
2994 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
2995 items (separated by commas), and items in the rule can be negated by prefixing
2996 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
2997 longer necessray to control the order that the 'permit' and 'deny' columns are
2998 returned from queries.
3000 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
3001 be used within the dynamic weight attribute when specifying a mapping.
3003 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
3004 header, instead of putting the user defined event name there. When enabled
3005 the UserDefType header is added for user defined events. This feature is
3006 enabled with the setting show_user_defined.
3008 * Macro has been deprecated in favor of GoSub. For redirecting and connected
3009 line purposes use the following variables instead of their macro equivalents:
3010 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
3011 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
3012 cc_callback_macro in channel configurations.
3014 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
3017 * Call files now support the "early_media" option to connect with an outgoing
3018 extension when early media is received.
3020 * Added support to use private party ID information with calls.
3025 * A new channel variable, AGIEXITONHANGUP, has been added which allows
3026 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
3027 AGI application would exit immediately after a channel hangup is detected.
3029 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
3030 are resolved and each address is attempted in turn until one succeeds or
3034 AMI (Asterisk Manager Interface)
3036 * The originate action now has an option "EarlyMedia" that enables the
3037 call to bridge when we get early media in the call. Previously,
3038 early media was disregarded always when originating calls using AMI.
3040 * Added setvar= option to manager accounts (much like sip.conf)
3042 * Originate now generates an error response if the extension given is not found
3045 * MixMonitor will now show IDs associated with the mixmonitor upon creating
3046 them if the i(variable) option is used. StopMixMonitor will accept
3047 MixMonitorID as an option to close specific MixMonitors.
3049 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
3050 updated to include information about peers configured with
3051 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
3052 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
3053 returned if auto_force_rport is not enabled.
3055 * Added SIPpeerstatus manager command which will generate PeerStatus events
3056 similar to the existing PeerStatus events found in chan_sip on demand.
3058 * Hangup now can take a regular expression as the Channel option. If you want
3059 to hangup multiple channels, use /regex/ as the Channel option. Existing
3060 behavior to hanging up a single channel is unchanged, but if you pass a regex,
3061 the manager will send you a list of channels back that were hung up.
3063 * Support for IPv6 addresses has been added.
3065 * AMI Events can now be documented in the Asterisk source. Note that AMI event
3066 documentation is only generated when Asterisk is compiled using 'make full'.
3067 See the CLI section for commands to display AMI event information.
3069 * The AMI Hangup event now includes the AccountCode header so you can easily
3070 correlate with AMI Newchannel events.
3072 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
3073 the StateInterface of the queue member.
3075 * Added AMI event SessionTimeout in the Call category that is issued when a
3076 call is terminated due to either RTP stream inactivity or SIP session timer
3079 * CEL events can now contain a user defined header UserDefType. See core
3080 changes for more information.
3082 * OOH323 ChannelUpdate events now contain a CallRef header.
3084 * Added PresenceState command. This command will report the presence state for
3085 the given presence provider.
3087 * Added Parkinglots command. This will list all parking lots as a series of
3088 AMI Parkinglot events.
3090 * Added MessageSend command. This behaves in the same manner as the
3091 MessageSend application, and is a technolgoy agnostic mechanism to send out
3092 of call text messages.
3094 * Added "message" class authorization. This grants an account permission to
3095 send out of call messages. Write-only.
3100 * The "dialplan add include" command has been modified to create context a context
3101 if one does not already exist. For instance, "dialplan add include foo into bar"
3102 will create context "bar" if it does not already exist.
3104 * A "dialplan remove context" command has been added to remove a context from
3107 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
3108 filenames of all running mixmonitors on a channel.
3110 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
3111 numeric instead of 0, 1, or 2.
3113 * "stun show status" will show a table describing how the STUN client is
3116 * "acl show [named acl]" will show information regarding a Named ACL. The
3117 acl module can be reloaded with "reload acl".
3119 * Added CLI command to display AMI event information - "manager show events",
3120 which shows a list of all known and documented AMI events, and "manager show
3121 event [event name]", which shows detail information about a specific AMI
3124 * The result of the CLI command "queue show" now includes the state interface
3125 information of the queue member.
3127 * The command "core set verbose" will now set a separate level of logging for
3128 each remote console without affecting any other console.
3130 * Added command "cdr show pgsql status" to check connection status
3132 * "sip show channel" will now display the complete route set.
3134 * Added "presencestate list" command. This command will list all custom
3135 presence states that have been set by using the PRESENCE_STATE dialplan
3138 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
3139 command. This changes a custom presence to a new state.
3144 * Codec lists may now be modified by the '!' character, to allow succinct
3145 specification of a list of codecs allowed and disallowed, without the
3146 requirement to use two different keywords. For example, to specify all
3147 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
3149 * Add support for parsing SDP attributes, generating SDP attributes, and
3150 passing it through. This support includes codecs such as H.263, H.264, SILK,
3151 and CELT. You are able to set up a call and have attribute information pass.
3152 This should help considerably with video calls.
3154 * The iLBC codec can now use a system-provided iLBC library if one is installed,
3155 just like the GSM codec.
