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 * Added the ability to pass options to MixMonitor when recording is used with
21 ConfBridge. This includes the addition of the following configuration
22 parameters for the 'bridge' object:
23 - record_file_timestamp: whether or not to append the start time to the
25 - record_options: the options to pass to the MixMonitor application
26 - record_command: a command to execute when recording is finished
27 Note that these options may also be with the CONFBRIDGE function.
35 * The CALLERID(ani2) value for incoming calls is now populated in featdmf
36 signaling mode. The information was previously discarded.
40 * New 'rtpbindaddr' global setting. This allows a user to define which
41 ipaddress to bind the rtpengine to. For example, chan_sip might bind
42 to eth0 (10.0.0.2) but rtpengine to eth1 (192.168.1.10).
43 * DTLS related configuration options can now be set at a general level.
44 Enabling DTLS support, though, requires enabling it at the user
49 * New 'user_eq_phone' endpoint setting. This adds a 'user=phone' parameter
50 to the request URI and From URI if the user is determined to be a phone number.
51 * New 'moh_passthrough' endpoint setting. This will pass hold and unhold requests
52 through using SIP re-invites with sendonly and sendrecv accordingly.
53 * Added the pjsip.conf system type disable_tcp_switch option. The option
54 allows the user to disable switching from UDP to TCP transports described
55 by RFC 3261 section 18.1.1.
56 * New 'line' and 'endpoint' options added on outbound registrations. This allows some
57 identifying information to be added to the Contact of the outbound registration.
58 If this information is present on messages received from the remote server
59 the message will automatically be associated with the configured endpoint on the
60 outbound registration.
65 * The core of Asterisk uses a message bus called "Stasis" to distribute
66 information to internal components. For performance reasons, the message
67 distribution was modified to make use of a thread pool instead of a
68 dedicated thread per consumer in certain cases. The initial settings for
69 the thread pool can now be configured in 'stasis.conf'.
77 * Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for
78 the hold status of a channel.
82 * The transferdialattempts default value has been changed from 1 to 3. The
83 transferinvalidsound has been changed from "pbx-invalid" to "privacy-incorrect".
84 These were changed to make DTMF transfers be more user-friendly by default.
92 * Added sort=randstart to the sort options. It sorts the files by name and
93 then chooses the first file to play at random.
94 * Added preferchannelclass=no option to prefer the application-passed class
95 over the channel-set musicclass. This allows separate hold-music from
96 application (e.g. Queue or Dial) specified music.
98 ------------------------------------------------------------------------------
99 --- Functionality changes from Asterisk 13.1.0 to Asterisk 13.2.0 ------------
100 ------------------------------------------------------------------------------
102 res_pjsip_config_wizard
104 * This is a new module that adds streamlined configuration capability for
105 chan_pjsip. It's targetted at users who have lots of basic configuration
106 scenarios like 'phone' or 'agent' or 'trunk'. Additional information
107 can be found in the sample configuration file at
108 config/samples/pjsip_wizard.conf.sample.
112 * The Originate operation now takes in an originator channel. The linked ID of
113 this originator channel is applied to the newly originated outgoing channel.
114 If using CEL this allows an association to be established between the two so
115 it can be recognized that the originator is dialing the originated channel.
117 * "language" (the default spoken language for the channel) is now included in
118 the standard channel state output for suitable events.
122 * "Language" (the default spoken language for the channel) is now included in
123 the standard channel state output for suitable events.
125 ------------------------------------------------------------------------------
126 --- Functionality changes from Asterisk 13.0.0 to Asterisk 13.1.0 ------------
127 ------------------------------------------------------------------------------
131 * Event NewConnectedLine is emitted when the connected line information on
136 * Event ChannelConnectedLine is emitted when the connected line information
137 on a channel changes.
142 The features.conf general section has three new configurable options:
143 * transferdialattempts
145 * transferinvalidsound
146 For more information on what these options do, see the Asterisk wiki:
147 https://wiki.asterisk.org/wiki/x/W4fAAQ
154 * New 'media_encryption_optimistic' endpoint setting. This will use SRTP
155 when possible but does not consider lack of it a failure.
157 res_pjsip_endpoint_identifer_ip
159 * New CLI commands have been added: "pjsip show identif(y|ies)", which lists
160 all configured PJSIP identify objects
162 ------------------------------------------------------------------------------
163 --- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
164 ------------------------------------------------------------------------------
169 Asterisk 13 is the next Long Term Support (LTS) release of Asterisk. As such,
170 the focus of development for this release of Asterisk was on improving the
171 usability and features developed in the previous Standard release, Asterisk 12.
172 Beyond a general refinement of end user features, development focussed heavily
173 on the Asterisk APIs - the Asterisk Manager Interface (AMI) and the Asterisk
174 REST Interface (ARI) - and the PJSIP stack in Asterisk. Some highlights of the
175 new features include:
177 * Asterisk security events are now provided via AMI, allowing end users to
178 monitor their Asterisk system in real time for security related issues.
179 * External control of Message Waiting Indicators (MWI) through both AMI and ARI.
