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 * Remote files can now be retrieved and played back. See the Playback
37 dialplan application for more details.
41 * Remote files can now be retrieved and played back via the Playback and other
42 media playback dialplan applications. This is done by directly providing
43 the URL to play to the dialplan application:
44 same => n,Playback(http://1.1.1.1/howler-monkeys-fl.wav)
45 Note that unlike 'normal' media files, the entire URI to the file must be
46 provided, including the file extension. Currently, on HTTP and HTTPS URI
47 schemes are supported.
51 * Added the 'n' option, which prevents the SMS from being written to the log
52 file. This is needed for those countries with privacy laws that require
53 providers to not log SMS content.
60 * Added a new configuration option, "newcdrcolumns", which enables use of the
61 post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.
66 * Added a new configuration option, "newcdrcolumns", which enables use of the
67 post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.
75 * The CALLERID(ani2) value for incoming calls is now populated in featdmf
76 signaling mode. The information was previously discarded.
77 * Added the force_restart_unavailable_chans compatibility option. When
78 enabled it causes Asterisk to restart the ISDN B channel if an outgoing
79 call receives cause 44 (Requested channel not available).
83 * The iax.conf forcejitterbuffer option has been removed. It is now always
84 forced if you set iax.conf jitterbuffer=yes. If you put a jitter buffer
85 on a channel it will be on the channel.
86 * A new configuration parameters, 'calltokenexpiration', has been added that
87 controls the duration before a call token expires. Default duration is 10
88 seconds. Setting this to a higher value may help in lagged networks or those
89 experiencing high packet loss.
93 * New 'rtpbindaddr' global setting. This allows a user to define which
94 ipaddress to bind the rtpengine to. For example, chan_sip might bind
95 to eth0 (10.0.0.2) but rtpengine to eth1 (192.168.1.10).
96 * DTLS related configuration options can now be set at a general level.
97 Enabling DTLS support, though, requires enabling it at the user
99 * Added the possibility to set the From: header through the the SIP dial
100 string (populating the fromuser/fromdomain fields), complementing the
101 [!dnid] option for the To: header that has existed since 1.6.0 (1d6b192).
102 NOTE: This is again separated by an exclamation mark, so the To: header may
103 not contain one of those.
107 * New 'user_eq_phone' endpoint setting. This adds a 'user=phone' parameter
108 to the request URI and From URI if the user is determined to be a phone number.
109 * New 'moh_passthrough' endpoint setting. This will pass hold and unhold requests
110 through using SIP re-invites with sendonly and sendrecv accordingly.
111 * Added the pjsip.conf system type disable_tcp_switch option. The option
112 allows the user to disable switching from UDP to TCP transports described
113 by RFC 3261 section 18.1.1.
114 * New 'line' and 'endpoint' options added on outbound registrations. This allows some
115 identifying information to be added to the Contact of the outbound registration.
116 If this information is present on messages received from the remote server
117 the message will automatically be associated with the configured endpoint on the
118 outbound registration.
122 * The core of Asterisk uses a message bus called "Stasis" to distribute
123 information to internal components. For performance reasons, the message
124 distribution was modified to make use of a thread pool instead of a
125 dedicated thread per consumer in certain cases. The initial settings for
126 the thread pool can now be configured in 'stasis.conf'.
128 * A new core DNS API has been implemented which provides a common interface
129 for DNS functionality. Modules that use this functionality will require that
130 a DNS resolver module is loaded and available.
132 * Modified processing of command-line options to first parse only what
133 is necessary to read asterisk.conf. Once asterisk.conf is fully loaded,
134 the remaining options are processed. The -X option now applies to
135 asterisk.conf only. To enable #exec for other config files you must
136 set execincludes=yes in asterisk.conf. Any other option set on the
137 command-line will now override the equivalent setting from asterisk.conf.
139 * The TLS core in Asterisk now supports X.509 certificate subject alternative
140 names. This way one X.509 certificate can be used for hosts that can be
141 reached under multiple DNS names or for multiple hosts.
143 * The Asterisk logging system now supports JSON structured logging. Log
144 channels specified in logger.conf or added dynamically via CLI commands now
145 support an optional specifier prior to their levels that determines their
146 formatting. To set a log channel to format its entries as JSON, a formatter
147 of '[json]' can be set, e.g.,
148 full => [json]debug,verbose,notice,warning,error
150 * The core now supports a 'media cache', which stores temporary media files
151 retrieved from external sources. CLI commands have been added to manipulate
152 and display the cached files, including:
153 - 'media cache show <all>' - show all cached media files, or details about
154 one particular cached media file
155 - 'media cache refresh <item>' - force a refresh of a particular media file
157 - 'media cache delete <item>' - remove an item from the cache
158 - 'media cache create <uri>' - retrieve a URI and store it in the cache
166 * Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for
167 the hold status of a channel.
171 * The CURL function now supports a write option, which will save the retrieved
172 file to a location on disk. As an example:
173 same => n,Set(CURL(https://1.1.1.1/foo.wav)=/tmp/foo.wav)
174 will save 'foo.wav' to /tmp.
178 * The transferdialattempts default value has been changed from 1 to 3. The
179 transferinvalidsound has been changed from "pbx-invalid" to "privacy-incorrect".
180 These were changed to make DTMF transfers be more user-friendly by default.
188 * A backend for the core media cache, this module retrieves media files from
189 a remote HTTP(S) server and stores them in the core media cache for later
194 * Added sort=randstart to the sort options. It sorts the files by name and
195 then chooses the first file to play at random.
196 * Added preferchannelclass=no option to prefer the application-passed class
197 over the channel-set musicclass. This allows separate hold-music from
198 application (e.g. Queue or Dial) specified music.
202 * Added a res_resolver_unbound module which uses the libunbound resolver library
203 to perform DNS resolution. This module requires the libunbound library to be
204 installed in order to be used.
208 * A new SIP resolver using the core DNS API has been implemented. This relies on
209 external SIP resolver support in PJSIP which is only available as of PJSIP
210 2.4. If this support is unavailable the existing built-in PJSIP SIP resolver
211 will be used instead. The new SIP resolver provides NAPTR support, improved
212 SRV support, and AAAA record support.
214 res_pjsip_outbound_registration
215 -------------------------------
216 * A new 'fatal_retry_interval' option has been added to outbound registration.
217 When set (default is zero), and upon receiving a failure response to an
218 outbound registration, registration is retried at the given interval up to
226 * Added a new option, 'usegmtime', which causes timestamps in CEL events
229 * Added support to set schema where located the table cel. This settings is
230 configurable for cel_pgsql via the 'schema' in configuration file
238 * Added the ability to set the character to quote identifiers. This
239 allows adding the character at the start and end of table and column
240 names. This setting is configurable for cdr_adaptive_odbc via the
241 quoted_identifiers in configuration file cdr_adaptive_odbc.conf.
245 * Added field ReasonPause on QueueMemberStatus if set when paused, the reason
246 the queue member was paused.
247 * Added field LastPause on QueueMemberStatus for time when started the last
248 pause for a queue member.
249 * Show the time when started the last pause for queue member on CLI for command
253 ------------------------------------------------------------------------------
254 --- Functionality changes from Asterisk 13.7.0 to Asterisk 13.8.0 ------------
255 ------------------------------------------------------------------------------
259 * Per RFC3325, the 'From' header is now anonymized on outgoing calls when
260 caller id presentation is prohibited.
262 res_pjsip_config_wizard
264 * A new command (pjsip export config_wizard primitives) has been added that
265 will export all the pjsip objects it created to the console or a file
266 suitable for reuse in a pjsip.conf file.
270 * To help insure that Asterisk is compiled and run with the same known
271 version of pjproject, a new option (--with-pjproject-bundled) has been
272 added to ./configure. When specified, the version of pjproject specified
273 in third-party/versions.mak will be downloaded and configured. When you
274 make Asterisk, the build process will also automatically build pjproject
275 and Asterisk will be statically linked to it. Once a particular version
276 of pjproject is configured and built, it won't be configured or built
277 again unless you run a 'make distclean'.
279 To facilitate testing, when 'make install' is run, the pjsua and pjsystest
280 utilities and the pjproject python bindings will be installed in
281 ASTDATADIR/third-party/pjproject.
283 The default behavior remains building with the shared pjproject
284 installation, if any.
288 * Added CONFBRIDGE_INFO(muted,) for querying the muted conference state.
290 * Added Muted header to AMI ConfbridgeListRooms action response list events
291 to indicate the muted conference state.
293 * Added Muted column to CLI "confbridge list" output to indicate the muted
294 conference state and made the locked column a yes/no value instead of a
295 locked/unlocked value.
299 * The REDIRECTING(reason) value is now treated consistently between
300 chan_sip and chan_pjsip.
302 Both channel drivers match incoming reason values with values documented
303 by REDIRECTING(reason) and values documented by RFC5806 regardless of
304 whether they are quoted or not. RFC5806 values are mapped to the
305 equivalent REDIRECTING(reason) documented value and is set in
306 REDIRECTING(reason). e.g., an incoming RFC5806 'unconditional' value or a
307 quoted string version ('"unconditional"') is converted to
308 REDIRECTING(reason)'s 'cfu' value. The user's dialplan only needs to deal
309 with 'cfu' instead of any of the aliases.
311 The incoming 480 response reason text supported by chan_sip checks for
312 known reason values and if not matched then puts quotes around the reason
313 string and assigns that to REDIRECTING(reason).
315 Both channel drivers send outgoing known REDIRECTING(reason) values as the
316 unquoted RFC5806 equivalent. User custom values are either sent as is or
317 with added quotes if SIP doesn't allow a character within the value as
318 part of a RFC3261 Section 25.1 token. Note that there are still
319 limitations on what characters can be put in a custom user value. e.g.,
320 embedding quotes in the middle of the reason string is just going to cause
323 * Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases.
324 e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the
329 * This module is the successor of res_pjsip_log_forwarder. As well as
330 handling the log forwarding (which now displays as 'pjproject:0' instead
331 of 'pjsip:0'), it also adds a 'pjproject show buildopts' command to the CLI.
332 This displays the compiled-in options of the pjproject installation
333 Asterisk is currently running against.
335 * Another feature of this module is the ability to map pjproject log levels
336 to Asterisk log levels, or to suppress the pjproject log messages
337 altogether. Many of the messages emitted by pjproject itself are the result
338 of errors which Asterisk will ultimately handle so the messages can be
339 misleading or just noise. A new config file (pjproject.conf) has been added
340 to configure the mapping and a new CLI command (pjproject show log mappings)
341 has been added to display the mappings currently in use.
345 * Transports are now reloadable. In testing, no in-progress calls were
346 disrupted if the ip address or port weren't changed, but the possibility
347 still exists. To make sure there are no unintentional drops, a new option
348 'allow_reload', which defaults to 'no' has been added to transport. If
349 left at the default, changes to the particular transport will be ignored.
350 If set to 'yes', changes (if any) will be applied.
352 * Added new global option (regcontext) to pjsip. When set, Asterisk will
353 dynamically create and destroy a NoOp priority 1 extension
354 for a given endpoint who registers or unregisters with us.
358 * A new module, res_pjsip_history, has been added that provides SIP history
359 viewing/filtering from the CLI. The module is intended to be used on systems
360 with busy SIP traffic, where existing forms of viewing SIP messages - such
361 as the res_pjsip_logger - may be inadequate. The module provides two new
363 - 'pjsip set history {on|off|clear}' - this enables/disables SIP history
364 capturing, as well as clears an existing history capture. Note that SIP
365 packets captured are stored in memory until cleared. As a result, the
366 history capture should only be used for debugging/viewing purposes, and
367 should *NOT* be left permanently enabled on a system.
368 - 'pjsip show history' - displays the captured SIP history. When invoked
369 with no options, the entire captured history is displayed. Two options
371 -- 'entry <num>' - display a detailed view of a single SIP message in
373 -- 'where ...' - filter the history based on some expression. For more
374 information on filtering, view the current CLI help for the
375 'pjsip show history' command.
379 * app_voicemail and res_mwi_external can now be built together. The default
380 remains to build app_voicemail and not res_mwi_external but if they are
381 both built, the load order will cause res_mwi_external to load first and
382 app_voicemail will be skipped. Use 'preload=app_voicemail.so' in
383 modules.conf to force app_voicemail to be the voicemail provider.
387 * A new option (bind_rtp_to_media_address) has been added to endpoint which
388 will cause res_pjsip_sdp_rtp to actually bind the RTP instance to the
389 media_address as well as using it in the SDP. If set, RTP packets will now
390 originate from the media address instead of the operating system's "primary"
395 * A new configuration section - ice_host_candidates - has been added to
396 rtp.conf, allowing automatically discovered ICE host candidates to be
397 overriden. This allows an Asterisk server behind a 1:1 NAT to send its
398 external IP as a host candidate rather than relying on STUN to discover it.
400 ------------------------------------------------------------------------------
401 --- Functionality changes from Asterisk 13.6.0 to Asterisk 13.7.0 ------------
402 ------------------------------------------------------------------------------
406 * Added format attribute negotiation for the VP8 video codec. Format attribute
407 negotiation is provided by the res_format_attr_vp8 module.
411 * A new "timeout" user profile option has been added. This configures the number
412 of seconds that a participant may stay in the ConfBridge after joining. When
413 the time expires, the user is ejected from the conference and CONFBRIDGE_RESULT
414 is set to "TIMEOUT" on the channel.
418 * The websockets_enabled option has been added to the general section of
419 sip.conf. The option is enabled by default to match the previous behavior.
420 The option should be disabled when using res_pjsip_transport_websockets to
421 ensure chan_sip will not conflict with PJSIP websockets.
425 * The HOLD_INTERCEPT dialplan function now actually exists in the source tree.
426 While support for the events was added in Asterisk 13.4.0, the function
427 accidentally never made it in. That function is now present, and will cause
428 the 'hold' raised by a channel to be intercepted and converted into an
431 res_pjsip_outbound_registration
432 -------------------------------
433 * If res_statsd is loaded and a StatsD server is configured, basic statistics
434 regarding the state of outbound registrations will now be emitted. This
436 - A GAUGE statistic for the overall number of outbound registrations, i.e.:
437 PJSIP.registrations.count
438 - A GAUGE statistic for the overall number of outbound registrations in a
439 particular state, e.g.:
440 PJSIP.registrations.state.Registered
444 * The ability to use "like" has been added to the pjsip list and show
445 CLI commands. For instance: CLI> pjsip list endpoints like abc
447 * If res_statsd is loaded and a StatsD server is configured, basic statistics
448 regarding the state of PJSIP contacts will now be emitted. This includes:
449 - A GAUGE statistic for the overall number of contacts in a particular
451 PJSIP.contacts.states.Reachable
452 - A TIMER statistic for the RTT time for each qualified contact, e.g.:
453 PJSIP.contacts.alice@@127.0.0.1:5061.rtt
455 res_sorcery_memory_cache
456 ------------------------
457 * A new caching strategy, full_backend_cache, has been added which caches
458 all stored objects in the backend. When enabled all objects will be
459 expired or go stale according to the configuration. As well when enabled
460 all retrieval operations will be performed against the cache instead of
465 * CALLERID(pres) is now documented as a valid alternative to setting both
466 CALLERID(name-pres) and CALLERID(num-pres) at once. Some channel drivers,
467 like chan_sip, don't make a distinction between the two: they take the
468 least public value from name-pres and num-pres. By using CALLERID(pres)
469 for reading and writing, you touch the same combined value in the dialplan.
470 The same applies to CONNECTEDLINE(pres), REDIRECTING(orig-pres),
471 REDIRECTING(to-pres) and REDIRECTING(from-pres).
475 * A new module that emits StatsD statistics regarding Asterisk endpoints.
476 This includes a total count of the number of endpoints, the count of the
477 number of endpoints in the technology agnostic state of the endpoint -
478 online or offline - as well as the number of channels associated with each
479 endpoint. These are recorded as three different GAUGE statistics:
481 - endpoints.state.{unknown|offline|online}
482 - endpoints.{tech}.{resource}.channels
485 ------------------------------------------------------------------------------
486 --- Functionality changes from Asterisk 13.5.0 to Asterisk 13.6.0 ------------
487 ------------------------------------------------------------------------------
491 * The CHANNEL function, when used on a PJSIP channel, now exposes a 'call-id'
492 extraction option when using with the 'pjsip' signalling option. It will
493 return the SIP Call-ID associated with the INVITE request that established
498 * Two new endpoint related events are now available: PeerStatusChange and
499 ContactStatusChange. In particular, these events are useful when subscribing
500 to all event sources, as they provide additional endpoint related
501 information beyond the addition/removal of channels from an endpoint.
503 * Added the ability to subscribe to all ARI events in Asterisk, regardless
504 of whether the application 'controls' the resource. This is useful for
505 scenarios where an ARI application merely wants to observe the system,
506 as opposed to control it. There are two ways to accomplish this:
507 (1) Via the WebSocket connection URI. A new query paramter, 'subscribeAll',
508 has been added that, when present and True, will subscribe all
509 specified applications to all ARI event sources in Asterisk.