3159 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
3160 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
3164 * Asterisk version and build information is now logged at the beginning of a
3167 * Threads belonging to a particular call are now linked with callids which get
3168 added to any log messages produced by those threads. Log messages can now be
3169 easily identified as involved with a certain call by looking at their call id.
3170 Call ids may also be attached to log messages for just about any case where
3171 it can be determined to be related to a particular call.
3173 * Each logging destination and console now have an independent notion of the
3174 current verbosity level. Logger.conf now allows an optional argument to
3175 the 'verbose' specifier, indicating the level of verbosity sent to that
3176 particular logging destination. Additionally, remote consoles now each
3177 have their own verbosity level. The command 'core set verbose' will now set
3178 a separate level for each remote console without affecting any other
3184 * Added 'announcement' option which will play at the start of MOH and between
3185 songs in modes of MOH that can detect transitions between songs (eg.
3191 * New per parking lot options: comebackcontext and comebackdialtime. See
3192 configs/features.conf.sample for more details.
3194 * Channel variable PARKER is now set when comebacktoorigin is disabled in
3197 * Channel variable PARKEDCALL is now set with the name of the parking lot
3198 when a timeout occurs.
3204 CDR Postgresql Driver
3206 * Added command "cdr show pgsql status" to check connection status
3209 CDR Adaptive ODBC Driver
3211 * Added schema option for databases that support specifying a schema.
3219 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
3220 CALENDAR_WRITE has completed successfully.
3225 * A new option, 'probation' has been added to rtp.conf
3226 RTP in strictrtp mode can now require more than 1 packet to exit learning
3227 mode with a new source (and by default requires 4). The probation option
3228 allows the user to change the required number of packets in sequence to any
3229 desired value. Use a value of 1 to essentially restore the old behavior.
3230 Also, with strictrtp on, Asterisk will now drop all packets until learning
3231 mode has successfully exited. These changes are based on how pjmedia handles
3232 media sources and source changes.
3234 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
3235 enabled or disabled using the icesupport setting. A variety of other
3236 settings have been introduced to configure STUN/TURN connections.
3241 * A new module, res_corosync, has been introduced. This module uses the
3242 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
3243 of Asterisk servers to both Message Waiting Indication (MWI) and/or
3244 Device State (presence) information. This module is very similar to, and
3245 is a replacement for the res_ais module that was in previous releases of
3251 * This module adds a cleaned up, drop-in replacement for res_jabber called
3252 res_xmpp. This provides the same externally facing functionality but is
3253 implemented differently internally. res_jabber has been deprecated in favor
3254 of res_xmpp; please see the UPGRADE.txt file for more information.
3259 * The safe_asterisk script has been updated to allow several of its parameters
3260 to be set from environment variables. This also enables a custom run
3261 directory of Asterisk to be specified, instead of defaulting to /tmp.
3263 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
3264 its value to determine the directory to assume is the top-level directory of
3265 the source tree. If the variable is not set, it defaults to the current
3266 behavior and uses the current working directory.
3268 ------------------------------------------------------------------------------
3269 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
3270 ------------------------------------------------------------------------------
3274 * Asterisk now has protocol independent support for processing text messages
3275 outside of a call. Messages are routed through the Asterisk dialplan.
3276 SIP MESSAGE and XMPP are currently supported. There are options in
3277 jabber.conf and sip.conf to allow enabling these features.
3278 -> jabber.conf: see the "sendtodialplan" and "context" options.
3279 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
3280 and "outofcall_message_context" options.
3281 The MESSAGE() dialplan function and MessageSend() application have been
3282 added to go along with this functionality. More detailed usage information
3283 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
3284 * If real-time text support (T.140) is negotiated, it will be preferred for
3285 sending text via the SendText application. For example, via SIP, messages
3286 that were once sent via the SIP MESSAGE request would be sent via RTP if
3287 T.140 text is negotiated for a call.
3291 * parkedmusicclass can now be set for non-default parking lots.
3293 Asterisk Manager Interface
3294 --------------------------
3295 * PeerStatus now includes Address and Port.
3296 * Added Hold events for when the remote party puts the call on and off hold
3297 for chan_dahdi ISDN channels.
3298 * Added new action MeetmeListRooms to list active conferences (shows same
3299 data as "meetme list" at the CLI).
3300 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
3301 Description field that is set by 'description' in the channel configuration
3303 * Added Uniqueid header to UserEvent.
3304 * Added new action FilterAdd to control event filters for the current session.
3305 This requires the system permission and uses the same filter syntax as
3306 filters that can be defined in manager.conf
3307 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
3308 versions had some instances of the event converted, but others were left
3309 as-is. All Unlink events should now be converted to Bridge events. The AMI
3310 protocol version number was incremented to 1.2 as a result of this change.
3312 Asterisk HTTP Server
3313 --------------------------
3314 * The HTTP Server can bind to IPv6 addresses.
3317 --------------------------
3318 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
3319 with busydetect. usage example: busypattern=200,200,200,600
3322 --------------------------
3323 * New 'gtalk show settings' command showing the current settings loaded from
3325 * The 'logger reload' command now supports an optional argument, specifying an
3326 alternate configuration file to use.
3327 * 'dialplan add extension' command will now automatically create a context if
3328 the specified context does not exist with a message indicated it did so.