180 * Reception/transmission of out of call text messages using any supported
181 channel driver/protocol stack through ARI.
182 * Resource List Server support in the PJSIP stack, providing subscriptions to
183 lists of resources and batched delivery of NOTIFY requests.
184 * Inter-Asterisk distributed device state and mailbox state using the PJSIP
187 It is important to note that Asterisk 13 is built on the architecture developed
188 during the previous Standard release, Asterisk 12. Users upgrading to
189 Asterisk 13 should read about the new features in Asterisk 12 later in this file
190 (see Functionality changes from Asterisk 11 to Asterisk 12), as well as the
191 UPGRADE-12.txt delivered with this release. In particular, users upgrading to
192 Asterisk 13 from a release prior to Asterisk 12 should read the specifications
193 on AMI, CDRs, and CEL on the Asterisk wiki:
194 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
195 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
196 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
198 Many new featuers in Asterisk 13 were introduced in point releases of
199 Asterisk 12. Following this section - which documents the changes from all
200 versions of Asterisk 12 to Asterisk 13 - users should examine the new features
201 that were introduced in the point releases of Asterisk 12, as they are also
202 included in Asterisk 13.
204 Finally, all users upgrading to Asterisk 13 should read the UPGRADE.txt file
205 delivered with this release.
210 * Sample config files have been moved from configs/ to a sub-folder of that
213 * The menuselect utility has been pulled into the Asterisk repository. As a
214 result, the libxml2 development library is now a required dependency for
217 * A new Compiler Flag, REF_DEBUG, has been added. When enabled, reference
218 counted objects will emit additional debug information to the refs log file
219 located in the standard Asterisk log file directory. This log file is useful
220 in tracking down object leaks and other reference counting issues. Prior to
221 this version, this option was only available by modifying the source code
222 directly. This change also includes a new script, refcounter.py, in the
223 contrib folder that will process the refs log file. Note that this replaces
224 the refcounter utility that could be built from the utils directory.
232 * This module was deprecated and has been removed. Users of app_dahdibarge
233 should use ChanSpy instead.
237 * New options to play a beep when starting a recording and stopping a recording
238 have been added. The option "p" will play a beep to the channel that starts
239 the recording. The option "P" will play a beep to the channel that stops the
244 * Queue rules can now be stored in a database table, queue_rules. Unlike other
245 RealTime tables, the queue_rules table is only examined on module load or
246 module reload. A new general setting has been added to queuerules.conf,
247 'realtime_rules', which, when set to 'yes', will cause app_queue to look in
248 RealTime for additional queue rules to parse. Note that both the file and
249 the database can be used as a provide of queue rules when 'realtime_rules'
252 When app_queue is reloaded, all rules are re-parsed and loaded into memory.
253 There is no caching of RealTime queue rules.
257 * This module was deprecated and has been removed. Users of app_readfile
258 should use func_env's FILE function instead.
262 * The 'say' family of dialplan applications now support the Japanese
263 language. The 'language' parameter in say.conf now recognizes a setting of
264 'ja', which will enable Japanese language specific mechanisms for playing
265 back numbers, dates, and other items.
269 * This module was deprecated and has been removed. Users of app_saycountpl
270 should use the Say family of applications.
274 * The SetMusicOnHold dialplan application was deprecated and has been removed.
275 Users of the application should use the CHANNEL function's musicclass
280 * The WaitMusicOnHold dialplan application was deprecated and has been
281 removed. Users of the application should use MusicOnHold with a duration
286 * VoiceMail and VoiceMailMain now support the Japanese language. The
287 'language' parameter in voicemail.conf now recognizes a setting of 'ja',
288 which will enable prompts to be played back using a Japanese grammatical
289 structure. Additional prompts are necessary for this functionality,
291 - jb-arimasu: there is
292 - jb-arimasen: there is not
293 - jb-oshitekudasai: please press
299 * Add the ability to specify multiple email addresses in configuration,
308 * This module was deprecated and has been removed. Users of cdr_sqlite
309 should use cdr_sqlite3_custom.
313 * Added the ability to support PostgreSQL application_name on connections.
314 This allows PostgreSQL to display the configured name in the
315 pg_stat_activity view and CSV log entries. This setting is configurable
316 for cdr_pgsql via the appname configuration setting in cdr_pgsql.conf.
324 * Added the ability to support PostgreSQL application_name on connections.
325 This allows PostgreSQL to display the configured name in the
326 pg_stat_activity view and CSV log entries. This setting is configurable
327 for cel_pgsql via the appname configuration setting in cel_pgsql.conf.
335 * SS7 support now requires libss7 v2.0 or later.
337 * Added SS7 support for connected line and redirecting.
339 * Most SS7 CLI commands are reworked as well as new SS7 commands added.
342 * Added several SS7 config option parameters described in
343 chan_dahdi.conf.sample.
347 * This module was deprecated and has been removed. Users of chan_gtalk
348 should use chan_motif.
352 * This module was deprecated and has been removed. Users of chan_h323
353 should use chan_ooh323.