510 (2) Via the applications resource. An ARI client can, at any time, subscribe
511 to all resources in an event source merely by not providing an explicit
512 resource. For example, subscribing to an event source of 'channels:'
513 as opposed to 'channels:12345' will subscribe the application to all
516 ------------------------------------------------------------------------------
517 --- Functionality changes from Asterisk 13.4.0 to Asterisk 13.5.0 ------------
518 ------------------------------------------------------------------------------
522 * A new ContactStatus event has been added that reflects res_pjsip contact
523 lifecycle changes: Created, Removed, Reachable, Unreachable, Unknown.
525 * Added the Linkedid header to the common channel headers listed for each
526 channel in AMI events.
530 * A new feature has been added that enables the retrieval of modules and
531 module information through an HTTP request. Information on a single module
532 can be also be retrieved. Individual modules can be loaded to Asterisk, as
533 well as unloaded and reloaded.
535 * A new resource has been added to the 'asterisk' resource, 'config/dynamic'.
536 This resource allows for push configuration of sorcery derived objects
537 within Asterisk. The resource supports creation, retrieval, updating, and
538 deletion. Sorcery derived objects that are manipulated by this resource
539 must have a sorcery wizard that supports the desired operations.
541 * A new feature has been added that allows for the rotation of log channels
542 through HTTP requests.
547 * A new 'g726_non_standard' endpoint option has been added that, when set to
548 'yes' and g.726 audio is negotiated, forces the codec to be treated as if it
549 is AAL2 packed on the channel.
551 * A new 'rtp_keepalive' endpoint option has been added. This option specifies
552 an interval, in seconds, at which we will send RTP comfort noise packets to
553 the endpoint. This functions identically to chan_sip's "rtpkeepalive" option.
555 * New 'rtp_timeout' and 'rtp_timeout_hold' endpoint options have been added.
556 These options specify the amount of time, in seconds, that Asterisk will wait
557 before terminating the call due to lack of received RTP. These are identical
558 to chan_sip's rtptimeout and rtpholdtimeout options.
560 ------------------------------------------------------------------------------
561 --- Functionality changes from Asterisk 13.3.0 to Asterisk 13.4.0 ------------
562 ------------------------------------------------------------------------------
566 * New 'rpid_immediate' option to control if connected line update information
567 goes to the caller immediately or waits for another reason to send the
568 connected line information update. See the online option documentation for
569 more information. Defaults to 'no' as setting it to 'yes' can result in
570 many unnecessary messages being sent to the caller.
572 * The configuration setting 'progressinband' now defaults to 'no', which
573 matches the actual behavior of previous versions.
577 * A new CLI command has been added: "pjsip show settings", which shows
578 both the global and system configuration settings.
580 * A new aor option has been added: "qualify_timeout", which sets the timeout
581 in seconds for a qualify. The default is 3 seconds. This overrides the
582 hard coded 32 seconds in pjproject.
584 * Endpoint status will now change to "Unreachable" when all contacts are
585 unavailable. When any contact becomes available, the endpoint will status
586 will change back to "Reachable".
588 * A new global option has been added: "max_initial_qualify_time", which
589 sets the maximum amount of time from startup that qualifies should be
590 attempted on all contacts.
594 * Two new events, 'ChannelHold' and 'ChannelUnhold', have been added to the
595 events data model. These events are raised when a channel indicates a hold
596 or unhold, respectively.
600 * A new dialplan function, HOLD_INTERCEPT, has been added. This function, when
601 placed on a channel, intercepts hold/unhold indications signalled by the
602 channel and prevents them from moving on to other channels in a bridge with
603 the hold initiator. Instead, AMI or ARI events are raised indicating that
604 the channel wanted to place someone on hold. This allows external
605 applications to implement their own custom hold/unhold logic.
607 ------------------------------------------------------------------------------
608 --- Functionality changes from Asterisk 13.2.0 to Asterisk 13.3.0 ------------
609 ------------------------------------------------------------------------------
611 chan_pjsip/app_transfer
613 * The Transfer application, when used with chan_pjsip, now supports using
614 a PJSIP endpoint as the transfer destination. This is in addition to
615 explicitly specifying a SIP URI to transfer to.
619 * The ARI /channels resource now supports a new operation, 'redirect'. The
620 redirect operation will perform a technology and state specific redirection
621 on the channel to a specified endpoint or destination. In the case of SIP
622 technologies, this is either a 302 Redirect response to an on-going INVITE
623 dialog or a SIP REFER request.
627 * A new 'endpoint_identifier_order' option has been added that allows one to
628 set the order by which endpoint identifiers are processed and checked. This
629 option is specified under the 'global' type configuration section.
631 ------------------------------------------------------------------------------
632 --- Functionality changes from Asterisk 13.1.0 to Asterisk 13.2.0 ------------
633 ------------------------------------------------------------------------------
635 * New 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions have been added which
636 allow examining PJSIP AORs or contacts from the dialplan.
638 res_pjsip_outbound_registration
640 * The 'pjsip send unregister' command now stops further registrations.
642 * A new command 'pjsip send register' has been added which allows you to
643 start or restart periodic registration. It can be used after a
644 'send unregister' or after a 401 permanent error.
646 res_pjsip_config_wizard
648 * This is a new module that adds streamlined configuration capability for
649 chan_pjsip. It's targeted at users who have lots of basic configuration
650 scenarios like 'phone' or 'agent' or 'trunk'. Additional information
651 can be found in the sample configuration file at
652 config/samples/pjsip_wizard.conf.sample.
656 * The T.38 negotiation timeout was previously hard coded at 5000 milliseconds
657 and is now configurable via the 't38timeout' configuration option in
658 res_fax.conf and via the fax options dialplan function 'FAXOPT(t38timeout)'.
659 The default remains at 5000 milliseconds.
663 * The ca_list_path transport parameter has been added for TLS transports. This
664 option behaves similarly to the old sip.conf option "tlscapath". In order to
665 use this, you must be using PJProject version 2.4 or higher.
669 * The Originate operation now takes in an originator channel. The linked ID of
670 this originator channel is applied to the newly originated outgoing channel.
671 If using CEL this allows an association to be established between the two so
672 it can be recognized that the originator is dialing the originated channel.
674 * "language" (the default spoken language for the channel) is now included in
675 the standard channel state output for suitable events.
677 * The POST channels/{id} operation and the POST channels/{id}/continue operation
678 now have a new "label" parameter. This allows for origination or continuation
679 to a labeled priority in the dialplan instead of requiring a specific priority
680 number. The ARI version has been bumped to 1.7.0 as a result.
684 * "Language" (the default spoken language for the channel) is now included in
685 the standard channel state output for suitable events.
687 * AMI actions that return a list of events have been made to return consistent
688 headers for the action response event starting the list and the list complete
689 event. The AMI version has been bumped to 2.7.0 as a result.
691 ------------------------------------------------------------------------------
692 --- Functionality changes from Asterisk 13.0.0 to Asterisk 13.1.0 ------------
693 ------------------------------------------------------------------------------
697 * Event NewConnectedLine is emitted when the connected line information on
702 * Event ChannelConnectedLine is emitted when the connected line information
703 on a channel changes.
708 The features.conf general section has three new configurable options:
709 * transferdialattempts
711 * transferinvalidsound
712 For more information on what these options do, see the Asterisk wiki:
713 https://wiki.asterisk.org/wiki/x/W4fAAQ
720 * New 'media_encryption_optimistic' endpoint setting. This will use SRTP
721 when possible but does not consider lack of it a failure.
723 res_pjsip_endpoint_identifer_ip
725 * New CLI commands have been added: "pjsip show identif(y|ies)", which lists
726 all configured PJSIP identify objects
728 ------------------------------------------------------------------------------
729 --- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
730 ------------------------------------------------------------------------------
735 Asterisk 13 is the next Long Term Support (LTS) release of Asterisk. As such,
736 the focus of development for this release of Asterisk was on improving the
737 usability and features developed in the previous Standard release, Asterisk 12.
738 Beyond a general refinement of end user features, development focussed heavily
739 on the Asterisk APIs - the Asterisk Manager Interface (AMI) and the Asterisk
740 REST Interface (ARI) - and the PJSIP stack in Asterisk. Some highlights of the
741 new features include:
743 * Asterisk security events are now provided via AMI, allowing end users to
744 monitor their Asterisk system in real time for security related issues.
745 * External control of Message Waiting Indicators (MWI) through both AMI and ARI.
746 * Reception/transmission of out of call text messages using any supported
747 channel driver/protocol stack through ARI.
748 * Resource List Server support in the PJSIP stack, providing subscriptions to
749 lists of resources and batched delivery of NOTIFY requests.
750 * Inter-Asterisk distributed device state and mailbox state using the PJSIP
753 It is important to note that Asterisk 13 is built on the architecture developed
754 during the previous Standard release, Asterisk 12. Users upgrading to
755 Asterisk 13 should read about the new features in Asterisk 12 later in this file
756 (see Functionality changes from Asterisk 11 to Asterisk 12), as well as the
757 UPGRADE-12.txt delivered with this release. In particular, users upgrading to
758 Asterisk 13 from a release prior to Asterisk 12 should read the specifications
759 on AMI, CDRs, and CEL on the Asterisk wiki:
760 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
761 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
762 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
764 Many new featuers in Asterisk 13 were introduced in point releases of
765 Asterisk 12. Following this section - which documents the changes from all
766 versions of Asterisk 12 to Asterisk 13 - users should examine the new features
767 that were introduced in the point releases of Asterisk 12, as they are also
768 included in Asterisk 13.
770 Finally, all users upgrading to Asterisk 13 should read the UPGRADE.txt file
771 delivered with this release.
776 * Sample config files have been moved from configs/ to a sub-folder of that
779 * The menuselect utility has been pulled into the Asterisk repository. As a
780 result, the libxml2 development library is now a required dependency for
783 * A new Compiler Flag, REF_DEBUG, has been added. When enabled, reference
784 counted objects will emit additional debug information to the refs log file
785 located in the standard Asterisk log file directory. This log file is useful
786 in tracking down object leaks and other reference counting issues. Prior to
787 this version, this option was only available by modifying the source code
788 directly. This change also includes a new script, refcounter.py, in the
789 contrib folder that will process the refs log file. Note that this replaces
790 the refcounter utility that could be built from the utils directory.
798 * This module was deprecated and has been removed. Users of app_dahdibarge
799 should use ChanSpy instead.
803 * New options to play a beep when starting a recording and stopping a recording
804 have been added. The option "p" will play a beep to the channel that starts
805 the recording. The option "P" will play a beep to the channel that stops the
810 * Queue rules can now be stored in a database table, queue_rules. Unlike other
811 RealTime tables, the queue_rules table is only examined on module load or
812 module reload. A new general setting has been added to queuerules.conf,
813 'realtime_rules', which, when set to 'yes', will cause app_queue to look in
814 RealTime for additional queue rules to parse. Note that both the file and
815 the database can be used as a provide of queue rules when 'realtime_rules'
818 When app_queue is reloaded, all rules are re-parsed and loaded into memory.
819 There is no caching of RealTime queue rules.
823 * This module was deprecated and has been removed. Users of app_readfile
824 should use func_env's FILE function instead.
828 * The 'say' family of dialplan applications now support the Japanese
829 language. The 'language' parameter in say.conf now recognizes a setting of
830 'ja', which will enable Japanese language specific mechanisms for playing
831 back numbers, dates, and other items.
835 * This module was deprecated and has been removed. Users of app_saycountpl
836 should use the Say family of applications.
840 * The SetMusicOnHold dialplan application was deprecated and has been removed.
841 Users of the application should use the CHANNEL function's musicclass
846 * The WaitMusicOnHold dialplan application was deprecated and has been
847 removed. Users of the application should use MusicOnHold with a duration
852 * VoiceMail and VoiceMailMain now support the Japanese language. The
853 'language' parameter in voicemail.conf now recognizes a setting of 'ja',
854 which will enable prompts to be played back using a Japanese grammatical
855 structure. Additional prompts are necessary for this functionality,
857 - jb-arimasu: there is
858 - jb-arimasen: there is not
859 - jb-oshitekudasai: please press
865 * Add the ability to specify multiple email addresses in configuration,
874 * This module was deprecated and has been removed. Users of cdr_sqlite
875 should use cdr_sqlite3_custom.
879 * Added the ability to support PostgreSQL application_name on connections.
880 This allows PostgreSQL to display the configured name in the
881 pg_stat_activity view and CSV log entries. This setting is configurable
882 for cdr_pgsql via the appname configuration setting in cdr_pgsql.conf.
890 * Added the ability to support PostgreSQL application_name on connections.
891 This allows PostgreSQL to display the configured name in the
892 pg_stat_activity view and CSV log entries. This setting is configurable
893 for cel_pgsql via the appname configuration setting in cel_pgsql.conf.
901 * SS7 support now requires libss7 v2.0 or later.
903 * Added SS7 support for connected line and redirecting.
905 * Most SS7 CLI commands are reworked as well as new SS7 commands added.
908 * Added several SS7 config option parameters described in
909 chan_dahdi.conf.sample.
913 * This module was deprecated and has been removed. Users of chan_gtalk
914 should use chan_motif.
918 * This module was deprecated and has been removed. Users of chan_h323
919 should use chan_ooh323.
923 * This module was deprecated and has been removed. Users of chan_jingle
924 should use chan_motif.
928 * Added the CLI command 'pjsip list ciphers' so a user can know what
929 OpenSSL names are available on their system for the pjsip.conf cipher
934 * The SIPPEER dialplan function no longer supports using a colon as a
935 delimiter for parameters. The parameters for the function should be
936 delimited using a comma.
938 * The SIPCHANINFO dialplan function was deprecated and has been removed. Users
939 of the function should use the CHANNEL function instead.
947 * Added functional peeraccount support. Except for Queue, the
948 accountcode propagation is now consistently propagated to outgoing
949 channels before dialing. The channel accountcode can change from its
950 original non-empty value on channel creation for the following specific
951 reasons. One, dialplan sets it using CHANNEL(accountcode). Two, an
952 originate method that can specify an accountcode value. Three, the
953 calling channel propagates its peeraccount or accountcode to the
954 outgoing channel's accountcode before dialing. The change has two
955 visible effects. One, local channels now cross accountcode and
956 peeraccount across the special bridge between the ;1 and ;2 channels
957 just like channels between normal bridges. Two, the
958 CHANNEL(peeraccount) value can now be set before Dial and FollowMe to
959 set the accountcode on the outgoing channel(s).
961 For Queue, an outgoing channel's non-empty accountcode will not change
962 unless explicitly set by CHANNEL(accountcode). The change has three
963 visible effects. One, local channels now cross accountcode and
964 peeraccount across the special bridge between the ;1 and ;2 channels
965 just like channels between normal bridges. Two, the queue member will
966 get an accountcode if it doesn't have one and one is available from the
967 calling channel's peeraccount. Three, accountcode propagation includes
968 local channel members where the accountcodes are propagated early
969 enough to be available on the ;2 channel.
973 * New DeviceStateChanged and PresenceStateChanged AMI events have been added.
974 These events are emitted whenever a device state or presence state change
975 occurs. The events are controlled by res_manager_device_state.so and
976 res_manager_presence_state.so. If the high frequency of these events is
977 problematic for you, do not load these modules.
979 * Added DialplanExtensionAdd and DialplanExtensionRemove AMI commands. They
980 work in basically the same way as the 'dialplan add extension' and
981 'dialplan remove extension' CLI commands respectively.
983 * New AMI action LoggerRotate reloads and rotates logger in the same manner
984 as CLI command 'logger rotate'
986 * New AMI Actions FAXSessions, FAXSession, and FAXStats replicate the
987 functionality of CLI commands 'fax show sessions', 'fax show session',
988 and fax show stats' respectively.
990 * New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset
991 enable manager control over PRI debugging levels and file output.
993 * AMI action PJSIPNotify may now send to a URI instead of only to a PJSIP
994 endpoint as long as a default outbound endpoint is set. This also applies
995 to the equivalent CLI command (pjsip send notify)
997 * The AMI action PJSIPShowEndpoint now includes ContactStatusDetail sections
998 that give information on Asterisk's attempts to qualify the endpoint.
1000 * The DialEnd event will now contain a Forward header if the dial is ending
1001 due to the call being forwarded. The contents of the Forward header is the
1002 extension in the number to which the call is being forwarded.
1006 * The "bridge_technology" extra field key has been added to BRIDGE_ENTER
1007 and BRIDGE_EXIT events.
1011 * Channel variables are now substituted in arguments passed to applications
1012 run by using dynamic features.
1016 * The TLS core in Asterisk now supports Perfect Forward Secrecy (PFS).
1017 Enabling PFS is attempted by default, and is dependent on the configuration
1018 of the module using TLS.
1019 - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
1020 specify a ECDHE cipher suite in sip.conf, for example:
1021 tlscipher=AES128-SHA:DES-CBC3-SHA
1022 - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
1023 into the private key file, e.g., sip.conf tlsprivatekey. For example, the
1024 default dh2048.pem - see
1025 http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt
1026 - Because clients expect the server to prefer PFS, and because OpenSSL sorts
1027 its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
1028 Consider re-ordering your cipher suites in the respective configuration
1030 tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
1031 will use PFS when offered by the client. Clients which do not offer PFS
1032 fall-back to AES-128 (or even 3DES, as recommended by RFC 3261).