3329 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
3330 Description field which can be populated with 'description' in the channel
3331 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
3334 --------------------------
3335 * The filter option in cdr_adaptive_odbc now supports negating the argument,
3336 thus allowing records which do NOT match the specified filter.
3337 * Added ability to log CONGESTION calls to CDR
3340 --------------------------
3341 * Ability to define custom SILK formats in codecs.conf.
3342 * Addition of speex32 audio format with translation.
3343 * CELT codec pass-through support and ability to define
3344 custom CELT formats in codecs.conf.
3345 * Ability to read raw signed linear files with sample rates
3346 ranging from 8khz - 192khz. The new file extensions introduced
3347 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
3348 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
3349 Skinny, H.323, etc) can still only support the following codecs:
3350 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
3351 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
3352 Video: h261, h263, h263p, h264, mpeg4
3357 --------------------------
3358 * New highly optimized and customizable ConfBridge application capable of
3359 mixing audio at sample rates ranging from 8khz-96khz.
3360 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
3361 and bridge profiles on a channel.
3362 * CONFBRIDGE_INFO dialplan function capable of retrieving information
3363 about a conference such as locked status and number of parties, admins,
3365 * Addition of video_mode option in confbridge.conf for adding video support
3366 into a bridge profile.
3367 * Addition of the follow_talker video_mode in confbridge.conf. This video
3368 mode dynamically switches the video feed to always display the loudest talker
3369 supplying video in the conference.
3373 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
3374 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
3375 variables from asterisk.conf.
3379 * Addition of the JITTERBUFFER dialplan function. This function allows
3380 for jitterbuffering to occur on the read side of a channel. By using
3381 this function conference applications such as ConfBridge and MeetMe can
3382 have the rx streams jitterbuffered before conference mixing occurs.
3383 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
3385 * Added STRREPLACE function. This function let's the user search a variable
3386 for a given string to replace with another string as many times as the
3387 user specifies or just throughout the whole string.
3388 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
3389 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
3390 * Added extensions to chan_ooh323 in function CHANNEL()
3392 libpri channel driver (chan_dahdi) DAHDI changes
3393 --------------------------
3394 * Added moh_signaling option to specify what to do when the channel's bridged
3395 peer puts the ISDN channel on hold.
3396 * Added display_send and display_receive options to control how the display ie
3397 is handled. To send display text from the dialplan use the SendText()
3398 application when the option is enabled.
3399 * Added mcid_send option to allow sending a MCID request on a span.
3402 --------------------------
3403 * Added setvar option to calendar.conf to allow setting channel variables on
3404 notification channels.
3405 * Added "calendar show types" CLI command to list registered calendar
3409 --------------------------
3410 * Added two new options, r and t with file name arguments to record
3411 single direction (unmixed) audio recording separate from the bidirectional
3412 (mixed) recording. The mixed file name argument is optional now as long
3413 as at least one recording option is used.
3416 --------------------------
3417 * Added a new option, l, which will disable local call optimization for
3418 channels involved with the FollowMe thread. Use this option to improve
3419 compatability for a FollowMe call with certain dialplan apps, options, and
3423 --------------------------
3424 * Added option "k" that will automatically close the conference when there's
3425 only one person left when a user exits the conference.
3428 --------------------------
3429 * cel_pgsql now supports the 'extra' column for data added using the
3430 CELGenUserEvent() application.
3433 --------------------------
3434 * Support for defining hints has been added to pbx_lua. See the 'hints' table
3435 in the sample extensions.lua file for syntax details.
3436 * Applications that perform jumps in the dialplan such as Goto will now
3437 execute properly. When pbx_lua detects that the context, extension, or
3438 priority we are executing on has changed it will immediately return control
3439 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
3440 the priority after the currently executing priority.
3441 * An autoservice is now started by default for pbx_lua channels. It can be
3442 stopped and restarted using the autoservice_stop() and autoservice_start()
3446 --------------------------
3447 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
3448 into a FAXStatus event with an 'Operation' header that will be either
3449 'send', 'receive', and 'gateway'.
3450 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
3451 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
3452 feature will handle converting a fax call between an audio T.30 fax terminal
3453 and an IFP T.38 fax terminal.
3457 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
3458 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
3459 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
3463 * Added general option negative_penalty_invalid default off. when set
3464 members are seen as invalid/logged out when there penalty is negative.
3465 for realtime members when set remove from queue will set penalty to -1.
3466 * Added queue option autopausedelay when autopause is enabled it will be
3467 delayed for this number of seconds since last successful call if there
3468 was no prior call the agent will be autopaused immediately.
3469 * Added member option ignorebusy this when set and ringinuse is not
3470 will allow per member control of multiple calls as ringinuse does for
3475 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
3477 * Added 'k' option to MeetMe to automatically kill the conference when there's only
3478 one participant left (much like a normal call bridge)
3479 * Added extra argument to Originate to set timeout.
3483 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
3484 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
3485 utility in the UTILS section of menuselect. If an existing astdb is found and no
3486 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
3487 convert an existing astdb to the SQLite3 version automatically at runtime.
3491 * Modules marked as deprecated are no longer marked as building by default. Enabling
3492 these modules is still available via menuselect.