357 * This module was deprecated and has been removed. Users of chan_jingle
358 should use chan_motif.
362 * Added the CLI command 'pjsip list ciphers' so a user can know what
363 OpenSSL names are available on their system for the pjsip.conf cipher
368 * The SIPPEER dialplan function no longer supports using a colon as a
369 delimiter for parameters. The parameters for the function should be
370 delimited using a comma.
372 * The SIPCHANINFO dialplan function was deprecated and has been removed. Users
373 of the function should use the CHANNEL function instead.
381 * Added functional peeraccount support. Except for Queue, the
382 accountcode propagation is now consistently propagated to outgoing
383 channels before dialing. The channel accountcode can change from its
384 original non-empty value on channel creation for the following specific
385 reasons. One, dialplan sets it using CHANNEL(accountcode). Two, an
386 originate method that can specify an accountcode value. Three, the
387 calling channel propagates its peeraccount or accountcode to the
388 outgoing channel's accountcode before dialing. The change has two
389 visible effects. One, local channels now cross accountcode and
390 peeraccount across the special bridge between the ;1 and ;2 channels
391 just like channels between normal bridges. Two, the
392 CHANNEL(peeraccount) value can now be set before Dial and FollowMe to
393 set the accountcode on the outgoing channel(s).
395 For Queue, an outgoing channel's non-empty accountcode will not change
396 unless explicitly set by CHANNEL(accountcode). The change has three
397 visible effects. One, local channels now cross accountcode and
398 peeraccount across the special bridge between the ;1 and ;2 channels
399 just like channels between normal bridges. Two, the queue member will
400 get an accountcode if it doesn't have one and one is available from the
401 calling channel's peeraccount. Three, accountcode propagation includes
402 local channel members where the accountcodes are propagated early
403 enough to be available on the ;2 channel.
407 * New DeviceStateChanged and PresenceStateChanged AMI events have been added.
408 These events are emitted whenever a device state or presence state change
409 occurs. The events are controlled by res_manager_device_state.so and
410 res_manager_presence_state.so. If the high frequency of these events is
411 problematic for you, do not load these modules.
413 * Added DialplanExtensionAdd and DialplanExtensionRemove AMI commands. They
414 work in basically the same way as the 'dialplan add extension' and
415 'dialplan remove extension' CLI commands respectively.
417 * New AMI action LoggerRotate reloads and rotates logger in the same manner
418 as CLI command 'logger rotate'
420 * New AMI Actions FAXSessions, FAXSession, and FAXStats replicate the
421 functionality of CLI commands 'fax show sessions', 'fax show session',
422 and fax show stats' respectively.
424 * New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset
425 enable manager control over PRI debugging levels and file output.
427 * AMI action PJSIPNotify may now send to a URI instead of only to a PJSIP
428 endpoint as long as a default outbound endpoint is set. This also applies
429 to the equivalent CLI command (pjsip send notify)
431 * The AMI action PJSIPShowEndpoint now includes ContactStatusDetail sections
432 that give information on Asterisk's attempts to qualify the endpoint.
434 * The DialEnd event will now contain a Forward header if the dial is ending
435 due to the call being forwarded. The contents of the Forward header is the
436 extension in the number to which the call is being forwarded.
440 * The "bridge_technology" extra field key has been added to BRIDGE_ENTER
441 and BRIDGE_EXIT events.
445 * Channel variables are now substituted in arguments passed to applications
446 run by using dynamic features.
450 * The TLS core in Asterisk now supports Perfect Forward Secrecy (PFS).
451 Enabling PFS is attempted by default, and is dependent on the configuration
452 of the module using TLS.
453 - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
454 specify a ECDHE cipher suite in sip.conf, for example:
455 tlscipher=AES128-SHA:DES-CBC3-SHA
456 - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
457 into the private key file, e.g., sip.conf tlsprivatekey. For example, the
458 default dh2048.pem - see
459 http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt
460 - Because clients expect the server to prefer PFS, and because OpenSSL sorts
461 its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
462 Consider re-ordering your cipher suites in the respective configuration
464 tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
465 will use PFS when offered by the client. Clients which do not offer PFS
466 fall-back to AES-128 (or even 3DES, as recommended by RFC 3261).
474 * The JACK_HOOK function now supports audio with a sample rate higher than
483 * Added the ability to support PostgreSQL application_name on connections.
484 This allows PostgreSQL to display the configured name in the
485 pg_stat_activity view and CSV log entries. This setting is configurable
486 for res_config_pgsql via the dbappname configuration setting in
489 res_pjsip_outbound_publish
491 * A new module, res_pjsip_outbound_publish provides the mechanisms for sending
492 PUBLISH requests for specific event packages to another SIP User Agent.
496 * The publish/subscribe core module has been updated to support RFC 4662
497 Resource Lists, allowing Asterisk to act as a Resource List Server (RLS).
498 Resource lists are configured in pjsip.conf under a new object type,
499 resource_list. Resource lists can contain either message-summary or presence
500 events, and can be composed of specific resources that provide the event or
501 other resource lists.