1040 * The JACK_HOOK function now supports audio with a sample rate higher than
1049 * Added the ability to support PostgreSQL application_name on connections.
1050 This allows PostgreSQL to display the configured name in the
1051 pg_stat_activity view and CSV log entries. This setting is configurable
1052 for res_config_pgsql via the dbappname configuration setting in
1055 res_pjsip_outbound_publish
1057 * A new module, res_pjsip_outbound_publish provides the mechanisms for sending
1058 PUBLISH requests for specific event packages to another SIP User Agent.
1062 * The publish/subscribe core module has been updated to support RFC 4662
1063 Resource Lists, allowing Asterisk to act as a Resource List Server (RLS).
1064 Resource lists are configured in pjsip.conf under a new object type,
1065 resource_list. Resource lists can contain either message-summary or presence
1066 events, and can be composed of specific resources that provide the event or
1067 other resource lists.
1069 * Inbound publication support is provided by a new object, inbound-publication.
1070 This configures res_pjsip_pubsub to accept PUBLISH requests from a particular
1071 resource. Which events are accepted is constructed dynamically; see
1072 res_pjsip_publish_asterisk for more information.
1074 res_pjsip_publish_asterisk
1076 * A new module, res_pjsip_publish_asterisk adds support for PUBLISH requests of
1077 Asterisk information to other Asterisk servers. This module is intended only
1078 for Asterisk to Asterisk exchanges of information. Currently, this includes
1079 both mailbox state and device state information.
1081 ------------------------------------------------------------------------------
1082 --- Functionality changes from Asterisk 12.4.0 to Asterisk 12.5.0 ------------
1083 ------------------------------------------------------------------------------
1087 * Stored recordings now support a new operation, copy. This will take an
1088 existing stored recording and copy it to a new location in the recordings
1091 * LiveRecording objects now have three additional fields that can be reported
1092 in a RecordingFinished ARI event:
1093 - total_duration: the duration of the recording
1094 - talking_duration: optional. The duration of talking detected in the
1095 recording. This is only available if max_silence_seconds was specified
1096 when the recording was started.
1097 - silence_duration: optional. The duration of silence detected in the
1098 recording. This is only available if max_silence_seconds was specified
1099 when the recording was started.
1100 Note that all duration values are reported in seconds.
1102 * Users of ARI can now send and receive out of call text messages. Messages
1103 can be sent directly to a particular endpoint, or can be sent to the
1104 endpoints resource directly and inferred from the URI scheme. Text
1105 messages are passed to ARI clients as TextMessageReceived events. ARI
1106 clients can choose to receive text messages by subscribing to the particular
1107 endpoint technology or endpoints that they are interested in.
1109 * The applications resource now supports subscriptions to all endpoints of
1110 a particular channel technology. For example, subscribing to an eventSource
1111 of 'endpoint:PJSIP' will subscribe to all PJSIP endpoints.
1115 * The endpoint configuration object now supports 'accountcode'. Any channel
1116 created for an endpoint with this setting will have its accountcode set
1117 to the specified value.
1121 * A new module, res_hep_rtcp, has been added that will forward RTCP call
1122 statistics to a HEP capture server. See res_hep for more information.
1126 * Function AUDIOHOOK_INHERIT has been deprecated. Audiohooks are now
1127 unconditionally inhereted through masquerades. As a side benefit, more
1128 than one audiohook of a given type may persist through a masquerade now.
1130 ------------------------------------------------------------------------------
1131 --- Functionality changes from Asterisk 12.3.0 to Asterisk 12.4.0 ------------
1132 ------------------------------------------------------------------------------
1136 * Returns new AGENT_STATUS value "NOT_CONNECTED" if the agent fails to
1137 connect with an incoming caller after being alerted to the presence
1138 of the incoming caller. The most likely reason this would happen is
1139 the agent did not acknowledge the call in time.
1143 * New events have been added for the TALK_DETECT function. When the function
1144 is used on a channel, ChannelTalkingStart/ChannelTalkingStop events will be
1145 emitted to connected AMI clients indicating the start/stop of talking on
1150 * New event models have been aded for the TALK_DETECT function. When the
1151 function is used on a channel, ChannelTalkingStarted/ChannelTalkingFinished
1152 events will be emitted to connected WebSockets subscribed to the channel,
1153 indicating the start/stop of talking on the channel.
1157 * A new function, TALK_DETECT, has been added. When set on a channel, this
1158 fucntion causes events indicating the starting/stoping of talking on said
1159 channel to be emitted to both AMI and ARI clients.
1161 ------------------------------------------------------------------------------
1162 --- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------
1163 ------------------------------------------------------------------------------
1167 * A new Playback URI 'tone' has been added. Tones are specified either as
1168 an indication name (e.g. 'tone:busy') from indications.conf or as a tone
1169 pattern (e.g. 'tone:240/250,0/250'). Tones differ from normal playback
1170 URIs in that they must be stopped manually and will continue to occupy
1171 a channel's ARI control queue until they are stopped. They also can not
1172 be rewound or fastforwarded.
1174 * User events can now be generated from ARI. Events can be signalled with
1175 arbitrary json variables, and include one or more of channel, bridge, or
1176 endpoint snapshots. An application must be specified which will receive
1177 the event message (other applications can subscribe to it). The message
1178 will also be delivered via AMI provided a channel is attached. Dialplan
1179 generated user event messages are still transmitted via the channel, and
1180 will only be received by a stasis application they are attached to or if
1181 the channel is subscribed to.
1185 * SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI
1186 fields for prohibited callingpres information. Values are legacy, no, and
1187 yes. By default, legacy is used.
1188 trust_id_outbound=legacy - behavior remains the same as 1.8.26.1. When
1189 dealing with prohibited callingpres and sendrpid=pai/rpid, RPID/PAI
1190 headers are appended to outbound SIP messages just as they are with
1191 allowed callingpres values, but data about the remote party's identity is
1193 When sendrpid=rpid, only the remote party's domain is anonymized.
1194 trust_id_outbound=no - when dealing with prohibited callingpres, RPID/PAI
1195 headers are not sent.
1196 trust_id_outbound=yes - RPID/PAI headers are applied with the full remote
1197 party information in tact even for prohibited callingpres information.
1198 In the case of PAI, a Privacy: id header will be appended for prohibited
1199 calling information to communicate that the private information should
1200 not be relayed to untrusted parties.
1204 * Manager action 'Park' now takes an additional argument 'AnnounceChannel'
1205 which can be used to announce the parked call's location to an arbitrary
1206 channel in a bridge. If 'Channel' and 'TimeoutChannel' are now the two
1207 parties in a one to one bridge, 'TimeoutChannel' is treated as having
1208 parked 'Channel' like with the Park Call DTMF feature and will receive
1209 announcements prior to being hung up.
1211 ------------------------------------------------------------------------------
1212 --- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
1213 ------------------------------------------------------------------------------
1217 * Record application now has an option 'o' which allows 0 to act as an exit
1218 key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'
1221 --------------------------
1222 * ChanSpy now accepts a channel uniqueid or a fully specified channel name
1223 as the chanprefix parameter if the 'u' option is specified.
1226 --------------------------
1227 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
1228 conference user menus.
1230 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
1231 menus, bridge settings, and user settings that have been applied by the
1232 CONFBRIDGE dialplan function.
1234 * The ConfBridge dialplan application now sets a channel variable,
1235 CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
1236 how a channel exited the conference.
1238 * Added conference user option 'announce_join_leave_review'. This option
1239 implies 'announce_join_leave' with the added effect that the user will
1240 be asked if they want to confirm or re-record the recording of their
1241 name when entering the conference
1244 --------------------------
1245 * At exit, the Directory application now sets a channel variable
1246 DIRECTORY_RESULT to one of the following based on the reason for exiting:
1247 OPERATOR user requested operator by pressing '0' for operator
1248 ASSISTANT user requested assistant by pressing '*' for assistant
1249 TIMEOUT user pressed nothing and Directory stopped waiting
1250 HANGUP user's channel hung up
1251 SELECTED user selected a user from the directory and is routed
1252 USEREXIT user pressed '#' from the selection prompt to exit
1253 FAILED directory failed in a way that wasn't accounted for. Dang.
1257 * Monitor() - A new option, B(), has been added that will turn on a periodic
1258 beep while the call is being recorded.
1261 --------------------------
1262 * MusicOnHold streams (all modes other than "files") now support wide band
1266 --------------------------
1267 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
1268 and for the channel executing Page respectively.
1271 --------------------------
1272 * PickupChan now accepts channel uniqueids of channels to pickup.
1275 --------------------------
1276 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
1277 to 'true' (case insensitive), then any Say application (SayNumber,
1278 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
1279 anticipate DTMF. If DTMF is received, these applications will behave like
1280 the background application and jump to the received extension once a match
1281 is established or after a short period of inactivity.
1284 -------------------------
1285 * A new function, MIXMONITOR, has been added to allow access to individual
1286 instances of MixMonitor on a channel.
1288 * A new option, B(), has been added that will turn on a periodic beep while the
1289 call is being recorded.
1293 -------------------------
1296 -------------------------
1297 * TEL URI support for inbound INVITE requests has been added. chan_sip will
1298 now handle TEL schemes in the Request and From URIs. The phone-context in
1299 the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on
1300 the inbound channel.
1304 * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
1305 the new AST_SORCERY diaplan function.
1307 * Core Show Locks output now includes Thread/LWP ID if the platform
1308 supports this feature.
1310 * New "logger add channel" and "logger remove channel" CLI commands have
1311 been added to allow creation and deletion of dynamic logger channels
1312 without configuration changes. These dynamic logger channels will only
1313 exist until the next restart of asterisk.
1317 * The live recording object on recording events now contains a target_uri
1318 field which contains the URI of what is being recorded.
1320 * The bridge type used when creating a bridge is now a comma separated list of
1321 bridge properties. Valid options are: mixing, holding, dtmf_events, and
1324 * A channelId can now be provided when creating a channel, either in the
1325 uri (POST channels/my-channel-id) or as query parameter. A local channel
1326 will suffix the second channel id with ';2' unless provided as query
1327 parameter otherChannelId.
1329 * A bridgeId can now be provided when creating a bridge, either in the uri
1330 (POST bridges/my-bridge-id) or as a query parameter.
1332 * A playbackId can be provided when starting a playback, either in the uri
1333 (POST channels/my-channel-id/play/my-playback-id /
1334 POST bridges/my-bridge-id/play/my-playback-id) or as a query parameter.
1336 * A snoop channel can be started with a snoopId, in the uri or query.
1340 * Originate now takes optional parameters ChannelId and OtherChannelId,
1341 used to set the UniqueId on creation. The other id is assigned to the
1342 second channel when dialing LOCAL, or defaults to appending ;2 if only
1343 the single Id is given.
1345 * The Mixmonitor action now has a "Command" header that can be used to
1346 indicate a post-process command to run once recording finishes.
1350 * A new set of Alembic scripts has been added for CDR tables. This will create
1351 a 'cdr' table with the default schema that Asterisk expects.
1356 * A new function was added: PERIODIC_HOOK. This allows running a periodic
1357 dialplan hook on a channel. Any audio generated by this hook will be
1358 injected into the call.
1366 * A new module, res_hep, has been added, that acts as a generic packet
1367 capture agent for the Homer Encapsulation Protocol (HEP) version 3.
1368 It can be configured via hep.conf. Other modules can use res_hep to send
1369 message traffic to a HEP capture server.
1373 * A new module, res_hep_pjsip, has been added that will forward PJSIP
1374 message traffic to a HEP capture server. See res_hep for more
1379 * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
1380 be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
1382 * Added the following new CLI commands:
1383 - "pjsip show contacts" - list all current PJSIP contacts.
1384 - "pjsip show contact" - show specific information about a current PJSIP
1386 - "pjsip show channel" - show detailed information about a PJSIP channel.
1388 res_pjsip_multihomed
1390 * A new module, res_pjsip_multihomed handles situations where the system
1391 Asterisk is running out has multiple interfaces. res_pjsip_multihomed
1392 determines which interface should be used during message sending.
1394 res_pjsip_pidf_digium_body_supplement
1396 * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
1397 request body formatting for presence support in Digium phones.
1399 res_pjsip_send_to_voicemail
1401 * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
1402 particular headers to transfer a PJSIP channel directly to a particular
1403 extension that has VoiceMail. This is intended to be used with Digium
1404 phones that support this feature.
1406 res_pjsip_outbound_registration
1408 * A new CLI command has been added: "pjsip show registrations", which lists
1409 all configured PJSIP registrations
1412 ------------------------------------------------------------------------------
1413 --- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
1414 ------------------------------------------------------------------------------
1418 * Added a new module that provides AMI control over MWI within Asterisk,
1419 res_mwi_external_ami. Note that this module depends on res_mwi_external;
1420 for more information on enabling this module, see res_mwi_external.
1421 This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
1422 the MWIGet/MWIGetComplete events.
1424 * The DialStatus field in the DialEnd event can now contain additional
1425 statuses that convey how the dial operation terminated. This includes
1426 ABORT, CONTINUE, and GOTO.
1428 * AMI will now emit security events. A new class authorization has been
1429 added in manager.conf for the security events, 'security'. The new events
1431 - FailedACL - raised when a request violates an ACL check
1432 - InvalidAccountID - raised when a request fails an authentication
1433 check due to an invalid account ID
1434 - SessionLimit - raised when a request fails due to exceeding the
1435 number of allowed concurrent sessions for a service
1436 - MemoryLimit - raised when a request fails due to an internal memory
1438 - LoadAverageLimit - raised when a request fails because a configured
1439 load average limit has been reached
1440 - RequestNotAllowed - raised when a request is not allowed by
1442 - AuthMethodNotAllowed - raised when a request used an authentication
1443 method not allowed by the service
1444 - RequestBadFormat - raised when a request is received with bad formatting
1445 - SuccessfulAuth - raised when a request successfully authenticates
1446 - UnexpectedAddress - raised when a request has a different source address
1447 then what is expected for a session already in progress with a service
1448 - ChallengeResponseFailed - raised when a request's attempt to authenticate
1449 has been challenged, and the request failed the authentication challenge
1450 - InvalidPassword - raised when a request provides an invalid password
1451 during an authentication attempt
1452 - ChallengeSent - raised when an Asterisk service send an authentication
1453 challenge to a request
1454 - InvalidTransport - raised when a request attempts to use a transport not
1455 allowed by the Asterisk service
1457 * Bridge related events now have two additional fields: BridgeName and
1458 BridgeCreator. BridgeName is a descriptive name for the bridge;
1459 BridgeCreator is the name of the entity that created the bridge. This
1460 affects the following events: ConfbridgeStart, ConfbridgeEnd,
1461 ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
1462 ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
1463 AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
1467 * The Bridge data model now contains the additional fields 'name' and
1468 'creator'. The 'name' field conveys a descriptive name for the bridge;
1469 the 'creator' field conveys the name of the entity that created the bridge.
1470 This affects all responses to HTTP requests that return a Bridge data model
1471 as well as all event derived data models that contain a Bridge data model.
1472 The POST /bridges operation may now optionally specify a name to give to
1473 the bridge being created.
1475 * Added a new ARI resource 'mailboxes' which allows the creation and
1476 modification of mailboxes managed by external MWI. Modules res_mwi_external
1477 and res_stasis_mailbox must be enabled to use this resource. For more
1478 information on external MWI control, see res_mwi_external.
1480 * Added new events for externally initiated transfers. The event
1481 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
1482 of a bridge in the ARI controlled application to the dialplan; the
1483 BridgeAttendedTransfer event is raised when a channel initiates an
1484 attended transfer of a bridge in the ARI controlled application to the
1487 * Channel variables may now be specified as a body parameter to the
1488 POST /channels operation. The 'variables' key in the JSON is interpreted
1489 as a sequence of key/value pairs that will be added to the created channel
1490 as channel variables. Other parameters in the JSON body are treated as
1491 query parameters of the same name.
1495 * Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be
1496 automatically handled by the HTTP server if a request is received with a
1497 Transfer-Encoding type of "chunked".
1501 * Path support has been added with the 'support_path' option in registration
1504 * A 'debug' option has been added to the globals section that will allow
1505 sip messages to be logged.
1507 * A 'set_var' option has been added to endpoints that will automatically
1508 set the desired variable(s) on a channel created for that endpoint.
1510 * Several new tables and columns have been added to the realtime schema for
1511 the res_pjsip related modules. See the UPGRADE.txt notes for updating
1512 the database schema.
1516 * A new module, res_mwi_external, has been added to Asterisk. This module
1517 acts as a base framework that other modules can build on top of to allow
1518 an external system to control MWI within Asterisk. For implementations
1519 that make use of res_mwi_external, see res_mwi_external_ami and
1520 res_ari_mailboxes. Note that res_mwi_external canflicts with other modules
1521 that may produce MWI themselves, such as app_voicemail. res_mwi_external
1522 and other modules that depend on it cannot be built or loaded with
1523 app_voicemail present.