3496 * authdebug is now disabled by default. To enable this functionaility again
3497 set authdebug = yes in iax.conf.
3501 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
3502 releases it was disabled.
3506 * The PBX core previously made a call with a non-existing extension test for
3507 extension s@default and jump there if the extension existed.
3508 This was a bad default behaviour and violated the principle of least surprise.
3509 It has therefore been changed in this release. It may affect some
3510 applications and configurations that rely on this behaviour. Most channel
3511 drivers have avoided this for many releases by testing whether the extension
3512 called exists before starting the PBX and generating a local error.
3513 This behaviour still exists and works as before.
3515 Extension "s" is used when no extension is given in a channel driver,
3516 like immediate answer in DAHDI or calling to a domain with no user part
3519 ------------------------------------------------------------------------------
3520 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
3521 ------------------------------------------------------------------------------
3525 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
3526 now defaults to force_rport. It is very important that phones requiring nat=no be
3527 specifically set as such instead of relying on the default setting. If at all
3528 possible, all devices should have nat settings configured in the general section as
3529 opposed to configuring nat per-device.
3530 * Added preferred_codec_only option in sip.conf. This feature limits the joint
3531 codecs sent in response to an INVITE to the single most preferred codec.
3532 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
3533 to be used for the outgoing call. It must be one of the codecs configured
3535 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
3536 to be used for holding a private key. If tlsprivatekey is not specified,
3537 tlscertfile is searched for both public and private key.
3538 * Added tlsclientmethod option to sip.conf. This allows the protocol for
3539 outbound client connections to be specified.
3540 * The sendrpid parameter has been expanded to include the options
3541 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
3542 header to be sent (equivalent to setting sendrpid=yes) and setting
3543 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
3544 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
3545 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
3546 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
3547 will accept the SDP even if the SDP version number is not properly incremented,
3548 but will generate a warning in the log indicating that the SIP peer that sent
3549 the SDP should have the 'ignoresdpversion' option set.
3550 * The 'nat' option has now been been changed to have yes, no, force_rport, and
3551 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
3552 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
3553 remote side requests it and disables symmetric RTP support. Setting it to
3554 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
3555 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
3556 and enables symmetric RTP support.
3557 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
3558 response. This permits the master channel to know how each channel dialled
3559 in a multi-channel setup resolved in an individual way. This carries a
3560 performance penalty and can be disabled in sip.conf using the
3561 'storesipcause' option.
3562 * Added 'externtcpport' and 'externtlsport' options to allow custom port
3563 configuration for the externip and externhost options when tcp or tls is used.
3564 * Added support for message body (stored in content variable) to SIP NOTIFY message
3565 accessible via AMI and CLI.
3566 * Added 'media_address' configuration option which can be used to explicitly specify
3567 the IP address to use in the SDP for media (audio, video, and text) streams.
3568 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
3569 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
3571 * Added 'use_q850_reason' configuration option for generating and parsing
3572 if available Reason: Q.850;cause=<cause code> header. It is implemented
3573 in some gateways for better passing PRI/SS7 cause codes via SIP.
3574 * When dialing SIP peers, a new component may be added to the end of the dialstring
3575 to indicate that a specific remote IP address or host should be used when dialing
3576 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
3577 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
3578 ability to selectively force bridged channels to also be encrypted is also
3579 implemented. Branching in the dialplan can be done based on whether or not
3580 a channel has secure media and/or signaling.
3581 * Added directmediapermit/directmediadeny to limit which peers can send direct media
3583 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
3584 Charge messages to snom phones.
3585 * Added support for G.719 media streams.
3586 * Added support for 16khz signed linear media streams.
3587 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
3588 RTP has been outfitted with the same abilities.
3589 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
3590 available in device configurations as well as in the dial plan.
3591 * Addition of the 'subscribe_network_change' option for turning on and off
3592 res_stun_monitor module support in chan_sip.
3593 * Addition of the 'auth_options_requests' option for turning on and off
3594 authentication for OPTIONS requests in chan_sip.
3598 * Add #tryinclude statement for config files. This provides the same
3599 functionality as the #include statement however an asterisk module will
3600 still load if the filename does not exist. Using the #include statement
3601 Asterisk will not allow the module to load.
3605 * Added rtsavesysname option into iax.conf to allow the systname to be saved
3606 on realtime updates.
3607 * Added the ability for chan_iax2 to inform the dialplan whether or not
3608 encryption is being used. This interoperates with the SIP SRTP implementation
3609 so that a secure SIP call can be bridged to a secure IAX call when the
3610 dialplan requires bridged channels to be "secure".
3611 * Addition of the 'subscribe_network_change' option for turning on and off
3612 res_stun_monitor module support in chan_iax.
3617 * Added ability to preset channel variables on indicated lines with the setvar
3618 configuration option. Also, clearvars=all resets the list of variables back
3620 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
3621 See configs/res_pktccops.conf for more information.
3623 XMPP Google Talk/Jingle changes
3624 -------------------------------
3625 * Added the externip option to gtalk.conf.
3626 * Added the stunaddr option to gtalk.conf which allows for the automatic
3627 retrieval of the external ip from a stun server.