503 * Inbound publication support is provided by a new object, inbound-publication.
504 This configures res_pjsip_pubsub to accept PUBLISH requests from a particular
505 resource. Which events are accepted is constructed dynamically; see
506 res_pjsip_publish_asterisk for more information.
508 res_pjsip_publish_asterisk
510 * A new module, res_pjsip_publish_asterisk adds support for PUBLISH requests of
511 Asterisk information to other Asterisk servers. This module is intended only
512 for Asterisk to Asterisk exchanges of information. Currently, this includes
513 both mailbox state and device state information.
515 ------------------------------------------------------------------------------
516 --- Functionality changes from Asterisk 12.4.0 to Asterisk 12.5.0 ------------
517 ------------------------------------------------------------------------------
521 * Stored recordings now support a new operation, copy. This will take an
522 existing stored recording and copy it to a new location in the recordings
525 * LiveRecording objects now have three additional fields that can be reported
526 in a RecordingFinished ARI event:
527 - total_duration: the duration of the recording
528 - talking_duration: optional. The duration of talking detected in the
529 recording. This is only available if max_silence_seconds was specified
530 when the recording was started.
531 - silence_duration: optional. The duration of silence detected in the
532 recording. This is only available if max_silence_seconds was specified
533 when the recording was started.
534 Note that all duration values are reported in seconds.
536 * Users of ARI can now send and receive out of call text messages. Messages
537 can be sent directly to a particular endpoint, or can be sent to the
538 endpoints resource directly and inferred from the URI scheme. Text
539 messages are passed to ARI clients as TextMessageReceived events. ARI
540 clients can choose to receive text messages by subscribing to the particular
541 endpoint technology or endpoints that they are interested in.
543 * The applications resource now supports subscriptions to all endpoints of
544 a particular channel technology. For example, subscribing to an eventSource
545 of 'endpoint:PJSIP' will subscribe to all PJSIP endpoints.
549 * The endpoint configuration object now supports 'accountcode'. Any channel
550 created for an endpoint with this setting will have its accountcode set
551 to the specified value.
555 * A new module, res_hep_rtcp, has been added that will forward RTCP call
556 statistics to a HEP capture server. See res_hep for more information.
560 * Function AUDIOHOOK_INHERIT has been deprecated. Audiohooks are now
561 unconditionally inhereted through masquerades. As a side benefit, more
562 than one audiohook of a given type may persist through a masquerade now.
564 ------------------------------------------------------------------------------
565 --- Functionality changes from Asterisk 12.3.0 to Asterisk 12.4.0 ------------
566 ------------------------------------------------------------------------------
570 * Returns new AGENT_STATUS value "NOT_CONNECTED" if the agent fails to
571 connect with an incoming caller after being alerted to the presence
572 of the incoming caller. The most likely reason this would happen is
573 the agent did not acknowledge the call in time.
577 * New events have been added for the TALK_DETECT function. When the function
578 is used on a channel, ChannelTalkingStart/ChannelTalkingStop events will be
579 emitted to connected AMI clients indicating the start/stop of talking on
584 * New event models have been aded for the TALK_DETECT function. When the
585 function is used on a channel, ChannelTalkingStarted/ChannelTalkingFinished
586 events will be emitted to connected WebSockets subscribed to the channel,
587 indicating the start/stop of talking on the channel.
591 * A new function, TALK_DETECT, has been added. When set on a channel, this
592 fucntion causes events indicating the starting/stoping of talking on said
593 channel to be emitted to both AMI and ARI clients.
595 ------------------------------------------------------------------------------
596 --- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------
597 ------------------------------------------------------------------------------
601 * A new Playback URI 'tone' has been added. Tones are specified either as
602 an indication name (e.g. 'tone:busy') from indications.conf or as a tone
603 pattern (e.g. 'tone:240/250,0/250'). Tones differ from normal playback
604 URIs in that they must be stopped manually and will continue to occupy
605 a channel's ARI control queue until they are stopped. They also can not
606 be rewound or fastforwarded.
608 * User events can now be generated from ARI. Events can be signalled with
609 arbitrary json variables, and include one or more of channel, bridge, or
610 endpoint snapshots. An application must be specified which will receive
611 the event message (other applications can subscribe to it). The message
612 will also be delivered via AMI provided a channel is attached. Dialplan
613 generated user event messages are still transmitted via the channel, and
614 will only be received by a stasis application they are attached to or if
615 the channel is subscribed to.
619 * SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI
620 fields for prohibited callingpres information. Values are legacy, no, and
621 yes. By default, legacy is used.
622 trust_id_outbound=legacy - behavior remains the same as 1.8.26.1. When
623 dealing with prohibited callingpres and sendrpid=pai/rpid, RPID/PAI
624 headers are appended to outbound SIP messages just as they are with
625 allowed callingpres values, but data about the remote party's identity is
627 When sendrpid=rpid, only the remote party's domain is anonymized.
628 trust_id_outbound=no - when dealing with prohibited callingpres, RPID/PAI
629 headers are not sent.