1527 * DNS functionality will now automatically be enabled if the system configured
1528 nameservers can be retrieved. If the system configured nameservers can not be
1529 retrieved the functionality will resort to using system resolution. Functionalty
1530 such as SRV records and failover will not be available if system resolution
1533 ------------------------------------------------------------------------------
1534 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
1535 ------------------------------------------------------------------------------
1540 Asterisk 12 is a standard release of the Asterisk project. As such, the
1541 focus of development for this release was on core architectural changes and
1542 major new features. This includes:
1543 * A more flexible bridging core based on the Bridging API
1544 * A new internal message bus, Stasis
1545 * Major standardization and consistency improvements to AMI
1546 * Addition of the Asterisk RESTful Interface (ARI)
1547 * A new SIP channel driver, chan_pjsip
1548 In addition, as the vast majority of bridging in Asterisk was migrated to the
1549 Bridging API used by ConfBridge, major changes were made to most of the
1550 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
1552 Specifications have been written for the affected interfaces. These
1553 specifications are available on the Asterisk wiki:
1554 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
1555 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
1556 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
1558 It is *highly* recommended that anyone migrating to Asterisk 12 read the
1559 information regarding its release both in this file and in the accompanying
1560 UPGRADE.txt file. More detailed information on the major changes can be found
1561 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
1566 * Added build option DISABLE_INLINE. This option can be used to work around a
1567 bug in gcc. For more information, see
1568 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
1570 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
1571 the CHANNEL_TRACE build option were incompatible with the new bridging
1574 * Asterisk now optionally uses libxslt to improve XML documentation generation
1575 and maintainability. If libxslt is not available on the system, some XML
1576 documentation will be incomplete.
1578 * Asterisk now depends on libjansson. If a package of libjansson is not
1579 available on your distro, please see http://www.digip.org/jansson/.
1581 * Asterisk now depends on libuuid and, optionally, uriparser. It is
1582 recommended that you install uriparser, even if it is optional.
1584 * The new SIP stack and channel driver uses a particular version of PJSIP.
1585 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
1586 configuring and installing PJSIP for usage with Asterisk.
1588 * Optional API was re-implemented to be more portable, and no longer requires
1589 weak reference support from the compiler. The build option OPTIONAL_API may
1590 be disabled to disable Optional API support.
1597 * Along with AgentRequest, this application has been modified to be a
1598 replacement for chan_agent. The act of a channel calling the AgentLogin
1599 application places the channel into a pool of agents that can be
1600 requested by the AgentRequest application. Note that this application, as
1601 well as all other agent related functionality, is now provided by the
1602 app_agent_pool module. See chan_agent and AgentRequest for more information.
1604 * This application no longer performs agent authentication. If authentication
1605 is desired, the dialplan needs to perform this function using the
1606 Authenticate or VMAuthenticate application or through an AGI script before
1609 * If this application is called and the agent is already logged in, the
1610 dialplan will continue exection with the AGENT_STATUS channel variable set
1611 to ALREADY_LOGGED_IN.
1613 * The agents.conf schema has changed. Rather than specifying agents on a
1614 single line in comma delineated fashion, each agent is defined in a separate
1615 context. This allows agents to use the power of context templates in their
1618 * A number of parameters from agents.conf have been removed. This includes
1619 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
1620 urlprefix, and savecallsin. These options were obsoleted by the move from
1621 a channel driver model to the bridging/application model provided by
1626 * A new application, this will request a logged in agent from the pool and
1627 bridge the requested channel with the channel calling this application.
1628 Logged in agents are those channels that called the AgentLogin application.
1629 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
1630 application will be set with an appropriate error value.
1632 AgentMonitorOutgoing
1634 * This application has been removed. It was a holdover from when
1635 AgentCallbackLogin was removed.
1639 * Added support for additional Ademco DTMF signalling formats, including
1640 Express 4+1, Express 4+2, High Speed and Super Fast.
1642 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
1643 call time, in milliseconds, to run the application.
1645 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
1646 maximum number of times to retry the call.
1648 * Added a new configuration option answait. If set, the AlarmReceiver
1649 application will wait the number of milliseconds specified by answait
1650 after the channel has answered. Valid values range between 500
1651 milliseconds and 10000 milliseconds.
1653 * Added configuration option no_group_meta. If enabled, grouping of metadata
1654 information in the AlarmReceiver log file will be skipped.
1658 * It is now no longer possible to bypass updating the CDR on the channel
1659 when answering. CDRs reflect the state of the channel and will always
1660 reflect the time they were Answered.
1664 * A new application in Asterisk, this will place the calling channel
1665 into a holding bridge, optionally entertaining them with some form of
1666 media. Channels participating in a holding bridge do not interact with
1667 other channels in the same holding bridge. Optionally, however, a channel
1668 may join as an announcer. Any media passed from an announcer channel is
1669 played to all channels in the holding bridge. Channels leave a holding
1670 bridge either when an optional timer expires, or via the ChannelRedirect
1671 application or AMI Redirect action.
1675 * All participants in a bridge can now be kicked out of a conference room
1676 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
1677 command, i.e., 'confbridge kick <conference> all'
1679 * CLI output for the 'confbridge list' command has been improved. When
1680 displaying information about a particular bridge, flags will now be shown
1681 for the participating users indicating properties of that user.
1683 * The ConfbridgeList event now contains the following fields: WaitMarked,
1684 EndMarked, and Waiting. This displays additional properties about the
1685 user's profile, as well as whether or not the user is waiting for a
1686 Marked user to enter the conference.
1688 * Added a new option for conference recording, record_file_append. If enabled,
1689 when the recording is stopped and then re-started, the existing recording
1690 will be used and appended to.
1692 * ConfBridge now has the ability to set the language of announcements to the
1693 conference. The language can be set on a bridge profile in confbridge.conf
1694 or by the dialplan function CONFBRIDGE(bridge,language)=en.
1698 * The channel variable CPLAYBACKSTATUS may now return the value
1699 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
1700 such as AMI. See the AMI action ControlPlayback for more information.
1704 * Added the 'a' option, which allows the caller to enter in an additional
1705 alias for the user in the directory. This option must be used in conjunction
1706 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
1707 specified in voicemail.conf.
1711 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
1712 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
1713 containing the unique ID of the bridge that the channel happens to be in.
1717 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
1718 for more information.
1720 * Variables are no longer purged from the original CDR. See the 'v' option for
1723 * The 'A' option has been removed. The Answer time on a CDR is never updated
1726 * The 'd' option has been removed. The disposition on a CDR is a function of
1727 the state of the channel and cannot be altered.
1729 * The 'D' option has been removed. Who the Party B is on a CDR is a function
1730 of the state of the respective channels involved in the CDR and cannot be
1733 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
1734 such that the start time and, if applicable, the answer time was updated.
1735 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
1736 'r' option now triggers the Reset, setting the start time (and answer time
1737 if applicable) to the current time. Note that the 'a' option still sets
1738 the answer time to the current time if the channel was already answered.
1740 * The 's' option has been removed. A variable can be set on the original CDR
1741 if desired using the CDR function, and removed from a forked CDR using the
1744 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
1745 longer applies in the CDR engine.
1747 * The 'v' option now prevents the copy of the variables from the original CDR
1748 to the forked CDR. Previously the variables were always copied but were
1749 removed from the original. This was changed as removing variables from a CDR
1750 can have unintended side effects - this option allows the user to prevent
1751 propagation of variables from the original to the forked without modifying
1756 * Added the 'n' option to MeetMe to prevent application of the DENOISE
1757 function to a channel joining a conference. Some channel drivers that vary
1758 the number of audio samples in a voice frame will experience significant
1759 quality problems if a denoiser is attached to the channel; this option gives
1760 them the ability to remove the denoiser without having to unload func_speex.
1764 * The 'b' option now includes conferences as well as sounds played to the
1767 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
1768 running during a transfer. If a MixMonitor is started on a channel,
1769 the MixMonitor will continue to record the audio passing through the
1770 channel even in the presence of transfers.
1774 * The NoCDR application is deprecated. Please use the CDR_PROP function to
1777 * While the NoCDR application will prevent CDRs for a channel from being
1778 propagated to registered CDR backends, it will not prevent that data from
1779 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
1780 function that enables CDRs on a channel will restore those records that have
1781 not yet been finalized.
1785 * The app_parkandannounce module has been removed. The application
1786 ParkAndAnnounce is now provided by the res_parking module. See the
1787 res_parking changes for more information.
1791 * Added queue available hint. The hint can be added to the dialplan using the
1792 following syntax: exten,hint,Queue:{queue_name}_avail
1793 For example, if the name of the queue is 'markq':
1794 exten => 8501,hint,Queue:markq_avail
1795 This will report 'InUse' if there are no logged in agents or no free agents.
1796 It will report 'Idle' when an agent is free.
1798 * Queues now support a hint for member paused state. The hint uses the form
1799 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
1800 are the name of the queue and the name of the member to subscribe to,
1801 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
1802 Members will show as In Use when paused.
1804 * The configuration options eventwhencalled and eventmemberstatus have been
1805 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
1806 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
1807 sent. The "Variable" fields will also no longer exist on the Agent* events.
1808 These events can be filtered out from a connected AMI client using the
1809 eventfilter setting in manager.conf.
1811 * The queue log now differentiates between blind and attended transfers. A
1812 blind transfer will result in a BLINDTRANSFER message with the destination
1813 context and extension. An attended transfer will result in an
1814 ATTENDEDTRANSFER message. This message will indicate the method by which
1815 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
1816 for running an application on a bridge or channel, or "LINK" for linking
1817 two bridges together with local channels. The queue log will also now detect
1818 externally initiated blind and attended transfers and record the transfer
1821 * When performing queue pause/unpause on an interface without specifying an
1822 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
1823 least one member of any queue exists for that interface.
1825 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
1826 for realtime queue log entries.
1830 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
1831 CDRs when they were previously disabled on a channel.
1833 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
1834 backends occurs on an as-needed basis in order to preserve linkedid
1835 propagation and other needed behavior.
1839 * A new application, this is similar to SayAlpha except that it supports
1840 case sensitive playback of the specified characters. For example,
1841 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
1845 * This application is deprecated in favor of CHANNEL(amaflags).
1849 * The SendDTMF application will now accept 'W' as valid input. This will cause
1850 the application to delay one second while streaming DTMF.
1854 * A new application in Asterisk 12, this hands control of the channel calling
1855 the application over to an external system. Currently, external systems
1856 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
1860 * UserEvent will now handle duplicate keys by overwriting the previous value
1861 assigned to the key.
1863 * In addition to AMI, UserEvent invocations will now be distributed to any
1864 interested Stasis applications.
1868 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1869 system as mailbox@context. The rest of the system cannot add @default
1870 to mailbox identifiers for app_voicemail that do not specify a context
1871 any longer. It is a mailbox identifier format that should only be
1872 interpreted by app_voicemail.
1874 * The voicemail.conf configuration file now has an 'alias' configuration
1875 parameter for use with the Directory application. The voicemail realtime
1876 database table schema has also been updated with an 'alias' column.
1881 * Pass through support has been added for both VP8 and Opus.
1883 * Added format attribute negotiation for the Opus codec. Format attribute
1884 negotiation is provided by the res_format_attr_opus module.
1889 * Masquerades as an operation inside Asterisk have been effectively hidden
1890 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
1891 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
1892 dropping of frame/audio hooks, and other internal implementation details
1893 that users had to deal with. This fundamental change has large implications
1894 throughout the changes documented for this version. For more information
1895 about the new core architecture of Asterisk, please see the Asterisk wiki.
1897 * Multiple parties in a bridge may now be transferred. If a participant in a
1898 multi-party bridge initiates a blind transfer, a Local channel will be used
1899 to execute the dialplan location that the transferer sent the parties to. If
1900 a participant in a multi-party bridge initiates an attended transfer,
1901 several options are possible. If the attended transfer results in a transfer
1902 to an application, a Local channel is used. If the attended transfer results
1903 in a transfer to another channel, the resulting channels will be merged into
1906 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
1907 driver specific. If the channel variable is set on the transferrer channel,
1908 the sound will be played to the target of an attended transfer.
1910 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
1911 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
1912 listed. Any more peers in the bridge will not be included in the list.
1913 BRIDGEPEER is not valid in holding bridges like parking since those channels
1914 do not talk to each other even though they are in a bridge.
1916 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
1917 and will contain a value if the BRIDGEPEER's channel driver supports it.
1919 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
1920 was responsible for an attended transfer in a similar fashion to
1923 * Modules using the Configuration Framework or Sorcery must have XML
1924 configuration documentation. This configuration documentation is included
1925 with the rest of Asterisk's XML documentation, and is accessible via CLI
1926 commands. See the CLI changes for more information.
1928 AMI (Asterisk Manager Interface)
1930 * Major changes were made to both the syntax as well as the semantics of the
1931 AMI protocol. In particular, AMI events have been substantially improved
1932 in this version of Asterisk. For more information, please see the AMI
1933 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
1935 * AMI events that reference a particular channel or bridge will now always
1936 contain a standard set of fields. When multiple channels or bridges are
1937 referenced in an event, fields for at least some subset of the channels
1938 and bridges in the event will be prefixed with a descriptive name to avoid
1939 name collisions. See the AMI event documentation on the Asterisk wiki for
1942 * The CLI command 'manager show commands' no longer truncates command names
1943 longer than 15 characters and no longer shows authorization requirement
1944 for commands. 'manager show command' now displays the privileges needed
1945 for using a given manager command instead.
1947 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
1948 peer in its response if the peer has a subscribe context set.
1950 * The SIPqualifypeer action now acknowledges the request once it has
1951 established that the request is against a known peer. It also issues a new
1952 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
1954 * The PlayDTMF action now supports an optional 'Duration' parameter. This
1955 specifies the duration of the digit to be played, in milliseconds.
1957 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
1958 updates when changes occur instead of requiring the use of pollmailboxes.
1960 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
1961 AMI client to manipulate audio currently being played back on a channel. The
1962 supported operations depend on the application being used to send audio to
1963 the channel. When the audio playback was initiated using the ControlPlayback
1964 application or CONTROL STREAM FILE AGI command, the audio can be paused,
1965 stopped, restarted, reversed, or skipped forward. When initiated by other
1966 mechanisms (such as the Playback application), the audio can be stopped,
1967 reversed, or skipped forward.
1969 * Channel related events now contain a snapshot of channel state, adding new
1970 fields to many of these events.
1972 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
1973 in a future release. Please use the common 'Exten' field instead.
1975 * The AMI event 'UserEvent' from app_userevent now contains the channel state
1976 fields. The channel state fields will come before the body fields.
1978 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
1979 'UnParkedCall' have changed significantly in the new res_parking module.
1981 The 'Channel' and 'From' headers are gone. For the channel that was parked
1982 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
1983 has a number of fields associated with it. The old 'Channel' header relayed
1984 the same data as the new 'ParkeeChannel' header.
1986 The 'From' field was ambiguous and changed meaning depending on the event.
1987 for most of these, it was the name of the channel that parked the call
1988 (the 'Parker'). There is no longer a header that provides this channel name,
1989 however the 'ParkerDialString' will contain a dialstring to redial the
1990 device that parked the call.
1992 On UnParkedCall events, the 'From' header would instead represent the
1993 channel responsible for retrieving the parkee. It receives a channel
1994 snapshot labeled 'Retriever'. The 'from' field is is replaced with
1997 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
1999 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
2000 fashion has changed the field names 'StartExten' and 'StopExten' to
2001 'StartSpace' and 'StopSpace' respectively.
2003 * The deprecated use of | (pipe) as a separator in the channelvars setting in
2004 manager.conf has been removed.
2006 * Channel Variables conveyed with a channel no longer contain the name of the
2007 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
2008 ChanVariable: bar=baz. When multiple channels are present in a single AMI
2009 event, the various ChanVariable fields will contain a suffix that specifies
2010 which channel they correspond to.
2012 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
2013 event always conveys the AMI event for a particular channel.
2015 * All 'Reload' events have been consolidated into a single event type. This
2016 event will always contain a Module field specifying the name of the module
2017 and a Status field denoting the result of the reload. All modules now issue
2018 this event when being reloaded.
2020 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
2021 fail to receive this event due to being connected after modules have loaded.
2022 AMI connections that want to know when Asterisk is ready should listen for
2023 the 'FullyBooted' event.
2025 * app_fax now sends the same send fax/receive fax events as res_fax. The
2026 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
2027 now the 'ReceiveFAX' event.
2029 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
2030 'MusicOnHoldStop'. The sub type field has been removed.
2032 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
2033 carrier for another protocol.
2035 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
2036 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
2037 to the specific channel. 'Both' may be specified to play a tone to both
2038 channels. The old 'yes' option is still accepted as a way of playing the
2039 tone to Channel2 only.
2041 * The AMI 'Status' response event to the AMI Status action replaces the
2042 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
2043 indicate what bridge the channel is currently in.
2045 * The AMI 'Hold' event has been moved out of individual channel drivers, into
2046 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
2049 * The AMI events in app_queue have been made more consistent with each other.
2050 Events that reference channels (QueueCaller* and Agent*) will show
2051 information about each channel. The (infamous) 'Join' and 'Leave' AMI
2052 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
2054 * The 'MCID' AMI event now publishes a channel snapshot when available and
2055 its non-channel-snapshot parameters now use either the "MCallerID" or
2056 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
2057 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
2058 parameters in the channel snapshot.
2060 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
2061 'AgentLogin' and 'AgentLogoff' respectively.
2063 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
2064 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
2066 * 'ChannelUpdate' events have been removed.