3631 * Added 'p' option to PickupChan() to allow for picking up channel by the first
3632 match to a partial channel name.
3633 * Added .m3u support for Mp3Player application.
3634 * Added progress option to the app_dial D() option. When progress DTMF is
3635 present, those values are sent immediately upon receiving a PROGRESS message
3636 regardless if the call has been answered or not.
3637 * Added functionality to the app_dial F() option to continue with execution
3638 at the current location when no parameters are provided.
3639 * Added the 'a' option to app_dial to answer the calling channel before any
3640 announcements or macros are executed.
3641 * Modified app_dial to set answertime when the called channel answers even if
3642 the called channel hangs up during playback of an announcement.
3643 * Modified app_dial 'r' option to support an additional parameter to play an
3644 indication tone from indications.conf
3645 * Added c() option to app_chanspy. This option allows custom DTMF to be set
3646 to cycle through the next available channel. By default this is still '*'.
3647 * Added x() option to app_chanspy. This option allows DTMF to be set to
3648 exit the application.
3649 * The Voicemail application has been improved to automatically ignore messages
3650 that only contain silence.
3651 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
3652 associated mailbox(es) to be greetings-only.
3653 * The ChanSpy application now has the 'S' option, which makes the application
3654 automatically exit once it hits a point where no more channels are available
3656 * The ChanSpy application also now has the 'E' option, which spies on a single
3657 channel and exits when that channel hangs up.
3658 * The MeetMe application now turns on the DENOISE() function by default, for
3659 each participant. In our tests, this has significantly decreased background
3660 noise (especially noisy data centers).
3661 * Voicemail now permits storage of secrets in a separate file, located in the
3662 spool directory of each individual user. The control for this is located in
3663 the "passwordlocation" option in voicemail.conf. Please see the sample
3664 configuration for more information.
3665 * The ChanIsAvail application now exposes the returned cause code using a separate
3666 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
3667 * Added 'd' option to app_followme. This option disables the "Please hold"
3669 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
3670 received will terminate recording.
3671 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
3672 Previously the folder could only be set per context, but has now been extended
3673 using the imapfolder option.
3674 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
3675 * Voicemail now allows the pager date format to be specified separately from the
3677 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
3678 to allow joining, leaving, and sending text to group chats.
3679 * MeetMe has a new option 'G' to play an announcement before joining a conference.
3680 * Page has a new option 'A(x)' which will playback an announcement simultaneously
3681 to all paged phones (and optionally excluding the caller's one using the new
3682 option 'n') before the call is bridged.
3683 * The 'f' option to Dial has been augmented to take an optional argument. If no
3684 argument is provided, the 'f' option works as it always has. If an argument is
3685 provided, then the connected party information of all outgoing channels created
3686 during the Dial will be set to the argument passed to the 'f' option.
3687 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
3689 * The OSP lookup application adds in/outbound network ID, optional security,
3690 number portability, QoS reporting, destination IP port, custom info and service
3692 * Added new application VMSayName that will play the recorded name of the voicemail
3693 user if it exists, otherwise will play the mailbox number.
3694 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
3695 retrieve state for a particular bridge, where <name> is the conference name
3696 * app_directory now allows exiting at any time using the operator or pound key.
3697 * Voicemail now supports setting a locale per-mailbox.
3698 * Two new applications are provided for declining counting phrases in multiple
3699 languages. See the application notes for SayCountedNoun and SayCountedAdj for
3701 * Voicemail now runs the externnotify script when pollmailboxes is activated and
3703 * Voicemail now includes rdnis within msgXXXX.txt file.
3704 * ExternalIVR now supports IPv6 addresses.
3705 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
3706 at https://wiki.asterisk.org/wiki/x/oQBB
3707 * ParkedCall and Park can now specify the parking lot to use.
3711 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
3712 over SRV records associated with a specific service. From the CLI, type
3713 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
3714 details on how these may be used.
3715 * PITCH_SHIFT dialplan function added. This function can be used to modify the
3716 pitch of a channel's tx and rx audio streams.
3717 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
3718 setting various connected line and redirecting party information.
3719 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
3720 support ISDN subaddressing.
3721 * The CHANNEL() function now supports the "name" and "checkhangup" options.
3722 * For DAHDI channels, the CHANNEL() dialplan function now allows
3723 the dialplan to request changes in the configuration of the active
3724 echo canceller on the channel (if any), for the current call only.
3727 exten => s,n,Set(CHANNEL(echocan_mode)=off)
3729 The possible values are:
3731 on - normal mode (the echo canceller is actually reinitialized)
3733 fax - FAX/data mode (NLP disabled if possible, otherwise completely
3735 voice - voice mode (returns from FAX mode, reverting the changes that
3736 were made when FAX mode was requested)
3737 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
3738 and setting variables on the channel which created the current channel.
3739 Administrators should take care to avoid naming conflicts, when multiple
3740 channels are dialled at once, especially when used with the Local channel
3741 construct (which all could set variables on the master channel). Usage
3742 of the HASH() dialplan function, with the key set to the name of the slave
3743 channel, is one approach that will avoid conflicts.