630 trust_id_outbound=yes - RPID/PAI headers are applied with the full remote
631 party information in tact even for prohibited callingpres information.
632 In the case of PAI, a Privacy: id header will be appended for prohibited
633 calling information to communicate that the private information should
634 not be relayed to untrusted parties.
638 * Manager action 'Park' now takes an additional argument 'AnnounceChannel'
639 which can be used to announce the parked call's location to an arbitrary
640 channel in a bridge. If 'Channel' and 'TimeoutChannel' are now the two
641 parties in a one to one bridge, 'TimeoutChannel' is treated as having
642 parked 'Channel' like with the Park Call DTMF feature and will receive
643 announcements prior to being hung up.
645 ------------------------------------------------------------------------------
646 --- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
647 ------------------------------------------------------------------------------
651 * Record application now has an option 'o' which allows 0 to act as an exit
652 key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'
655 --------------------------
656 * ChanSpy now accepts a channel uniqueid or a fully specified channel name
657 as the chanprefix parameter if the 'u' option is specified.
660 --------------------------
661 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
662 conference user menus.
664 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
665 menus, bridge settings, and user settings that have been applied by the
666 CONFBRIDGE dialplan function.
668 * The ConfBridge dialplan application now sets a channel variable,
669 CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
670 how a channel exited the conference.
672 * Added conference user option 'announce_join_leave_review'. This option
673 implies 'announce_join_leave' with the added effect that the user will
674 be asked if they want to confirm or re-record the recording of their
675 name when entering the conference
678 --------------------------
679 * At exit, the Directory application now sets a channel variable
680 DIRECTORY_RESULT to one of the following based on the reason for exiting:
681 OPERATOR user requested operator by pressing '0' for operator
682 ASSISTANT user requested assistant by pressing '*' for assistant
683 TIMEOUT user pressed nothing and Directory stopped waiting
684 HANGUP user's channel hung up
685 SELECTED user selected a user from the directory and is routed
686 USEREXIT user pressed '#' from the selection prompt to exit
687 FAILED directory failed in a way that wasn't accounted for. Dang.
691 * Monitor() - A new option, B(), has been added that will turn on a periodic
692 beep while the call is being recorded.
695 --------------------------
696 * MusicOnHold streams (all modes other than "files") now support wide band
700 --------------------------
701 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
702 and for the channel executing Page respectively.
705 --------------------------
706 * PickupChan now accepts channel uniqueids of channels to pickup.
709 --------------------------
710 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
711 to 'true' (case insensitive), then any Say application (SayNumber,
712 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
713 anticipate DTMF. If DTMF is received, these applications will behave like
714 the background application and jump to the received extension once a match
715 is established or after a short period of inactivity.
718 -------------------------
719 * A new function, MIXMONITOR, has been added to allow access to individual
720 instances of MixMonitor on a channel.
722 * A new option, B(), has been added that will turn on a periodic beep while the
723 call is being recorded.
727 -------------------------
730 -------------------------
731 * TEL URI support for inbound INVITE requests has been added. chan_sip will
732 now handle TEL schemes in the Request and From URIs. The phone-context in
733 the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on
738 * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
739 the new AST_SORCERY diaplan function.
741 * Core Show Locks output now includes Thread/LWP ID if the platform
742 supports this feature.
744 * New "logger add channel" and "logger remove channel" CLI commands have
745 been added to allow creation and deletion of dynamic logger channels
746 without configuration changes. These dynamic logger channels will only
747 exist until the next restart of asterisk.
751 * The live recording object on recording events now contains a target_uri
752 field which contains the URI of what is being recorded.
754 * The bridge type used when creating a bridge is now a comma separated list of
755 bridge properties. Valid options are: mixing, holding, dtmf_events, and
758 * A channelId can now be provided when creating a channel, either in the
759 uri (POST channels/my-channel-id) or as query parameter. A local channel
760 will suffix the second channel id with ';2' unless provided as query
761 parameter otherChannelId.
763 * A bridgeId can now be provided when creating a bridge, either in the uri
764 (POST bridges/my-bridge-id) or as a query parameter.
766 * A playbackId can be provided when starting a playback, either in the uri
767 (POST channels/my-channel-id/play/my-playback-id /
768 POST bridges/my-bridge-id/play/my-playback-id) or as a query parameter.
770 * A snoop channel can be started with a snoopId, in the uri or query.
774 * Originate now takes optional parameters ChannelId and OtherChannelId,
775 used to set the UniqueId on creation. The other id is assigned to the
776 second channel when dialing LOCAL, or defaults to appending ;2 if only
777 the single Id is given.
779 * The Mixmonitor action now has a "Command" header that can be used to
780 indicate a post-process command to run once recording finishes.
784 * A new set of Alembic scripts has been added for CDR tables. This will create
785 a 'cdr' table with the default schema that Asterisk expects.
790 * A new function was added: PERIODIC_HOOK. This allows running a periodic
791 dialplan hook on a channel. Any audio generated by this hook will be
792 injected into the call.