2068 * All AMI events now contain a 'SystemName' field, if available.
2070 * Local channel optimization is now conveyed in two events:
2071 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
2072 when the Local channel driver begins attempting to optimize itself out of
2073 the media path; the End event is sent after the channel halves have
2074 successfully optimized themselves out of the media path.
2076 * Local channel information in events is now prefixed with 'LocalOne' and
2077 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
2078 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
2079 and 'LocalOptimizationEnd' events.
2081 * The option 'allowmultiplelogin' can now be set or overriden in a particular
2082 account. When set in the general context, it will act as the default
2083 setting for defined accounts.
2085 * The 'BridgeAction' event was removed. It technically added no value, as the
2086 Bridge Action already receives confirmation of the bridge through a
2087 successful completion Event.
2089 * The 'BridgeExec' events were removed. These events duplicated the events that
2090 occur in the Briding API, and are conveyed now through BridgeCreate,
2091 BridgeEnter, and BridgeLeave events.
2093 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
2094 previous versions. They now report all SR/RR packets sent/received, and
2095 have been restructured to better reflect the data sent in a SR/RR. In
2096 particular, the event structure now supports multiple report blocks.
2098 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
2099 raised when a blind transfer/attended transfer completes successfully.
2100 They contain information about the transfer that just completed, including
2101 the location of the transfered channel.
2103 * Added a 'security' class to AMI which outputs the required fields for
2104 security messages similar to the log messages from res_security_log
2106 * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
2107 that describes the status value in a human readable string.
2109 CDR (Call Detail Records)
2111 * Significant changes have been made to the behavior of CDRs. The CDR engine
2112 was effectively rewritten and built on the Stasis message bus. For a full
2113 definition of CDR behavior in Asterisk 12, please read the specification
2114 on the Asterisk wiki (wiki.asterisk.org).
2116 * CDRs will now be created between all participants in a bridge. For each
2117 pair of channels in a bridge, a CDR is created to represent the path of
2118 communication between those two endpoints. This lets an end user choose who
2119 to bill for what during bridge operations with multiple parties.
2121 * The duration, billsec, start, answer, and end times now reflect the times
2122 associated with the current CDR for the channel, as opposed to a cumulative
2123 measurement of all CDRs for that channel.
2125 * When a CDR is dispatched, user defined CDR variables from both parties are
2126 included in the resulting CDR. If both parties have the same variable, only
2127 the Party A value is provided.
2129 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
2130 information regarding the CDR engine is logged as verbose messages. This
2131 option should only be used if the behavior of the CDR engine needs to be
2134 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
2135 normally configured in cdr.conf.
2137 * Added CLI command 'cdr show active {channel}'. When {channel} is not
2138 specified, this command provides a summary of the channels with CDR
2139 information and their statistics. When {channel} is specified, it shows
2140 detailed information about all records associated with {channel}.
2142 CEL (Channel Event Logging)
2144 * CEL has undergone significant rework in Asterisk 12, and is now built on the
2145 Stasis message bus. Please see the specification for CEL on the Asterisk
2146 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
2149 * The 'extra' field of all CEL events that use it now consists of a JSON blob
2150 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
2152 * BLINDTRANSFER events now report the transferee bridge unique
2153 identifier, extension, and context in a JSON blob as the extra string
2154 instead of the transferee channel name as the peer.
2156 * ATTENDEDTRANSFER events now report the peer as NULL and additional
2157 information in the 'extra' string as a JSON blob. For transfers that occur
2158 between two bridged channels, the 'extra' JSON blob contains the primary
2159 bridge unique identifier, the secondary channel name, and the secondary
2160 bridge unique identifier. For transfers that occur between a bridged channel
2161 and a channel running an app, the 'extra' JSON blob contains the primary
2162 bridge unique identifier, the secondary channel name, and the app name.
2164 * LOCAL_OPTIMIZE events have been added to convey local channel
2165 optimizations with the record occurring for the semi-one channel and
2166 the semi-two channel name in the peer field.
2168 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
2169 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
2170 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
2171 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
2172 regardless of whether or not that bridge happens to contain multiple
2177 * When compiled with '--enable-dev-mode', the astobj2 library will now add
2178 several CLI commands that allow for inspection of ao2 containers that
2179 register themselves with astobj2. The CLI commands are 'astobj2 container
2180 dump', 'astobj2 container stats', and 'astobj2 container check'.
2182 * Added specific CLI commands for bridge inspection. This includes 'bridge
2183 show all', which lists all bridges in the system, and 'bridge show {id}',
2184 which provides specific information about a bridge.
2186 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
2187 ejecting the channels currently in the bridge. If the channels cannot
2188 continue in the dialplan or application that put them in the bridge, they
2191 * Added command 'bridge kick'. This will eject a single channel from a bridge.
2193 * Added commands to inspect and manipulate the registered bridge technologies.
2194 This include 'bridge technology show', which lists the registered bridge
2195 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
2196 which controls whether or not a registered bridge technology can be used
2197 during smart bridge operations. If a technology is suspended, it will not
2198 be used when a bridge technology is picked for channels; when unsuspended,
2199 it can be used again.
2201 * The command 'config show help {module} {type} {option}' will show
2202 configuration documentation for modules with XML configuration
2203 documentation. When {module}, {type}, and {option} are omitted, a listing
2204 of all modules with registered documentation is displayed. When {module}
2205 is specified, a listing of all configuration types for that module is
2206 displayed, along with their synopsis. When {module} and {type} are
2207 specified, a listing of all configuration options for that type are
2208 displayed along with their synopsis. When {module}, {type}, and {option}
2209 are specified, detailed information for that configuration option is
2212 * Added 'core show sounds' and 'core show sound' CLI commands. These display
2213 a listing of all installed media sounds available on the system and
2214 detailed information about a sound, respectively.
2216 * 'xmldoc dump' has been added. This CLI command will dump the XML
2217 documentation DOM as a string to the specified file. The Asterisk core
2218 will populate certain XML elements pulled from the source files with
2219 additional run-time information; this command lets a user produce the
2220 XML documentation with all information.
2224 * Parking has been pulled from core and placed into a separate module called
2225 res_parking. See Parking changes below for more details. Configuration for
2226 parking should now be performed in res_parking.conf. Configuration for
2227 parking in features.conf is now unsupported.
2229 * Core attended transfers now have several new options. While performing an
2230 attended transfer, the transferer now has the following options:
2231 - *1 - cancel the attended transfer (configurable via atxferabort)
2232 - *2 - complete the attended transfer, dropping out of the call
2233 (configurable via atxfercomplete)
2234 - *3 - complete the attended transfer, but stay in the call. This will turn
2235 the call into a multi-party bridge (configurable via atxferthreeway)
2236 - *4 - swap to the other party. Once an attended transfer has begun, this
2237 options may be used multiple times (configurable via atxferswap)
2239 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
2240 must be on the channel initiating the transfer to have any effect.
2242 * The BRIDGE_FEATURES channel variable would previously only set features for
2243 the calling party and would set this feature regardless of whether the
2244 feature was in caps or in lowercase. Use of a caps feature for a letter
2245 will now apply the feature to the calling party while use of a lowercase
2246 letter will apply that feature to the called party.
2248 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
2250 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
2251 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
2252 activated the dynamic feature.
2254 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
2255 only on the channel executing the dynamic feature. Executing a dynamic
2256 feature on the bridge peer in a multi-party bridge will execute it on all
2257 peers of the activating channel.
2259 * You can now have the settings for a channel updated using the FEATURE()
2260 and FEATUREMAP() functions inherited to child channels by setting
2261 FEATURE(inherit)=yes.
2263 * automixmon now supports additional channel variables from automon including:
2264 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
2265 and TOUCH_MIXMONITOR_MESSAGE_STOP
2267 * A new general features.conf option 'recordingfailsound' has been added which
2268 allowssetting a failure sound for a user tries to invoke a recording feature
2269 such as automon or automixmon and it fails.
2271 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
2272 features.c for atxferdropcall=no to work properly. This option now just
2277 * Added log rotation strategy 'none'. If set, no log rotation strategy will
2278 be used. Given that this can cause the Asterisk log files to grow quickly,
2279 this option should only be used if an external mechanism for log management
2284 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
2285 will store the path information for that peer when it registers. Realtime
2286 tables can also use the 'supportpath' field to enable Path header support.
2288 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
2289 objectIdentifier. This maps to the supportpath option in sip.conf.
2293 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
2294 provides modules a useful abstraction on top of the many storage mechanisms
2295 in Asterisk, including the Asterisk Database, static configuration files,
2296 static Realtime, and dynamic Realtime. It also provides a caching service.
2297 Users can configure a hierarchy of data storage layers for specific modules
2300 * All future modules which utilize Sorcery for object persistence must have a
2301 column named "id" within their schema when using the Sorcery realtime module.
2302 This column must be able to contain a string of up to 128 characters in length.
2304 Security Events Framework
2306 * Security Event timestamps now use ISO 8601 formatted date/time instead of
2307 the "seconds-microseconds" format that it was using previously.
2311 * The Stasis message bus is a publish/subscribe message bus internal to
2312 Asterisk. Many services in Asterisk are built on the Stasis message bus,
2313 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
2314 Stasis can be configured in stasis.conf. Note that these parameters operate
2315 at a very low level in Asterisk, and generally will not require changes.
2319 * When a channel driver is configured to enable jiterbuffers, they are now
2320 applied unconditionally when a channel joins a bridge. If a jitterbuffer
2321 is already set for that channel when it enters, such as by the JITTERBUFFER
2322 function, then the existing jitterbuffer will be used and the one set by
2323 the channel driver will not be applied.
2327 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
2328 dialplan applications provided by the app_agent_pool module. Agents are
2329 connected with callers using the new AgentRequest dialplan application.
2330 The Agents:<agent-id> device state is available to monitor the status of an
2331 agent. See agents.conf.sample for valid configuration options.
2333 * The updatecdr option has been removed. Altering the names of channels on a
2334 CDR is not supported - the name of the channel is the name of the channel,
2335 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
2336 has also been removed, for the same reason.
2338 * The endcall and enddtmf configuration options are removed. Use the
2339 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
2340 channel before calling AgentLogin.
2344 * chan_bridge has been removed. Its functionality has been incorporated
2345 directly into the ConfBridge application itself.
2349 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
2350 of the specified span and its B-channels. Note that this command should
2351 only be used if you understand the risks it entails.
2353 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
2354 A range of channels can be specified to be destroyed. Note that this command
2355 should only be used if you understand the risks it entails.
2357 * Added the CLI command 'dahdi create channels'. A range of channels can be
2358 specified to be created, or the keyword 'new' can be used to add channels
2361 * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
2362 the exact configured mailbox name. For app_voicemail mailboxes this is
2365 * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
2369 * IPv6 support has been added. We are now able to bind to and
2370 communicate using IPv6 addresses.
2374 * The /b option has been removed.
2376 * chan_local moved into the system core and is no longer a loadable module.
2380 * Added general support for busy detection.
2382 * Added ECAM command support for Sony Ericsson phones.
2386 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
2387 SIP stack. A collection of resource modules provides the bulk of the SIP
2388 functionality. For more information on the new SIP channel driver, see
2389 https://wiki.asterisk.org/wiki/x/JYGLAQ
2393 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
2394 using the 'supportpath' setting, either on a global basis or on a peer basis.
2395 This setting enables Asterisk to route outgoing out-of-dialog requests via a
2396 set of proxies by using a pre-loaded route-set defined by the Path headers in
2397 the REGISTER request. See Realtime updates for more configuration information.
2399 * The SIP_CODEC family of variables may now specify more than one codec. Each
2400 codec must be separated by a comma. The first codec specified is the
2401 preferred codec for the offer. This allows a dialplan writer to specify both
2402 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
2404 * The 'callevents' parameter has been removed. Hold AMI events are now raised
2405 in the core, and can be filtered out using the 'eventfilter' parameter
2408 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
2409 codecs configured for a peer instead of the requested codec.
2411 * The option "register_retry_403" has been added to chan_sip to work around
2412 servers that are known to erroneously send 403 in response to valid
2413 REGISTER requests and allows Asterisk to continue attepmting to connect.
2417 * Added the 'immeddialkey' parameter. If set, when the user presses the
2418 configured key the already entered number will be immediately dialed. This
2419 is useful when the dialplan allows for variable length pattern matching.
2420 Valid options are '*' and '#'.
2422 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
2423 milliseconds) before a call forward is considered to not be answered.
2425 * The 'serviceurl' parameter allows Service URLs to be attached to line
2434 * The password option has been disabled, as the AgentLogin application no
2435 longer provides authentication.
2439 * Due to changes in the Asterisk core, this function is no longer needed to
2440 preserve a MixMonitor on a channel during transfer operations and dialplan
2441 execution. It is effectively obsolete.
2445 * The 'amaflags' and 'accountcode' attributes for the CDR function are
2446 deprecated. Use the CHANNEL function instead to access these attributes.
2448 * The 'l' option has been removed. When reading a CDR attribute, the most
2449 recent record is always used. When writing a CDR attribute, all non-finalized
2452 * The 'r' option has been removed, for the same reason as the 'l' option.
2454 * The 's' option has been removed, as LOCKED semantics no longer exist in the
2459 * A new function CDR_PROP has been added. This function lets you set properties
2460 on a channel's active CDRs. This function is write-only. Properties accept
2461 boolean values to set/clear them on the channel's CDRs. Valid properties
2463 - 'party_a' - make this channel the preferred Party A in any CDR between two
2464 channels. If two channels have this property set, the creation time of the
2465 channel is used to determine who is Party A. Note that dialed channels are
2466 never Party A in a CDR.
2467 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
2468 application when set to True, and analogous to the 'e' option in ResetCDR
2473 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
2474 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
2475 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
2478 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
2479 string, i.e., [[context],extension],priority. If set on a channel, if a
2480 channel leaves a bridge but is not hung up it will resume dialplan execution
2485 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
2486 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
2487 The value of this setting is ignored when disabled is used for the argument.
2491 * A new function provided by chan_pjsip, this function can be used in
2492 conjunction with the Dial application to construct a dial string that will
2493 dial all contacts on an Address of Record associated with a chan_pjsip
2498 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
2499 outbound channel prior to dialing.
2503 * Redirecting reasons can now be set to arbitrary strings. This means
2504 that the REDIRECTING dialplan function can be used to set the redirecting
2505 reason to any string. It also allows for custom strings to be read as the
2506 redirecting reason from SIP Diversion headers.
2510 * The SPEECH_ENGINE function now supports read operations. When read from, it
2511 will return the current value of the requested attribute.
2515 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
2516 system as mailbox@context. The rest of the system cannot add @default
2517 to mailbox identifiers for app_voicemail that do not specify a context
2518 any longer. It is a mailbox identifier format that should only be
2519 interpreted by app_voicemail.
2525 res_agi (Asterisk Gateway Interface)
2527 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
2529 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
2532 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
2533 will start the playback of the audio at the position specified. It will
2534 also return the final position of the file in 'endpos'.
2536 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
2537 channel variable if the user stopped the file playback or if a remote
2538 entity stopped the playback. If neither stopped the playback, it will
2539 indicate the overall success/failure of the playback. If stopped early,
2540 the final offset of the file will be set in the CPLAYBACKOFFSET channel
2543 * The SAY ALPHA command now accepts an additional parameter to control
2544 whether it specifies the case of uppercase, lowercase, or all letters to
2545 provide functionality similar to SayAlphaCase.
2547 res_ari (Asterisk RESTful Interface) (and others)
2549 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
2550 control telephony primitives in Asterisk by remote client. This includes
2551 channels, bridges, endpoints, media, and other fundamental concepts. Users
2552 of ARI can develop their own communications applications, controlling
2553 multiple channels using an HTTP RESTful interface and receiving JSON events
2554 about the objects via a WebSocket connection. ARI can be configured in
2555 Asterisk via ari.conf. For more information on ARI, see
2556 https://wiki.asterisk.org/wiki/x/0YCLAQ
2560 * Parking has been extracted from the Asterisk core as a loadable module,
2561 res_parking. Configuration for parking is now provided by res_parking.conf.
2562 Configuration through features.conf is no longer supported.
2564 * res_parking uses the configuration framework. If an invalid configuration is
2565 supplied, res_parking will fail to load or fail to reload. Previously,
2566 invalid configurations would generally be accepted, with certain errors
2567 resulting in individually disabled parking lots.
2569 * Parked calls are now placed in bridges. While this is largely an
2570 architectural change, it does have implications on how channels in a parking
2571 lot are viewed. For example, commands that display channels in bridges will
2572 now also display the channels in a parking lot.
2574 * The order of arguments for the new parking applications have been modified.
2575 Timeout and return context/exten/priority are now implemented as options,
2576 while the name of the parking lot is now the first parameter. See the
2577 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
2578 in-depth information as well as syntax.
2580 * Extensions are by default no longer automatically created in the dialplan to
2581 park calls or pickup parked calls. Generation of dialplan extensions can be
2582 enabled using the 'parkext' configuration option.
2584 * ADSI functionality for parking is no longer supported. The 'adsipark'
2585 configuration option has been removed as a result.
2587 * The PARKINGSLOT channel variable has been deprecated in favor of
2588 PARKING_SPACE to match the naming scheme of the new system.
2590 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
2591 channel even when the configuration option 'comebactoorigin' is enabled.