3744 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
3746 * func_odbc now allows multiple row results to be retrieved without using
3747 mode=multirow. If rowlimit is set, then additional rows may be retrieved
3748 from the same query by using the name of the function which retrieved the
3749 first row as an argument to ODBC_FETCH().
3750 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
3751 dialplan. This function returns the content of the received message.
3752 * Added REPLACE, which searches a given variable name for a set of characters,
3753 then either replaces them with a single character or deletes them.
3754 * Added PASSTHRU, which literally passes the same argument back as its return
3755 value. The intent is to be able to use a literal string argument to
3756 functions that currently require a variable name as an argument.
3757 * HASH-associated variables now can be inherited across channel creation, by
3758 prefixing the name of the hash at assignment with the appropriate number of
3759 underscores, just like variables.
3760 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
3761 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
3762 whether or not channels that are bridged to the current channel will be
3763 required to have secure signaling and/or media.
3764 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
3765 the current channel has secure signaling and/or media.
3766 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
3767 "no_media_path" option.
3768 Returns "0" if there is a B channel associated with the call.
3769 Returns "1" if no B channel is associated with the call. The call is either
3770 on hold or is a call waiting call.
3771 * Added option to dialplan function CDR(), the 'f' option
3772 allows for high resolution times for billsec and duration fields.
3773 * FILE() now supports line-mode and writing.
3774 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
3775 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
3779 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
3780 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
3781 and is set when a dynamic feature is triggered.
3782 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
3783 to dynamically create a new parking lot matching the value this varible is
3785 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
3786 features.conf that should be the base for dynamic parkinglots.
3787 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
3788 parkinglot should have.
3789 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
3790 parkinglot should have.
3791 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
3796 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
3797 timeout has expired.
3798 * Added 'R' option to app_queue. This option stops moh and indicates ringing
3799 to the caller when an Agent's phone is ringing. This can be used to indicate
3800 to the caller that their call is about to be picked up, which is nice when
3801 one has been on hold for an extened period of time.
3802 * A new config option, penaltymemberslimit, has been added to queues.conf.
3803 When set this option will disregard penalty settings when a queue has too
3805 * A new option, 'I' has been added to both app_queue and app_dial.
3806 By setting this option, Asterisk will not update the caller with
3807 connected line changes or redirecting party changes when they occur.
3808 * A 'relative-periodic-announce' option has been added to queues.conf. When
3809 enabled, this option will cause periodic announce times to be calculated
3810 from the end of announcements rather than from the beginning.
3811 * The autopause option in queues.conf can be passed a new value, "all." The
3812 result is that if a member becomes auto-paused, he will be paused in all
3813 queues for which he is a member, not just the queue that failed to reach
3815 * Added dialplan function QUEUE_EXISTS to check if a queue exists
3816 * The queue logger now allows events to optionally propagate to a file,
3817 even when realtime logging is turned on. Additionally, realtime logging
3818 supports sending the event arguments to 5 individual fields, although it
3819 will fallback to the previous data definition, if the new table layout is
3822 mISDN channel driver (chan_misdn) changes
3823 ----------------------------------------
3824 * Added display_connected parameter to misdn.conf to put a display string
3825 in the CONNECT message containing the connected name and/or number if
3826 the presentation setting permits it.
3827 * Added display_setup parameter to misdn.conf to put a display string
3828 in the SETUP message containing the caller name and/or number if the
3829 presentation setting permits it.
3830 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
3831 indicate the dialplan settings are to be obtained from the asterisk
3833 * Made misdn.conf parameter callerid accept the "name" <number> format
3834 used by the rest of the system.
3835 * Made use the nationalprefix and internationalprefix misdn.conf
3836 parameters to prefix any received number from the ISDN link if that
3837 number has the corresponding Type-Of-Number. NOTE: This includes
3838 comparing the incoming call's dialed number against the MSN list.
3839 * Added the following new parameters: unknownprefix, netspecificprefix,
3840 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
3841 received number from the ISDN link if that number has the corresponding
3843 * Added new dialplan application misdn_command which permits controlling
3844 the CCBS/CCNR functionality.
3845 * Added new dialplan function mISDN_CC which permits retrieval of various
3846 values from an active call completion record.
3847 * For PTP, you should manually send the COLR of the redirected-to party
3848 for an incomming redirected call if the incoming call could experience
3849 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
3850 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
3851 if the REDIRECTING(from-num) is not empty.
3852 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
3853 option on all of the REDIRECTING statements before dialing the
3854 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
3855 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
3856 redirecting-to presentation (COLR) when it becomes available.
3857 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
3860 thirdparty mISDN enhancements
3861 -----------------------------
3862 mISDN has been modified by Digium, Inc. to greatly expand facility message
3864 * Enhanced COLP support for call diversion and transfer.
3865 * CCBS/CCNR support.
3867 The latest modified mISDN v1.1.x based version is available at:
3868 http://svn.digium.com/svn/thirdparty/mISDN/trunk
3869 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
3871 Tagged versions of the modified mISDN code are available under:
3872 http://svn.digium.com/svn/thirdparty/mISDN/tags
3873 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
3875 libpri channel driver (chan_dahdi) DAHDI changes
3876 -------------------------------------------
3877 * The channel variable PRIREDIRECTREASON is now just a status variable
3878 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
3879 to read and alter the reason.