800 * A new module, res_hep, has been added, that acts as a generic packet
801 capture agent for the Homer Encapsulation Protocol (HEP) version 3.
802 It can be configured via hep.conf. Other modules can use res_hep to send
803 message traffic to a HEP capture server.
807 * A new module, res_hep_pjsip, has been added that will forward PJSIP
808 message traffic to a HEP capture server. See res_hep for more
813 * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
814 be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
816 * Added the following new CLI commands:
817 - "pjsip show contacts" - list all current PJSIP contacts.
818 - "pjsip show contact" - show specific information about a current PJSIP
820 - "pjsip show channel" - show detailed information about a PJSIP channel.
824 * A new module, res_pjsip_multihomed handles situations where the system
825 Asterisk is running out has multiple interfaces. res_pjsip_multihomed
826 determines which interface should be used during message sending.
828 res_pjsip_pidf_digium_body_supplement
830 * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
831 request body formatting for presence support in Digium phones.
833 res_pjsip_send_to_voicemail
835 * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
836 particular headers to transfer a PJSIP channel directly to a particular
837 extension that has VoiceMail. This is intended to be used with Digium
838 phones that support this feature.
840 res_pjsip_outbound_registration
842 * A new CLI command has been added: "pjsip show registrations", which lists
843 all configured PJSIP registrations
846 ------------------------------------------------------------------------------
847 --- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
848 ------------------------------------------------------------------------------
852 * Added a new module that provides AMI control over MWI within Asterisk,
853 res_mwi_external_ami. Note that this module depends on res_mwi_external;
854 for more information on enabling this module, see res_mwi_external.
855 This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
856 the MWIGet/MWIGetComplete events.
858 * The DialStatus field in the DialEnd event can now contain additional
859 statuses that convey how the dial operation terminated. This includes
860 ABORT, CONTINUE, and GOTO.
862 * AMI will now emit security events. A new class authorization has been
863 added in manager.conf for the security events, 'security'. The new events
865 - FailedACL - raised when a request violates an ACL check
866 - InvalidAccountID - raised when a request fails an authentication
867 check due to an invalid account ID
868 - SessionLimit - raised when a request fails due to exceeding the
869 number of allowed concurrent sessions for a service
870 - MemoryLimit - raised when a request fails due to an internal memory
872 - LoadAverageLimit - raised when a request fails because a configured
873 load average limit has been reached
874 - RequestNotAllowed - raised when a request is not allowed by
876 - AuthMethodNotAllowed - raised when a request used an authentication
877 method not allowed by the service
878 - RequestBadFormat - raised when a request is received with bad formatting
879 - SuccessfulAuth - raised when a request successfully authenticates
880 - UnexpectedAddress - raised when a request has a different source address
881 then what is expected for a session already in progress with a service
882 - ChallengeResponseFailed - raised when a request's attempt to authenticate
883 has been challenged, and the request failed the authentication challenge
884 - InvalidPassword - raised when a request provides an invalid password
885 during an authentication attempt
886 - ChallengeSent - raised when an Asterisk service send an authentication
887 challenge to a request
888 - InvalidTransport - raised when a request attempts to use a transport not
889 allowed by the Asterisk service
891 * Bridge related events now have two additional fields: BridgeName and
892 BridgeCreator. BridgeName is a descriptive name for the bridge;
893 BridgeCreator is the name of the entity that created the bridge. This
894 affects the following events: ConfbridgeStart, ConfbridgeEnd,
895 ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
896 ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
897 AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
901 * The Bridge data model now contains the additional fields 'name' and
902 'creator'. The 'name' field conveys a descriptive name for the bridge;
903 the 'creator' field conveys the name of the entity that created the bridge.
904 This affects all responses to HTTP requests that return a Bridge data model
905 as well as all event derived data models that contain a Bridge data model.
906 The POST /bridges operation may now optionally specify a name to give to
907 the bridge being created.
909 * Added a new ARI resource 'mailboxes' which allows the creation and
910 modification of mailboxes managed by external MWI. Modules res_mwi_external
911 and res_stasis_mailbox must be enabled to use this resource. For more
912 information on external MWI control, see res_mwi_external.
914 * Added new events for externally initiated transfers. The event
915 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
916 of a bridge in the ARI controlled application to the dialplan; the
917 BridgeAttendedTransfer event is raised when a channel initiates an
918 attended transfer of a bridge in the ARI controlled application to the
921 * Channel variables may now be specified as a body parameter to the
922 POST /channels operation. The 'variables' key in the JSON is interpreted
923 as a sequence of key/value pairs that will be added to the created channel
924 as channel variables. Other parameters in the JSON body are treated as
925 query parameters of the same name.
929 * Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be
930 automatically handled by the HTTP server if a request is received with a
931 Transfer-Encoding type of "chunked".
935 * Path support has been added with the 'support_path' option in registration
938 * A 'debug' option has been added to the globals section that will allow
939 sip messages to be logged.
941 * A 'set_var' option has been added to endpoints that will automatically
942 set the desired variable(s) on a channel created for that endpoint.