2593 * A new CLI command 'parking show' has been added. This allows a user to
2594 inspect the parking lots that are currently in use.
2595 'parking show <parkinglot>' will also show the parked calls in a specific
2598 * The CLI command 'parkedcalls' is now deprecated in favor of
2599 'parking show <parkinglot>'.
2601 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
2602 can be used to get a list of parked calls for a specific parking lot.
2604 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
2605 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
2606 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
2607 longer a required argument.
2609 * The ParkAndAnnounce application is now provided through res_parking instead
2610 of through the separate app_parkandannounce module.
2612 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
2613 by default. Instead, it will follow the timeout rules of the parking lot. The
2614 old behavior can be reproduced by using the 'c' option.
2616 * Dynamic parking lots will now fail to be created under the following
2618 - if the parking lot specified by PARKINGDYNAMIC does not exist
2619 - if they require exclusive park and parkedcall extensions which overlap
2620 with existing parking lots.
2622 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
2623 currently contain no calls. Dynamic parking lots containing parked calls
2624 will persist through the reloads without alteration.
2626 * If 'parkext_exclusive' is set for a parking lot and that extension is
2627 already in use when that parking lot tries to register it, this is now
2628 considered a parking system configuration error. Configurations which do
2629 this will be rejected.
2631 * Added channel variable PARKER_FLAT. This contains the name of the extension
2632 that would be used if 'comebacktoorigin' is enabled. This can be useful when
2633 comebacktoorigin is disabled, but the dialplan or an external control
2634 mechanism wants to use the extension in the park-dial context that was
2635 generated to re-dial the parker on timeout.
2637 res_pjsip (and many others)
2639 * A large number of resource modules make up the SIP stack based on pjsip.
2640 The chan_pjsip channel driver users these resource modules to provide
2641 various SIP functionality in Asterisk. The majority of configuration for
2642 these modules is performed in pjsip.conf. Other modules may use their
2643 own configuration files.
2645 * Added 'set_var' option for an endpoint. For each variable specified that
2646 variable gets set upon creation of a channel involving the endpoint.
2650 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
2651 them, an Asterisk-specific version of PJSIP needs to be installed.
2652 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
2654 res_statsd/res_chan_stats
2656 * A new resource module, res_statsd, has been added, which acts as a statsd
2657 client. This module allows Asterisk to publish statistics to a statsd
2658 server. In conjunction with res_chan_stats, it will publish statistics about
2659 channels to the statsd server. It can be configured via res_statsd.conf.
2663 * Device state for XMPP buddies is now available using the following format:
2664 XMPP/<client name>/<buddy address>
2665 If any resource is available the device state is considered to be not in use.
2666 If no resources exist or all are unavailable the device state is considered
2673 Realtime/Database Scripts
2675 * Asterisk previously included example db schemas in the contrib/realtime/
2676 directory of the source tree. This has been replaced by a set of database
2677 migrations using the Alembic framework. This allows you to use alembic to
2678 initialize the database for you. It will also serve as a database migration
2679 tool when upgrading Asterisk in the future.
2681 See contrib/ast-db-manage/README.md for more details.
2685 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
2686 This python script will convert an existing sip.conf file to a
2687 pjsip.conf file, for use with the chan_pjsip channel driver. This script
2688 is meant to be an aid in converting an existing chan_sip configuration to
2689 a chan_pjsip configuration, but it is expected that configuration beyond
2690 what the script provides will be needed.
2692 ------------------------------------------------------------------------------
2693 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
2694 ------------------------------------------------------------------------------
2698 * The Asterisk build system will now build and install a shared library
2699 (libasteriskssl.so) used to wrap various initialization and shutdown functions
2700 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
2701 that Asterisk can ensure that these functions do *not* get called by any
2702 modules that are loaded into Asterisk, since they should only be called once
2703 in any single process. If desired, this feature can be disabled by supplying
2704 the "--disable-asteriskssl" option to the configure script.
2706 * A new make target, 'full', has been added to the Makefile. This performs
2707 the same compilation actions as make all, but will also scan the entirety of
2708 each source file for documentation. This option is needed to generate AMI
2709 event documentation. Note that your system must have Python in order for
2710 this make target to succeed.
2712 * The optimization portion of the build system has been reworked to avoid
2713 broken builds on certain architectures. All architecture-specific
2714 optimization has been removed in favor of using -march=native to allow gcc
2715 to detect the environment in which it is running when possible. This can
2716 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
2718 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
2719 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
2721 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
2722 previously parsed the header file to obtain the version of Asterisk, you
2723 will now have to go through Asterisk to get the version information.
2731 * Added 'F()' option. Similar to the dial option, this can be supplied with
2732 arguments indicating where the callee should go after the caller is hung up,
2733 or without options specified, the priority after the Queue will be used.
2738 * Added menu action admin_toggle_mute_participants. This will mute / unmute
2739 all non-admin participants on a conference. The confbridge configuration
2740 file also allows for the default sounds played to all conference users when
2741 this occurs to be overriden using sound_participants_unmuted and
2742 sound_participants_muted.
2744 * Added menu action participant_count. This will playback the number of
2745 current participants in a conference.
2747 * Added announcement configuration option to user profile. If set the sound
2748 file will be played to the user, and only the user, upon joining the
2751 * Added record_file_append option that defaults to "yes", but if set to no
2752 will create a new file between each start/stop recording.
2757 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
2758 channels respectively before the callee channels are called.
2763 * Added support for IPv6.
2765 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
2766 external process will cause the current playlist to be cleared, including
2767 stopping any audio file that is currently playing. This is useful when you
2768 want to interrupt audio playback only when specific DTMF is entered by the
2774 * A new option, 'I' has been added to app_followme. By setting this option,
2775 Asterisk will not update the caller with connected line changes when they
2776 occur. This is similar to app_dial and app_queue.
2778 * The 'N' option is now ignored if the call is already answered.
2780 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
2781 and caller channels respectively before the callee channels are called.
2783 * The winning FollowMe outgoing call is now put on hold if the caller put it on
2789 * MixMonitor hooks now have IDs associated with them which can be used to
2790 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
2791 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
2792 now accepts that ID as an argument.
2794 * Added 'm' option, which stores a copy of the recording as a voicemail in the
2795 indicated mailboxes.
2800 * The connect action in app_mysql now allows you to specify a port number to
2801 connect to. This is useful if you run a MySQL server on a non-standard
2807 * Increased the default number of allowed destinations from 5 to 12.
2812 * The app_page application now no longer depends on DAHDI or app_meetme. It
2813 has been re-architected to use app_confbridge internally.
2818 * Added queue options autopausebusy and autopauseunavail for automatically
2819 pausing a queue member when their device reports busy or congestion.
2821 * The 'ignorebusy' option for queue members has been deprecated in favor of
2822 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
2823 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
2824 per interface basis. Individual ringinuse values can now be set in
2825 queues.conf via an argument to member definitions. Lastly, the queue
2826 'ringinuse' setting now only determines defaults for the per member
2827 'ringinuse' setting and does not override per member settings like it does
2828 in earlier versions.
2830 * Added 'F()' option. Similar to the dial option, this can be supplied with
2831 arguments indicating where the callee should go after the caller is hung up,
2832 or without options specified, the priority after the Queue will be used.
2834 * Added new option log_member_name_as_agent, which will cause the membername to
2835 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
2836 state_interface has been set.
2838 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
2840 * App_queue will now play periodic announcements for the caller that
2841 holds the first position in the queue while waiting for answer.
2845 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
2846 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
2847 changed arguments to SayUnixTime so that every option is truly optional even
2848 when using multiple options (so that j option could be used without having to
2849 manually specify timezone and format) There are other benefits, e.g., format
2850 can now be used without specifying time zone as well.
2855 * Addition of the VM_INFO function - see Function changes.
2857 * The imapserver, imapport, and imapflags configuration options can now be
2858 overriden on a user by user basis.
2860 * When voicemail plays a message's envelope with saycid set to yes, when
2861 reaching the caller id field it will play a recording of a file with the same
2862 base name as the sender's callerid if there is a similarly named file in
2863 <astspooldir>/recordings/callerids/
2865 * Voicemails now contains a unique message identifier "msg_id", which is stored
2866 in the message envelope with the sound files. IMAP backends will now store
2867 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
2868 backends will store the message identifier in a "msg_id" column. See
2869 UPGRADE.txt for more information.
2871 * Added VoiceMailPlayMsg application. This application will play a single
2872 voicemail message from a mailbox. The result of the application, SUCCESS or
2873 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
2878 * Hangup handlers can be attached to channels using the CHANNEL() function.
2879 Hangup handlers will run when the channel is hung up similar to the h
2880 extension. The hangup_handler_push option will push a GoSub compatible
2881 location in the dialplan onto the channel's hangup handler stack. The
2882 hangup_handler_pop option will remove the last added location, and optionally
2883 replace it with a new GoSub compatible location. The hangup_handler_wipe
2884 option will remove all locations on the stack, and optionally add a new
2887 * The expression parser now recognizes the ABS() absolute value function,
2888 which will convert negative floating point values to positive values.
2890 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
2891 control of faxdetect.
2893 * Addition of the VM_INFO function that can be used to retrieve voicemail
2894 user information, such as the email address and full name.
2895 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
2898 * The REDIRECTING function now supports the redirecting original party id
2901 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
2902 lets you set some of the configuration options from the [general] section
2903 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
2904 the key sequence used to activate built-in features, such as blindxfer,
2905 and automon. See the built-in documentation for details.
2907 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
2908 instead of simply the uri. This is the format that MessageSend() can use
2909 in the from parameter for outgoing SIP messages.
2911 * Added the PRESENCE_STATE function. This allows retrieving presence state
2912 information from any presence state provider. It also allows setting
2913 presence state information from a CustomPresence presence state provider.
2914 See AMI/CLI changes for related commands.
2916 * Added the AMI_CLIENT function to make manager account attributes available
2917 to the dialplan. It currently supports returning the current number of
2918 active sessions for a given account.
2920 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
2921 and the REDIRECTING functions.
2929 * Added a manager event "LocalBridge" for local channel call bridges between
2930 the two pseudo-channels created.
2935 * Added dialtone_detect option for analog ports to disconnect incoming
2936 calls when dialtone is detected.
2938 * Added option colp_send to send ISDN connected line information. Allowed
2939 settings are block, to not send any connected line information; connect, to
2940 send connected line information on initial connect; and update, to send
2941 information on any update during a call. Default is update.
2943 * Add options namedcallgroup and namedpickupgroup to support installations
2944 where a higher number of groups (>64) is required.
2946 * Added support to use private party ID information with PRI calls.
2951 * A new channel driver named chan_motif has been added which provides support for
2952 Google Talk and Jingle in a single channel driver. This new channel driver includes
2953 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
2954 hold, unhold, and ringing notification. It is also compliant with the current Jingle
2955 specification, current Google Jingle specification, and the original Google Talk
2961 * Added NAT support for RTP. Setting in config is 'nat', which can be set
2962 globally and overriden on a peer by peer basis.
2964 * Direct media functionality has been added. Options in config are:
2965 directmedia (directrtp) and directrtpsetup (earlydirect)
2967 * ChannelUpdate events now contain a CallRef header.
2972 * Asterisk will no longer substitute CID number for CID name in the display
2973 name field if CID number exists without a CID name. This change improves
2974 compatibility with certain device features such as Avaya IP500's directory
2977 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
2978 created using that setting to not be removed during SIP reload.
2980 * Added settings recordonfeature and recordofffeature. When receiving an INFO
2981 request with a "Record:" header, this will turn the requested feature on/off.
2982 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
2983 dynamic features must be enabled and configured properly on the requesting
2984 channel for this to function properly.
2986 * Add support to realtime for the 'callbackextension' option.
2988 * When multiple peers exist with the same address, but differing
2989 callbackextension options, incoming requests that are matched by address
2990 will be matched to the peer with the matching callbackextension if it is
2993 * Two new NAT options, auto_force_rport and auto_comedia, have been added
2994 which set the force_rport and comedia options automatically if Asterisk
2995 detects that an incoming SIP request crossed a NAT after being sent by
2996 the remote endpoint.
2998 * The default global nat setting in sip.conf has been changed from force_rport
2999 to auto_force_rport.
3001 * NAT settings are now a combinable list of options. The equivalent of the
3002 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
3004 * Adds an option send_diversion which can be disabled to prevent
3005 diversion headers from automatically being added to INVITE requests.
3007 * Add support for lightweight NAT keepalive. If enabled a blank packet will
3008 be sent to the remote host at a given interval to keep the NAT mapping open.
3009 This can be enabled using the keepalive configuration option.
3011 * Add option 'tonezone' to specify country code for indications. This option
3012 can be set both globally and overridden for specific peers.
3014 * The SIP Security Events Framework now supports IPv6.
3016 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
3017 between multiple user agents. When set, for directmedia reinvites,
3018 Asterisk will not send an immediate reinvite on an incoming call leg. This
3019 option is useful when peered with another SIP user agent that is known to
3020 send immediate direct media reinvites upon call establishment.
3022 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
3025 * Add options subminexpiry and submaxexpiry to set limits of subscription
3026 timer independently from registration timer settings. The setting of the
3027 registration timer limits still is done by options minexpiry, maxexpiry
3028 and defaultexpiry. For backwards compatibility the setting of minexpiry
3029 and maxexpiry also is used to configure the subscription timer limits if
3030 subminexpiry and submaxexpiry are not set in sip.conf.
3032 * Set registration timer limits to default values when reloading sip
3033 configuration and values are not set by configuration.
3035 * Add options namedcallgroup and namedpickupgroup to support installations
3036 where a higher number of groups (>64) is required.
3038 * When a MESSAGE request is received, the address the request was received from
3039 is now saved in the SIP_RECVADDR variable.
3041 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
3042 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
3043 the ANI2/OLI information is set on the channel, which can be retrieved using
3044 the CALLERID function.
3046 * Peers can now be configured to support negotiation of ICE candidates using
3047 the setting icesupport. See res_rtp_asterisk changes for more information.
3049 * Added support for format attribute negotiation. See the Codecs changes for
3052 * Extra headers specified with SIPAddHeader are sent with the REFER message
3053 when using Transfer application. See refer_addheaders in sip.conf.sample.
3055 * Added support to use private party ID information with calls.
3057 * Adds an option discard_remote_hold_retrieval that when set stops telling
3058 the peer to start music on hold.
3063 * Added skinny version 17 protocol support.
3067 --------------------
3068 * Added option 'dtmf_duration' allowing playback time of DTMF tones to be set
3070 * Modified option 'date_format' to allow options to display date in 31Jan and Jan31
3071 formats as options 0 and 1. The previous options 0 and 1 now map to options 2 and 3
3072 as per the UNISTIM protocol.
3074 * Fixed issues with dialtone not matching indications.conf and mute stopping rx
3075 as well as tx. Also fixed issue with call "Timer" displaying as French "Dur\E9e"
3077 * Added ability to use multiple lines for a single phone. This allows multiple
3078 calls to occur on a single phone, using callwaiting and switching between calls.
3080 * Added option 'sharpdial' allowing end dialing by pressing # key
3082 * Added option 'interdigit_timer' to control phone dial timeout
3084 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
3086 * Added global 'debug' option, that enables debug in channel driver
3088 * Added ability to translate on-screen menu in multiple languages. Tested on
3089 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
3090 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
3093 * In addition to English added French and Russian languages for on-screen menus
3095 * Reworked dialing number input: added dialing by timeout, immediate dial on
3096 on dialplan compare, phone number length now not limited by screen size
3098 * Added ability to pickup a call using features.conf defined value and
3104 * Add options namedcallgroup and namedpickupgroup to support installations
3105 where a higher number of groups (>64) is required.
3107 * Added support to use private party ID information with calls.
3112 * The minimum DTMF duration can now be configured in asterisk.conf
3113 as "mindtmfduration". The default value is (as before) set to 80 ms.
3114 (previously it was only available in source code)
3116 * Named ACLs can now be specified in acl.conf and used in configurations that
3117 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
3118 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
3119 working ACL. In addition, some CLI commands have been added to provide
3120 show information and allow for module reloading - see CLI Changes.
3122 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
3123 items (separated by commas), and items in the rule can be negated by prefixing
3124 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
3125 longer necessray to control the order that the 'permit' and 'deny' columns are
3126 returned from queries.
3128 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
3129 be used within the dynamic weight attribute when specifying a mapping.
3131 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
3132 header, instead of putting the user defined event name there. When enabled
3133 the UserDefType header is added for user defined events. This feature is
3134 enabled with the setting show_user_defined.
3136 * Macro has been deprecated in favor of GoSub. For redirecting and connected
3137 line purposes use the following variables instead of their macro equivalents:
3138 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
3139 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
3140 cc_callback_macro in channel configurations.
3142 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
3145 * Call files now support the "early_media" option to connect with an outgoing
3146 extension when early media is received.
3148 * Added support to use private party ID information with calls.
3153 * A new channel variable, AGIEXITONHANGUP, has been added which allows
3154 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
3155 AGI application would exit immediately after a channel hangup is detected.
3157 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
3158 are resolved and each address is attempted in turn until one succeeds or
3162 AMI (Asterisk Manager Interface)
3164 * The originate action now has an option "EarlyMedia" that enables the
3165 call to bridge when we get early media in the call. Previously,
3166 early media was disregarded always when originating calls using AMI.