3880 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
3881 redirected-to party for an incomming redirected call if the incoming call
3882 could experience further redirects. Just set the
3883 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
3884 to the COLR. A call has been redirected if the REDIRECTING(count) is not
3886 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
3887 use the inhibit(i) option on all of the REDIRECTING statements before
3888 dialing the redirected-to party. You still have to set the
3889 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
3890 will update the redirecting-to presentation (COLR) when it becomes available.
3891 * Added the ability to ignore calls that are not in a Multiple Subscriber
3892 Number (MSN) list for PTMP CPE interfaces.
3893 * Added dynamic range compression support for dahdi channels. It is
3894 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
3895 * Added support for ISDN calling and called subaddress with partial support
3896 for connected line subaddress.
3897 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
3898 * Added handling of received HOLD/RETRIEVE messages and the optional ability
3899 to transfer a held call on disconnect similar to an analog phone.
3900 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
3901 Will reroute/deflect an outgoing call when receive the message.
3902 Can use the DAHDISendCallreroutingFacility to send the message for the
3904 * Added standard location to add options to chan_dahdi dialing:
3905 Dial(DAHDI/g1[/extension[/options]])
3908 R Reverse charging indication
3909 * Added Reverse Charging Indication (Collect calls) send/receive option.
3910 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
3911 Dial(DAHDI/g1/extension/R)
3912 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
3913 (requires latest LibPRI)
3914 * Added ability to send/receive keypad digits in the SETUP message.
3915 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
3916 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
3917 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
3918 (requires latest LibPRI)
3919 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
3920 to eliminate tromboned calls. A tromboned call goes out an interface and comes
3921 back into the same interface. Tromboned calls happen because of call routing,
3922 call deflection, call forwarding, and call transfer.
3923 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
3924 * Added the ability to support call waiting calls. (The SETUP has no B channel
3926 * Added Malicious Call ID (MCID) event to the AMI call event class.
3927 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
3929 Asterisk Manager Interface
3930 --------------------------
3931 * The Hangup action now accepts a Cause header which may be used to
3932 set the channel's hangup cause.
3933 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
3934 to specify a separate .pem file to hold a private key. By default sslcert
3935 is used to hold both the public and private key.
3936 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
3937 for options containing the 'tls' prefix. For example, 'sslenable' is now
3938 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
3939 across all .conf files. All affected sample.conf files have been modified to
3940 reflect this change. Previous options such as 'sslenable' still work,
3941 but options with the 'tls' prefix are preferred.
3942 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
3943 in a channel. (res_mutestream.so)
3944 * The configuration file manager.conf now supports a channelvars option, which
3945 specifies a list of channel variables to include in each channel-oriented
3947 * The redirect command now has new parameters ExtraContext, ExtraExtension,
3948 and ExtraPriority to allow redirecting the second channel to a different
3949 location than the first.
3950 * Added new event "JabberStatus" in the Jabber module to monitor buddies
3952 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
3953 in a MixMonitor recording.
3954 * The 'iax2 show peers' output is now similar to the expected output of
3956 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
3958 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
3959 AOC-E messages on a channel.
3960 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
3961 conform more closely to similar events.
3962 * Added a new eventfilter option per user to allow whitelisting and blacklisting
3964 * Added optional parkinglot variable for park command.
3965 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
3966 if CallerIDNum and CallerIDName headers are also present.
3968 Channel Event Logging
3969 ---------------------
3970 * A new interface, CEL, is introduced here. CEL logs single events, much like
3971 the AMI, but it differs from the AMI in that it logs to db backends much
3972 like CDR does; is based on the event subsystem introduced by Russell, and
3973 can share in all its benefits; allows multiple backends to operate like CDR;
3974 is specialized to event data that would be of concern to billing sytems,
3975 like CDR. Backends for logging and accounting calls have been produced,
3976 but a new CDR backend is still in development.
3980 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
3981 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
3982 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
3983 * Multiple files and formats can now be specified in cdr_custom.conf.
3984 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
3985 See configs/cdr_syslog.conf.sample for more information.
3986 * A 'sequence' field has been added to CDRs which can be combined with
3987 linkedid or uniqueid to uniquely identify a CDR.
3988 * Handling of billsec and duration field has changed. If your table definition
3989 specifies those fields as float,double or similar they will now be logged with
3990 microsecond accuracy instead of a whole integer.
3992 Calendaring for Asterisk
3993 ------------------------
3994 * A new set of modules were added supporing calendar integration with Asterisk.
3995 Dialplan functions for reading from and writing to calendars are included,
3996 as well as the ability to execute dialplan logic upon calendar event notifications.
3997 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
3998 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
3999 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
4000 2003 support does not support forms-based authentication).
4002 Call Completion Supplementary Services for Asterisk
4003 ---------------------------------------------------
4004 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
4005 DAHDI/ISDN supports call completion for the following switch types:
4006 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
4007 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
4009 Multicast RTP Support
4010 ---------------------
4011 * A new RTP engine and channel driver have been added which supports Multicast RTP.
4012 The channel driver can be used with the Page application to perform multicast RTP
4013 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
4014 Type can be either basic or linksys.
4015 Destination is the IP address and port for the RTP packets.
4016 Control address is specific to the linksys type and is used for sending the control
4017 packets unique to them.