944 * Several new tables and columns have been added to the realtime schema for
945 the res_pjsip related modules. See the UPGRADE.txt notes for updating
950 * A new module, res_mwi_external, has been added to Asterisk. This module
951 acts as a base framework that other modules can build on top of to allow
952 an external system to control MWI within Asterisk. For implementations
953 that make use of res_mwi_external, see res_mwi_external_ami and
954 res_ari_mailboxes. Note that res_mwi_external canflicts with other modules
955 that may produce MWI themselves, such as app_voicemail. res_mwi_external
956 and other modules that depend on it cannot be built or loaded with
957 app_voicemail present.
961 * DNS functionality will now automatically be enabled if the system configured
962 nameservers can be retrieved. If the system configured nameservers can not be
963 retrieved the functionality will resort to using system resolution. Functionalty
964 such as SRV records and failover will not be available if system resolution
967 ------------------------------------------------------------------------------
968 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
969 ------------------------------------------------------------------------------
974 Asterisk 12 is a standard release of the Asterisk project. As such, the
975 focus of development for this release was on core architectural changes and
976 major new features. This includes:
977 * A more flexible bridging core based on the Bridging API
978 * A new internal message bus, Stasis
979 * Major standardization and consistency improvements to AMI
980 * Addition of the Asterisk RESTful Interface (ARI)
981 * A new SIP channel driver, chan_pjsip
982 In addition, as the vast majority of bridging in Asterisk was migrated to the
983 Bridging API used by ConfBridge, major changes were made to most of the
984 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
986 Specifications have been written for the affected interfaces. These
987 specifications are available on the Asterisk wiki:
988 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
989 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
990 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
992 It is *highly* recommended that anyone migrating to Asterisk 12 read the
993 information regarding its release both in this file and in the accompanying
994 UPGRADE.txt file. More detailed information on the major changes can be found
995 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
1000 * Added build option DISABLE_INLINE. This option can be used to work around a
1001 bug in gcc. For more information, see
1002 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
1004 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
1005 the CHANNEL_TRACE build option were incompatible with the new bridging
1008 * Asterisk now optionally uses libxslt to improve XML documentation generation
1009 and maintainability. If libxslt is not available on the system, some XML
1010 documentation will be incomplete.
1012 * Asterisk now depends on libjansson. If a package of libjansson is not
1013 available on your distro, please see http://www.digip.org/jansson/.
1015 * Asterisk now depends on libuuid and, optionally, uriparser. It is
1016 recommended that you install uriparser, even if it is optional.
1018 * The new SIP stack and channel driver uses a particular version of PJSIP.
1019 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
1020 configuring and installing PJSIP for usage with Asterisk.
1022 * Optional API was re-implemented to be more portable, and no longer requires
1023 weak reference support from the compiler. The build option OPTIONAL_API may
1024 be disabled to disable Optional API support.
1031 * Along with AgentRequest, this application has been modified to be a
1032 replacement for chan_agent. The act of a channel calling the AgentLogin
1033 application places the channel into a pool of agents that can be
1034 requested by the AgentRequest application. Note that this application, as
1035 well as all other agent related functionality, is now provided by the
1036 app_agent_pool module. See chan_agent and AgentRequest for more information.
1038 * This application no longer performs agent authentication. If authentication
1039 is desired, the dialplan needs to perform this function using the
1040 Authenticate or VMAuthenticate application or through an AGI script before
1043 * If this application is called and the agent is already logged in, the
1044 dialplan will continue exection with the AGENT_STATUS channel variable set
1045 to ALREADY_LOGGED_IN.
1047 * The agents.conf schema has changed. Rather than specifying agents on a
1048 single line in comma delineated fashion, each agent is defined in a separate
1049 context. This allows agents to use the power of context templates in their
1052 * A number of parameters from agents.conf have been removed. This includes
1053 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
1054 urlprefix, and savecallsin. These options were obsoleted by the move from
1055 a channel driver model to the bridging/application model provided by
1060 * A new application, this will request a logged in agent from the pool and
1061 bridge the requested channel with the channel calling this application.
1062 Logged in agents are those channels that called the AgentLogin application.
1063 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
1064 application will be set with an appropriate error value.
1066 AgentMonitorOutgoing
1068 * This application has been removed. It was a holdover from when
1069 AgentCallbackLogin was removed.
1073 * Added support for additional Ademco DTMF signalling formats, including
1074 Express 4+1, Express 4+2, High Speed and Super Fast.
1076 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
1077 call time, in milliseconds, to run the application.
1079 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
1080 maximum number of times to retry the call.
1082 * Added a new configuration option answait. If set, the AlarmReceiver
1083 application will wait the number of milliseconds specified by answait
1084 after the channel has answered. Valid values range between 500
1085 milliseconds and 10000 milliseconds.
1087 * Added configuration option no_group_meta. If enabled, grouping of metadata
1088 information in the AlarmReceiver log file will be skipped.