3168 * Added setvar= option to manager accounts (much like sip.conf)
3170 * Originate now generates an error response if the extension given is not found
3173 * MixMonitor will now show IDs associated with the mixmonitor upon creating
3174 them if the i(variable) option is used. StopMixMonitor will accept
3175 MixMonitorID as an option to close specific MixMonitors.
3177 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
3178 updated to include information about peers configured with
3179 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
3180 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
3181 returned if auto_force_rport is not enabled.
3183 * Added SIPpeerstatus manager command which will generate PeerStatus events
3184 similar to the existing PeerStatus events found in chan_sip on demand.
3186 * Hangup now can take a regular expression as the Channel option. If you want
3187 to hangup multiple channels, use /regex/ as the Channel option. Existing
3188 behavior to hanging up a single channel is unchanged, but if you pass a regex,
3189 the manager will send you a list of channels back that were hung up.
3191 * Support for IPv6 addresses has been added.
3193 * AMI Events can now be documented in the Asterisk source. Note that AMI event
3194 documentation is only generated when Asterisk is compiled using 'make full'.
3195 See the CLI section for commands to display AMI event information.
3197 * The AMI Hangup event now includes the AccountCode header so you can easily
3198 correlate with AMI Newchannel events.
3200 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
3201 the StateInterface of the queue member.
3203 * Added AMI event SessionTimeout in the Call category that is issued when a
3204 call is terminated due to either RTP stream inactivity or SIP session timer
3207 * CEL events can now contain a user defined header UserDefType. See core
3208 changes for more information.
3210 * OOH323 ChannelUpdate events now contain a CallRef header.
3212 * Added PresenceState command. This command will report the presence state for
3213 the given presence provider.
3215 * Added Parkinglots command. This will list all parking lots as a series of
3216 AMI Parkinglot events.
3218 * Added MessageSend command. This behaves in the same manner as the
3219 MessageSend application, and is a technolgoy agnostic mechanism to send out
3220 of call text messages.
3222 * Added "message" class authorization. This grants an account permission to
3223 send out of call messages. Write-only.
3228 * The "dialplan add include" command has been modified to create context a context
3229 if one does not already exist. For instance, "dialplan add include foo into bar"
3230 will create context "bar" if it does not already exist.
3232 * A "dialplan remove context" command has been added to remove a context from
3235 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
3236 filenames of all running mixmonitors on a channel.
3238 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
3239 numeric instead of 0, 1, or 2.
3241 * "stun show status" will show a table describing how the STUN client is
3244 * "acl show [named acl]" will show information regarding a Named ACL. The
3245 acl module can be reloaded with "reload acl".
3247 * Added CLI command to display AMI event information - "manager show events",
3248 which shows a list of all known and documented AMI events, and "manager show
3249 event [event name]", which shows detail information about a specific AMI
3252 * The result of the CLI command "queue show" now includes the state interface
3253 information of the queue member.
3255 * The command "core set verbose" will now set a separate level of logging for
3256 each remote console without affecting any other console.
3258 * Added command "cdr show pgsql status" to check connection status
3260 * "sip show channel" will now display the complete route set.
3262 * Added "presencestate list" command. This command will list all custom
3263 presence states that have been set by using the PRESENCE_STATE dialplan
3266 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
3267 command. This changes a custom presence to a new state.
3272 * Codec lists may now be modified by the '!' character, to allow succinct
3273 specification of a list of codecs allowed and disallowed, without the
3274 requirement to use two different keywords. For example, to specify all
3275 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
3277 * Add support for parsing SDP attributes, generating SDP attributes, and
3278 passing it through. This support includes codecs such as H.263, H.264, SILK,
3279 and CELT. You are able to set up a call and have attribute information pass.
3280 This should help considerably with video calls.
3282 * The iLBC codec can now use a system-provided iLBC library if one is installed,
3283 just like the GSM codec.
3287 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
3288 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
3292 * Asterisk version and build information is now logged at the beginning of a
3295 * Threads belonging to a particular call are now linked with callids which get
3296 added to any log messages produced by those threads. Log messages can now be
3297 easily identified as involved with a certain call by looking at their call id.
3298 Call ids may also be attached to log messages for just about any case where
3299 it can be determined to be related to a particular call.
3301 * Each logging destination and console now have an independent notion of the
3302 current verbosity level. Logger.conf now allows an optional argument to
3303 the 'verbose' specifier, indicating the level of verbosity sent to that
3304 particular logging destination. Additionally, remote consoles now each
3305 have their own verbosity level. The command 'core set verbose' will now set
3306 a separate level for each remote console without affecting any other
3312 * Added 'announcement' option which will play at the start of MOH and between
3313 songs in modes of MOH that can detect transitions between songs (eg.
3319 * New per parking lot options: comebackcontext and comebackdialtime. See
3320 configs/features.conf.sample for more details.
3322 * Channel variable PARKER is now set when comebacktoorigin is disabled in
3325 * Channel variable PARKEDCALL is now set with the name of the parking lot
3326 when a timeout occurs.
3332 CDR Postgresql Driver
3334 * Added command "cdr show pgsql status" to check connection status
3337 CDR Adaptive ODBC Driver
3339 * Added schema option for databases that support specifying a schema.
3347 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
3348 CALENDAR_WRITE has completed successfully.
3353 * A new option, 'probation' has been added to rtp.conf
3354 RTP in strictrtp mode can now require more than 1 packet to exit learning
3355 mode with a new source (and by default requires 4). The probation option
3356 allows the user to change the required number of packets in sequence to any
3357 desired value. Use a value of 1 to essentially restore the old behavior.
3358 Also, with strictrtp on, Asterisk will now drop all packets until learning
3359 mode has successfully exited. These changes are based on how pjmedia handles
3360 media sources and source changes.
3362 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
3363 enabled or disabled using the icesupport setting. A variety of other
3364 settings have been introduced to configure STUN/TURN connections.
3369 * A new module, res_corosync, has been introduced. This module uses the
3370 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
3371 of Asterisk servers to both Message Waiting Indication (MWI) and/or
3372 Device State (presence) information. This module is very similar to, and
3373 is a replacement for the res_ais module that was in previous releases of
3379 * This module adds a cleaned up, drop-in replacement for res_jabber called
3380 res_xmpp. This provides the same externally facing functionality but is
3381 implemented differently internally. res_jabber has been deprecated in favor
3382 of res_xmpp; please see the UPGRADE.txt file for more information.
3387 * The safe_asterisk script has been updated to allow several of its parameters
3388 to be set from environment variables. This also enables a custom run
3389 directory of Asterisk to be specified, instead of defaulting to /tmp.
3391 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
3392 its value to determine the directory to assume is the top-level directory of
3393 the source tree. If the variable is not set, it defaults to the current
3394 behavior and uses the current working directory.
3396 ------------------------------------------------------------------------------
3397 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
3398 ------------------------------------------------------------------------------
3402 * Asterisk now has protocol independent support for processing text messages
3403 outside of a call. Messages are routed through the Asterisk dialplan.
3404 SIP MESSAGE and XMPP are currently supported. There are options in
3405 jabber.conf and sip.conf to allow enabling these features.
3406 -> jabber.conf: see the "sendtodialplan" and "context" options.
3407 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
3408 and "outofcall_message_context" options.
3409 The MESSAGE() dialplan function and MessageSend() application have been
3410 added to go along with this functionality. More detailed usage information
3411 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
3412 * If real-time text support (T.140) is negotiated, it will be preferred for
3413 sending text via the SendText application. For example, via SIP, messages
3414 that were once sent via the SIP MESSAGE request would be sent via RTP if
3415 T.140 text is negotiated for a call.
3419 * parkedmusicclass can now be set for non-default parking lots.
3421 Asterisk Manager Interface
3422 --------------------------
3423 * PeerStatus now includes Address and Port.
3424 * Added Hold events for when the remote party puts the call on and off hold
3425 for chan_dahdi ISDN channels.
3426 * Added new action MeetmeListRooms to list active conferences (shows same
3427 data as "meetme list" at the CLI).
3428 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
3429 Description field that is set by 'description' in the channel configuration
3431 * Added Uniqueid header to UserEvent.
3432 * Added new action FilterAdd to control event filters for the current session.
3433 This requires the system permission and uses the same filter syntax as
3434 filters that can be defined in manager.conf
3435 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
3436 versions had some instances of the event converted, but others were left
3437 as-is. All Unlink events should now be converted to Bridge events. The AMI
3438 protocol version number was incremented to 1.2 as a result of this change.
3440 Asterisk HTTP Server
3441 --------------------------
3442 * The HTTP Server can bind to IPv6 addresses.
3445 --------------------------
3446 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
3447 with busydetect. usage example: busypattern=200,200,200,600
3450 --------------------------
3451 * New 'gtalk show settings' command showing the current settings loaded from
3453 * The 'logger reload' command now supports an optional argument, specifying an
3454 alternate configuration file to use.
3455 * 'dialplan add extension' command will now automatically create a context if
3456 the specified context does not exist with a message indicated it did so.
3457 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
3458 Description field which can be populated with 'description' in the channel
3459 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
3462 --------------------------
3463 * The filter option in cdr_adaptive_odbc now supports negating the argument,
3464 thus allowing records which do NOT match the specified filter.
3465 * Added ability to log CONGESTION calls to CDR
3468 --------------------------
3469 * Ability to define custom SILK formats in codecs.conf.
3470 * Addition of speex32 audio format with translation.
3471 * CELT codec pass-through support and ability to define
3472 custom CELT formats in codecs.conf.
3473 * Ability to read raw signed linear files with sample rates
3474 ranging from 8khz - 192khz. The new file extensions introduced
3475 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
3476 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
3477 Skinny, H.323, etc) can still only support the following codecs:
3478 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
3479 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
3480 Video: h261, h263, h263p, h264, mpeg4
3485 --------------------------
3486 * New highly optimized and customizable ConfBridge application capable of
3487 mixing audio at sample rates ranging from 8khz-96khz.
3488 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
3489 and bridge profiles on a channel.
3490 * CONFBRIDGE_INFO dialplan function capable of retrieving information
3491 about a conference such as locked status and number of parties, admins,
3493 * Addition of video_mode option in confbridge.conf for adding video support
3494 into a bridge profile.
3495 * Addition of the follow_talker video_mode in confbridge.conf. This video
3496 mode dynamically switches the video feed to always display the loudest talker
3497 supplying video in the conference.
3501 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
3502 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
3503 variables from asterisk.conf.
3507 * Addition of the JITTERBUFFER dialplan function. This function allows
3508 for jitterbuffering to occur on the read side of a channel. By using
3509 this function conference applications such as ConfBridge and MeetMe can
3510 have the rx streams jitterbuffered before conference mixing occurs.
3511 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
3513 * Added STRREPLACE function. This function let's the user search a variable
3514 for a given string to replace with another string as many times as the
3515 user specifies or just throughout the whole string.
3516 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
3517 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
3518 * Added extensions to chan_ooh323 in function CHANNEL()
3520 libpri channel driver (chan_dahdi) DAHDI changes
3521 --------------------------
3522 * Added moh_signaling option to specify what to do when the channel's bridged
3523 peer puts the ISDN channel on hold.
3524 * Added display_send and display_receive options to control how the display ie
3525 is handled. To send display text from the dialplan use the SendText()
3526 application when the option is enabled.
3527 * Added mcid_send option to allow sending a MCID request on a span.
3530 --------------------------
3531 * Added setvar option to calendar.conf to allow setting channel variables on
3532 notification channels.
3533 * Added "calendar show types" CLI command to list registered calendar
3537 --------------------------
3538 * Added two new options, r and t with file name arguments to record
3539 single direction (unmixed) audio recording separate from the bidirectional
3540 (mixed) recording. The mixed file name argument is optional now as long
3541 as at least one recording option is used.
3544 --------------------------
3545 * Added a new option, l, which will disable local call optimization for
3546 channels involved with the FollowMe thread. Use this option to improve
3547 compatability for a FollowMe call with certain dialplan apps, options, and
3551 --------------------------
3552 * Added option "k" that will automatically close the conference when there's
3553 only one person left when a user exits the conference.
3556 --------------------------
3557 * cel_pgsql now supports the 'extra' column for data added using the
3558 CELGenUserEvent() application.
3561 --------------------------
3562 * Support for defining hints has been added to pbx_lua. See the 'hints' table
3563 in the sample extensions.lua file for syntax details.
3564 * Applications that perform jumps in the dialplan such as Goto will now
3565 execute properly. When pbx_lua detects that the context, extension, or
3566 priority we are executing on has changed it will immediately return control
3567 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
3568 the priority after the currently executing priority.
3569 * An autoservice is now started by default for pbx_lua channels. It can be
3570 stopped and restarted using the autoservice_stop() and autoservice_start()
3574 --------------------------
3575 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
3576 into a FAXStatus event with an 'Operation' header that will be either
3577 'send', 'receive', and 'gateway'.
3578 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
3579 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
3580 feature will handle converting a fax call between an audio T.30 fax terminal
3581 and an IFP T.38 fax terminal.
3585 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
3586 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
3587 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
3591 * Added general option negative_penalty_invalid default off. when set
3592 members are seen as invalid/logged out when there penalty is negative.
3593 for realtime members when set remove from queue will set penalty to -1.
3594 * Added queue option autopausedelay when autopause is enabled it will be
3595 delayed for this number of seconds since last successful call if there
3596 was no prior call the agent will be autopaused immediately.
3597 * Added member option ignorebusy this when set and ringinuse is not
3598 will allow per member control of multiple calls as ringinuse does for
3603 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
3605 * Added 'k' option to MeetMe to automatically kill the conference when there's only
3606 one participant left (much like a normal call bridge)
3607 * Added extra argument to Originate to set timeout.
3611 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
3612 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
3613 utility in the UTILS section of menuselect. If an existing astdb is found and no
3614 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
3615 convert an existing astdb to the SQLite3 version automatically at runtime.
3619 * Modules marked as deprecated are no longer marked as building by default. Enabling
3620 these modules is still available via menuselect.
3624 * authdebug is now disabled by default. To enable this functionaility again
3625 set authdebug = yes in iax.conf.
3629 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
3630 releases it was disabled.
3634 * The PBX core previously made a call with a non-existing extension test for
3635 extension s@default and jump there if the extension existed.
3636 This was a bad default behaviour and violated the principle of least surprise.
3637 It has therefore been changed in this release. It may affect some
3638 applications and configurations that rely on this behaviour. Most channel
3639 drivers have avoided this for many releases by testing whether the extension
3640 called exists before starting the PBX and generating a local error.
3641 This behaviour still exists and works as before.
3643 Extension "s" is used when no extension is given in a channel driver,
3644 like immediate answer in DAHDI or calling to a domain with no user part
3647 ------------------------------------------------------------------------------
3648 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
3649 ------------------------------------------------------------------------------
3653 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
3654 now defaults to force_rport. It is very important that phones requiring nat=no be
3655 specifically set as such instead of relying on the default setting. If at all
3656 possible, all devices should have nat settings configured in the general section as
3657 opposed to configuring nat per-device.
3658 * Added preferred_codec_only option in sip.conf. This feature limits the joint
3659 codecs sent in response to an INVITE to the single most preferred codec.
3660 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
3661 to be used for the outgoing call. It must be one of the codecs configured
3663 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
3664 to be used for holding a private key. If tlsprivatekey is not specified,
3665 tlscertfile is searched for both public and private key.
3666 * Added tlsclientmethod option to sip.conf. This allows the protocol for
3667 outbound client connections to be specified.
3668 * The sendrpid parameter has been expanded to include the options
3669 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
3670 header to be sent (equivalent to setting sendrpid=yes) and setting
3671 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
3672 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
3673 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
3674 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
3675 will accept the SDP even if the SDP version number is not properly incremented,
3676 but will generate a warning in the log indicating that the SIP peer that sent
3677 the SDP should have the 'ignoresdpversion' option set.
3678 * The 'nat' option has now been been changed to have yes, no, force_rport, and
3679 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
3680 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
3681 remote side requests it and disables symmetric RTP support. Setting it to
3682 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
3683 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
3684 and enables symmetric RTP support.
3685 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
3686 response. This permits the master channel to know how each channel dialled
3687 in a multi-channel setup resolved in an individual way. This carries a
3688 performance penalty and can be disabled in sip.conf using the
3689 'storesipcause' option.
3690 * Added 'externtcpport' and 'externtlsport' options to allow custom port
3691 configuration for the externip and externhost options when tcp or tls is used.
3692 * Added support for message body (stored in content variable) to SIP NOTIFY message
3693 accessible via AMI and CLI.
3694 * Added 'media_address' configuration option which can be used to explicitly specify
3695 the IP address to use in the SDP for media (audio, video, and text) streams.
3696 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
3697 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
3699 * Added 'use_q850_reason' configuration option for generating and parsing
3700 if available Reason: Q.850;cause=<cause code> header. It is implemented
3701 in some gateways for better passing PRI/SS7 cause codes via SIP.