4019 Security Events Framework
4020 -------------------------
4021 * Asterisk has a new C API for reporting security events. The module res_security_log
4022 sends these events to the "security" logger level. Currently, AMI is the only
4023 Asterisk component that reports security events. However, SIP support will be
4024 coming soon. For more information on the security events framework, see the
4025 "Asterisk Security Framework" section of the Asterisk wiki at
4026 https://wiki.asterisk.org/wiki/x/wgBQ
4027 * SIP support was added in Asterisk 10
4028 * This API now supports IPv6 addresses
4032 * A technology independent fax frontend (res_fax) has been added to Asterisk.
4033 * A spandsp based fax backend (res_fax_spandsp) has been added.
4034 * The app_fax module has been deprecated in favor of the res_fax module and
4035 the new res_fax_spandsp backend.
4036 * The SendFAX and ReceiveFAX applications now send their log messages to a
4037 'fax' logger level, instead of to the generic logger levels. To see these
4038 messages, the system's logger.conf file will need to direct the 'fax' logger
4039 level to one or more destinations; the logger.conf.sample file includes an
4040 example of how to do this. Note that if the 'fax' logger level is *not*
4041 directed to at least one destination, log messages generated by these
4042 applications will be lost, and that if the 'fax' logger level is directed to
4043 the console, the 'core set verbose' and 'core set debug' CLI commands will
4044 have no effect on whether the messages appear on the console or not.
4048 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
4049 Now, in order to enable transmitting silence during record the transmit_silence
4050 option should be used. transmit_silence_during_record remains a valid option, but
4051 defaults to the behavior of the transmit_silence option.
4052 * Addition of the Unit Test Framework API for managing registration and execution
4053 of unit tests with the purpose of verifying the operation of C functions.
4054 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
4055 XMPP text messages to the remote JID.
4056 * Modules.conf has a new option - "require" - that marks a module as critical for
4057 the execution of Asterisk.
4058 If one of the required modules fail to load, Asterisk will exit with a return
4060 * An 'X' option has been added to the asterisk application which enables #exec support.
4061 This allows #exec to be used in asterisk.conf.
4062 * jabber.conf supports a new option auth_policy that toggles auto user registration.
4063 * A new lockconfdir option has been added to asterisk.conf to protect the
4064 configuration directory (/etc/asterisk by default) during reloads.
4065 * The parkeddynamic option has been added to features.conf to enable the creation
4066 of dynamic parkinglots.
4067 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
4068 the reportalarms config option.
4069 * chan_dahdi supports dialing configuring and dialing by device file name.
4070 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
4071 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
4072 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
4073 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
4074 Handy for the above name-based syntax as it does not depend on
4075 initialization order.
4076 * The Realtime dialplan switch now caches entries for 1 second. This provides a
4077 significant increase in performance (about 3X) for installations using this switchtype.
4078 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
4079 AIS. For more information, please see the Distributed Device State section of the
4080 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
4081 * The addition of G.719 pass-through support.
4082 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
4083 during device configuration.
4084 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
4085 have less than 3 lines on the LCD.
4086 * Realtime now supports database failover. See the sample extconfig.conf for details.
4087 * The addition of improved translation path building for wideband codecs. Sample
4088 rate changes during translation are now avoided unless absolutely necessary.
4089 * The addition of the res_stun_monitor module for monitoring and reacting to network
4090 changes while behind a NAT.
4091 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
4092 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
4093 These allow support for any Administration. Default is AT&T values.
4097 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
4098 optionally accept a filename, to apply the setting only to the code generated from
4099 that source file when Asterisk was built. However, there are some modules in Asterisk
4100 that are composed of multiple source files, so this did not result in the behavior
4101 that users expected. In this version, 'core set debug' and 'core set verbose'
4102 can optionally accept *module* names instead (with or without the .so extension),
4103 which applies the setting to the entire module specified, regardless of which source
4104 files it was built from.
4105 * New 'manager show settings' command showing the current settings loaded from
4107 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
4108 the channel hangup request to all channels.
4109 * Added a "core reload" CLI command that executes a global reload of Asterisk.
4111 ------------------------------------------------------------------------------
4112 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
4113 ------------------------------------------------------------------------------
4117 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
4118 Snom phones use this for call pickup of extensions that the phone is
4120 * Added support for setting the domain in the URI for caller of an
4121 outbound call by using the SIPFROMDOMAIN channel variable.
4122 * Added a new configuration option "remotesecret" for authentication to
4123 remote services. For backwards compatibility, "secret" still has the
4124 same function as before, but now you can configure both a remote secret and a
4125 local secret for mutual authentication.
4126 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
4127 the sound will be played to the target of an attended transfer
4128 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
4129 finer control over how many peers Asterisk will qualify and the gap between them
4130 when all peers need to be qualified at the same time.
4131 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
4132 (either globally or for a specific peer), chan_sip will treat any SDP data
4133 it receives as new data and update the media stream accordingly. By
4134 default, Asterisk will only modify the media stream if the SDP session
4135 version received&nbs
projects / asterisk/asterisk.git / blob