3702 * When dialing SIP peers, a new component may be added to the end of the dialstring
3703 to indicate that a specific remote IP address or host should be used when dialing
3704 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
3705 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
3706 ability to selectively force bridged channels to also be encrypted is also
3707 implemented. Branching in the dialplan can be done based on whether or not
3708 a channel has secure media and/or signaling.
3709 * Added directmediapermit/directmediadeny to limit which peers can send direct media
3711 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
3712 Charge messages to snom phones.
3713 * Added support for G.719 media streams.
3714 * Added support for 16khz signed linear media streams.
3715 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
3716 RTP has been outfitted with the same abilities.
3717 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
3718 available in device configurations as well as in the dial plan.
3719 * Addition of the 'subscribe_network_change' option for turning on and off
3720 res_stun_monitor module support in chan_sip.
3721 * Addition of the 'auth_options_requests' option for turning on and off
3722 authentication for OPTIONS requests in chan_sip.
3726 * Add #tryinclude statement for config files. This provides the same
3727 functionality as the #include statement however an asterisk module will
3728 still load if the filename does not exist. Using the #include statement
3729 Asterisk will not allow the module to load.
3733 * Added rtsavesysname option into iax.conf to allow the systname to be saved
3734 on realtime updates.
3735 * Added the ability for chan_iax2 to inform the dialplan whether or not
3736 encryption is being used. This interoperates with the SIP SRTP implementation
3737 so that a secure SIP call can be bridged to a secure IAX call when the
3738 dialplan requires bridged channels to be "secure".
3739 * Addition of the 'subscribe_network_change' option for turning on and off
3740 res_stun_monitor module support in chan_iax.
3745 * Added ability to preset channel variables on indicated lines with the setvar
3746 configuration option. Also, clearvars=all resets the list of variables back
3748 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
3749 See configs/res_pktccops.conf for more information.
3751 XMPP Google Talk/Jingle changes
3752 -------------------------------
3753 * Added the externip option to gtalk.conf.
3754 * Added the stunaddr option to gtalk.conf which allows for the automatic
3755 retrieval of the external ip from a stun server.
3759 * Added 'p' option to PickupChan() to allow for picking up channel by the first
3760 match to a partial channel name.
3761 * Added .m3u support for Mp3Player application.
3762 * Added progress option to the app_dial D() option. When progress DTMF is
3763 present, those values are sent immediately upon receiving a PROGRESS message
3764 regardless if the call has been answered or not.
3765 * Added functionality to the app_dial F() option to continue with execution
3766 at the current location when no parameters are provided.
3767 * Added the 'a' option to app_dial to answer the calling channel before any
3768 announcements or macros are executed.
3769 * Modified app_dial to set answertime when the called channel answers even if
3770 the called channel hangs up during playback of an announcement.
3771 * Modified app_dial 'r' option to support an additional parameter to play an
3772 indication tone from indications.conf
3773 * Added c() option to app_chanspy. This option allows custom DTMF to be set
3774 to cycle through the next available channel. By default this is still '*'.
3775 * Added x() option to app_chanspy. This option allows DTMF to be set to
3776 exit the application.
3777 * The Voicemail application has been improved to automatically ignore messages
3778 that only contain silence.
3779 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
3780 associated mailbox(es) to be greetings-only.
3781 * The ChanSpy application now has the 'S' option, which makes the application
3782 automatically exit once it hits a point where no more channels are available
3784 * The ChanSpy application also now has the 'E' option, which spies on a single
3785 channel and exits when that channel hangs up.
3786 * The MeetMe application now turns on the DENOISE() function by default, for
3787 each participant. In our tests, this has significantly decreased background
3788 noise (especially noisy data centers).
3789 * Voicemail now permits storage of secrets in a separate file, located in the
3790 spool directory of each individual user. The control for this is located in
3791 the "passwordlocation" option in voicemail.conf. Please see the sample
3792 configuration for more information.
3793 * The ChanIsAvail application now exposes the returned cause code using a separate
3794 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
3795 * Added 'd' option to app_followme. This option disables the "Please hold"
3797 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
3798 received will terminate recording.
3799 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
3800 Previously the folder could only be set per context, but has now been extended
3801 using the imapfolder option.
3802 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
3803 * Voicemail now allows the pager date format to be specified separately from the
3805 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
3806 to allow joining, leaving, and sending text to group chats.
3807 * MeetMe has a new option 'G' to play an announcement before joining a conference.
3808 * Page has a new option 'A(x)' which will playback an announcement simultaneously
3809 to all paged phones (and optionally excluding the caller's one using the new
3810 option 'n') before the call is bridged.
3811 * The 'f' option to Dial has been augmented to take an optional argument. If no
3812 argument is provided, the 'f' option works as it always has. If an argument is
3813 provided, then the connected party information of all outgoing channels created
3814 during the Dial will be set to the argument passed to the 'f' option.
3815 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
3817 * The OSP lookup application adds in/outbound network ID, optional security,
3818 number portability, QoS reporting, destination IP port, custom info and service
3820 * Added new application VMSayName that will play the recorded name of the voicemail
3821 user if it exists, otherwise will play the mailbox number.
3822 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
3823 retrieve state for a particular bridge, where <name> is the conference name
3824 * app_directory now allows exiting at any time using the operator or pound key.
3825 * Voicemail now supports setting a locale per-mailbox.
3826 * Two new applications are provided for declining counting phrases in multiple
3827 languages. See the application notes for SayCountedNoun and SayCountedAdj for
3829 * Voicemail now runs the externnotify script when pollmailboxes is activated and
3831 * Voicemail now includes rdnis within msgXXXX.txt file.
3832 * ExternalIVR now supports IPv6 addresses.
3833 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
3834 at https://wiki.asterisk.org/wiki/x/oQBB
3835 * ParkedCall and Park can now specify the parking lot to use.
3839 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
3840 over SRV records associated with a specific service. From the CLI, type
3841 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
3842 details on how these may be used.
3843 * PITCH_SHIFT dialplan function added. This function can be used to modify the
3844 pitch of a channel's tx and rx audio streams.
3845 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
3846 setting various connected line and redirecting party information.
3847 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
3848 support ISDN subaddressing.
3849 * The CHANNEL() function now supports the "name" and "checkhangup" options.
3850 * For DAHDI channels, the CHANNEL() dialplan function now allows
3851 the dialplan to request changes in the configuration of the active
3852 echo canceller on the channel (if any), for the current call only.
3855 exten => s,n,Set(CHANNEL(echocan_mode)=off)
3857 The possible values are:
3859 on - normal mode (the echo canceller is actually reinitialized)
3861 fax - FAX/data mode (NLP disabled if possible, otherwise completely
3863 voice - voice mode (returns from FAX mode, reverting the changes that
3864 were made when FAX mode was requested)
3865 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
3866 and setting variables on the channel which created the current channel.
3867 Administrators should take care to avoid naming conflicts, when multiple
3868 channels are dialled at once, especially when used with the Local channel
3869 construct (which all could set variables on the master channel). Usage
3870 of the HASH() dialplan function, with the key set to the name of the slave
3871 channel, is one approach that will avoid conflicts.
3872 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
3874 * func_odbc now allows multiple row results to be retrieved without using
3875 mode=multirow. If rowlimit is set, then additional rows may be retrieved
3876 from the same query by using the name of the function which retrieved the
3877 first row as an argument to ODBC_FETCH().
3878 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
3879 dialplan. This function returns the content of the received message.
3880 * Added REPLACE, which searches a given variable name for a set of characters,
3881 then either replaces them with a single character or deletes them.
3882 * Added PASSTHRU, which literally passes the same argument back as its return
3883 value. The intent is to be able to use a literal string argument to
3884 functions that currently require a variable name as an argument.
3885 * HASH-associated variables now can be inherited across channel creation, by
3886 prefixing the name of the hash at assignment with the appropriate number of
3887 underscores, just like variables.
3888 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
3889 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
3890 whether or not channels that are bridged to the current channel will be
3891 required to have secure signaling and/or media.
3892 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
3893 the current channel has secure signaling and/or media.
3894 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
3895 "no_media_path" option.
3896 Returns "0" if there is a B channel associated with the call.
3897 Returns "1" if no B channel is associated with the call. The call is either
3898 on hold or is a call waiting call.
3899 * Added option to dialplan function CDR(), the 'f' option
3900 allows for high resolution times for billsec and duration fields.
3901 * FILE() now supports line-mode and writing.
3902 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
3903 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
3907 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
3908 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
3909 and is set when a dynamic feature is triggered.
3910 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
3911 to dynamically create a new parking lot matching the value this varible is
3913 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
3914 features.conf that should be the base for dynamic parkinglots.
3915 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
3916 parkinglot should have.
3917 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
3918 parkinglot should have.
3919 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
3924 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
3925 timeout has expired.
3926 * Added 'R' option to app_queue. This option stops moh and indicates ringing
3927 to the caller when an Agent's phone is ringing. This can be used to indicate
3928 to the caller that their call is about to be picked up, which is nice when
3929 one has been on hold for an extened period of time.
3930 * A new config option, penaltymemberslimit, has been added to queues.conf.
3931 When set this option will disregard penalty settings when a queue has too
3933 * A new option, 'I' has been added to both app_queue and app_dial.
3934 By setting this option, Asterisk will not update the caller with
3935 connected line changes or redirecting party changes when they occur.
3936 * A 'relative-periodic-announce' option has been added to queues.conf. When
3937 enabled, this option will cause periodic announce times to be calculated
3938 from the end of announcements rather than from the beginning.
3939 * The autopause option in queues.conf can be passed a new value, "all." The
3940 result is that if a member becomes auto-paused, he will be paused in all
3941 queues for which he is a member, not just the queue that failed to reach
3943 * Added dialplan function QUEUE_EXISTS to check if a queue exists
3944 * The queue logger now allows events to optionally propagate to a file,
3945 even when realtime logging is turned on. Additionally, realtime logging
3946 supports sending the event arguments to 5 individual fields, although it
3947 will fallback to the previous data definition, if the new table layout is
3950 mISDN channel driver (chan_misdn) changes
3951 ----------------------------------------
3952 * Added display_connected parameter to misdn.conf to put a display string
3953 in the CONNECT message containing the connected name and/or number if
3954 the presentation setting permits it.
3955 * Added display_setup parameter to misdn.conf to put a display string
3956 in the SETUP message containing the caller name and/or number if the
3957 presentation setting permits it.
3958 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
3959 indicate the dialplan settings are to be obtained from the asterisk
3961 * Made misdn.conf parameter callerid accept the "name" <number> format
3962 used by the rest of the system.
3963 * Made use the nationalprefix and internationalprefix misdn.conf
3964 parameters to prefix any received number from the ISDN link if that
3965 number has the corresponding Type-Of-Number. NOTE: This includes
3966 comparing the incoming call's dialed number against the MSN list.
3967 * Added the following new parameters: unknownprefix, netspecificprefix,
3968 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
3969 received number from the ISDN link if that number has the corresponding
3971 * Added new dialplan application misdn_command which permits controlling
3972 the CCBS/CCNR functionality.
3973 * Added new dialplan function mISDN_CC which permits retrieval of various
3974 values from an active call completion record.
3975 * For PTP, you should manually send the COLR of the redirected-to party
3976 for an incomming redirected call if the incoming call could experience
3977 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
3978 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
3979 if the REDIRECTING(from-num) is not empty.
3980 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
3981 option on all of the REDIRECTING statements before dialing the
3982 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
3983 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
3984 redirecting-to presentation (COLR) when it becomes available.
3985 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
3988 thirdparty mISDN enhancements
3989 -----------------------------
3990 mISDN has been modified by Digium, Inc. to greatly expand facility message
3992 * Enhanced COLP support for call diversion and transfer.
3993 * CCBS/CCNR support.
3995 The latest modified mISDN v1.1.x based version is available at:
3996 http://svn.digium.com/svn/thirdparty/mISDN/trunk
3997 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
3999 Tagged versions of the modified mISDN code are available under:
4000 http://svn.digium.com/svn/thirdparty/mISDN/tags
4001 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
4003 libpri channel driver (chan_dahdi) DAHDI changes
4004 -------------------------------------------
4005 * The channel variable PRIREDIRECTREASON is now just a status variable
4006 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
4007 to read and alter the reason.
4008 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
4009 redirected-to party for an incomming redirected call if the incoming call
4010 could experience further redirects. Just set the
4011 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
4012 to the COLR. A call has been redirected if the REDIRECTING(count) is not
4014 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
4015 use the inhibit(i) option on all of the REDIRECTING statements before
4016 dialing the redirected-to party. You still have to set the
4017 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
4018 will update the redirecting-to presentation (COLR) when it becomes available.
4019 * Added the ability to ignore calls that are not in a Multiple Subscriber
4020 Number (MSN) list for PTMP CPE interfaces.
4021 * Added dynamic range compression support for dahdi channels. It is
4022 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
4023 * Added support for ISDN calling and called subaddress with partial support
4024 for connected line subaddress.
4025 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
4026 * Added handling of received HOLD/RETRIEVE messages and the optional ability
4027 to transfer a held call on disconnect similar to an analog phone.
4028 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
4029 Will reroute/deflect an outgoing call when receive the message.
4030 Can use the DAHDISendCallreroutingFacility to send the message for the
4032 * Added standard location to add options to chan_dahdi dialing:
4033 Dial(DAHDI/g1[/extension[/options]])
4036 R Reverse charging indication
4037 * Added Reverse Charging Indication (Collect calls) send/receive option.
4038 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
4039 Dial(DAHDI/g1/extension/R)
4040 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
4041 (requires latest LibPRI)
4042 * Added ability to send/receive keypad digits in the SETUP message.
4043 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
4044 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
4045 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
4046 (requires latest LibPRI)
4047 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
4048 to eliminate tromboned calls. A tromboned call goes out an interface and comes
4049 back into the same interface. Tromboned calls happen because of call routing,
4050 call deflection, call forwarding, and call transfer.
4051 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
4052 * Added the ability to support call waiting calls. (The SETUP has no B channel
4054 * Added Malicious Call ID (MCID) event to the AMI call event class.
4055 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
4057 Asterisk Manager Interface
4058 --------------------------
4059 * The Hangup action now accepts a Cause header which may be used to
4060 set the channel's hangup cause.
4061 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
4062 to specify a separate .pem file to hold a private key. By default sslcert
4063 is used to hold both the public and private key.
4064 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
4065 for options containing the 'tls' prefix. For example, 'sslenable' is now
4066 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
4067 across all .conf files. All affected sample.conf files have been modified to
4068 reflect this change. Previous options such as 'sslenable' still work,
4069 but options with the 'tls' prefix are preferred.
4070 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
4071 in a channel. (res_mutestream.so)
4072 * The configuration file manager.conf now supports a channelvars option, which
4073 specifies a list of channel variables to include in each channel-oriented
4075 * The redirect command now has new parameters ExtraContext, ExtraExtension,
4076 and ExtraPriority to allow redirecting the second channel to a different
4077 location than the first.
4078 * Added new event "JabberStatus" in the Jabber module to monitor buddies
4080 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
4081 in a MixMonitor recording.
4082 * The 'iax2 show peers' output is now similar to the expected output of
4084 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
4086 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
4087 AOC-E messages on a channel.
4088 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
4089 conform more closely to similar events.
4090 * Added a new eventfilter option per user to allow whitelisting and blacklisting
4092 * Added optional parkinglot variable for park command.
4093 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
4094 if CallerIDNum and CallerIDName headers are also present.
4096 Channel Event Logging
4097 ---------------------
4098 * A new interface, CEL, is introduced here. CEL logs single events, much like
4099 the AMI, but it differs from the AMI in that it logs to db backends much
4100 like CDR does; is based on the event subsystem introduced by Russell, and
4101 can share in all its benefits; allows multiple backends to operate like CDR;
4102 is specialized to event data that would be of concern to billing sytems,
4103 like CDR. Backends for logging and accounting calls have been produced,
4104 but a new CDR backend is still in development.
4108 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
4109 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
4110 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
4111 * Multiple files and formats can now be specified in cdr_custom.conf.
4112 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
4113 See configs/cdr_syslog.conf.sample for more information.
4114 * A 'sequence' field has been added to CDRs which can be combined with
4115 linkedid or uniqueid to uniquely identify a CDR.
4116 * Handling of billsec and duration field has changed. If your table definition
4117 specifies those fields as float,double or similar they will now be logged with
4118 microsecond accuracy instead of a whole integer.
4120 Calendaring for Asterisk
4121 ------------------------
4122 * A new set of modules were added supporing calendar integration with Asterisk.
4123 Dialplan functions for reading from and writing to calendars are included,
4124 as well as the ability to execute dialplan logic upon calendar event notifications.
4125 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
4126 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
4127 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
4128 2003 support does not support forms-based authentication).
4130 Call Completion Supplementary Services for Asterisk
4131 ---------------------------------------------------
4132 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
4133 DAHDI/ISDN supports call completion for the following switch types:
4134 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
4135 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
4137 Multicast RTP Support
4138 ---------------------
4139 * A new RTP engine and channel driver have been added which supports Multicast RTP.
4140 The channel driver can be used with the Page application to perform multicast RTP
4141 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
4142 Type can be either basic or linksys.
4143 Destination is the IP address and port for the RTP packets.
4144 Control address is specific to the linksys type and is used for sending the control
4145 packets unique to them.
projects / asterisk/asterisk.git / blob