1092 * It is now no longer possible to bypass updating the CDR on the channel
1093 when answering. CDRs reflect the state of the channel and will always
1094 reflect the time they were Answered.
1098 * A new application in Asterisk, this will place the calling channel
1099 into a holding bridge, optionally entertaining them with some form of
1100 media. Channels participating in a holding bridge do not interact with
1101 other channels in the same holding bridge. Optionally, however, a channel
1102 may join as an announcer. Any media passed from an announcer channel is
1103 played to all channels in the holding bridge. Channels leave a holding
1104 bridge either when an optional timer expires, or via the ChannelRedirect
1105 application or AMI Redirect action.
1109 * All participants in a bridge can now be kicked out of a conference room
1110 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
1111 command, i.e., 'confbridge kick <conference> all'
1113 * CLI output for the 'confbridge list' command has been improved. When
1114 displaying information about a particular bridge, flags will now be shown
1115 for the participating users indicating properties of that user.
1117 * The ConfbridgeList event now contains the following fields: WaitMarked,
1118 EndMarked, and Waiting. This displays additional properties about the
1119 user's profile, as well as whether or not the user is waiting for a
1120 Marked user to enter the conference.
1122 * Added a new option for conference recording, record_file_append. If enabled,
1123 when the recording is stopped and then re-started, the existing recording
1124 will be used and appended to.
1126 * ConfBridge now has the ability to set the language of announcements to the
1127 conference. The language can be set on a bridge profile in confbridge.conf
1128 or by the dialplan function CONFBRIDGE(bridge,language)=en.
1132 * The channel variable CPLAYBACKSTATUS may now return the value
1133 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
1134 such as AMI. See the AMI action ControlPlayback for more information.
1138 * Added the 'a' option, which allows the caller to enter in an additional
1139 alias for the user in the directory. This option must be used in conjunction
1140 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
1141 specified in voicemail.conf.
1145 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
1146 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
1147 containing the unique ID of the bridge that the channel happens to be in.
1151 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
1152 for more information.
1154 * Variables are no longer purged from the original CDR. See the 'v' option for
1157 * The 'A' option has been removed. The Answer time on a CDR is never updated
1160 * The 'd' option has been removed. The disposition on a CDR is a function of
1161 the state of the channel and cannot be altered.
1163 * The 'D' option has been removed. Who the Party B is on a CDR is a function
1164 of the state of the respective channels involved in the CDR and cannot be
1167 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
1168 such that the start time and, if applicable, the answer time was updated.
1169 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
1170 'r' option now triggers the Reset, setting the start time (and answer time
1171 if applicable) to the current time. Note that the 'a' option still sets
1172 the answer time to the current time if the channel was already answered.
1174 * The 's' option has been removed. A variable can be set on the original CDR
1175 if desired using the CDR function, and removed from a forked CDR using the
1178 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
1179 longer applies in the CDR engine.
1181 * The 'v' option now prevents the copy of the variables from the original CDR
1182 to the forked CDR. Previously the variables were always copied but were
1183 removed from the original. This was changed as removing variables from a CDR
1184 can have unintended side effects - this option allows the user to prevent
1185 propagation of variables from the original to the forked without modifying
1190 * Added the 'n' option to MeetMe to prevent application of the DENOISE
1191 function to a channel joining a conference. Some channel drivers that vary
1192 the number of audio samples in a voice frame will experience significant
1193 quality problems if a denoiser is attached to the channel; this option gives
1194 them the ability to remove the denoiser without having to unload func_speex.
1198 * The 'b' option now includes conferences as well as sounds played to the
1201 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
1202 running during a transfer. If a MixMonitor is started on a channel,
1203 the MixMonitor will continue to record the audio passing through the
1204 channel even in the presence of transfers.
1208 * The NoCDR application is deprecated. Please use the CDR_PROP function to
1211 * While the NoCDR application will prevent CDRs for a channel from being
1212 propagated to registered CDR backends, it will not prevent that data from
1213 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
1214 function that enables CDRs on a channel will restore those records that have
1215 not yet been finalized.
1219 * The app_parkandannounce module has been removed. The application
1220 ParkAndAnnounce is now provided by the res_parking module. See the
1221 res_parking changes for more information.
1225 * Added queue available hint. The hint can be added to the dialplan using the
1226 following syntax: exten,hint,Queue:{queue_name}_avail
1227 For example, if the name of the queue is 'markq':
1228 exten => 8501,hint,Queue:markq_avail
1229 This will report 'InUse' if there are no logged in agents or no free agents.
1230 It will report 'Idle' when an agent is free.
1232 * Queues now support a hint for member paused state. The hint uses the form
1233 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
1234 are the name of the queue and the name of the member to subscribe to,
1235 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
1236 Members will show as In Use when paused.
1238 * The configuration options eventwhencalled and eventmemberstatus have been
1239 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
1240 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
1241 sent. The "Variable" fields will also no longer exist on the Agent* events.
1242 These events can be filtered out from a connected AMI client using the
1243 eventfilter setting in manager.conf.
1245 * The queue log now differentiates between blind and attended transfers. A
1246 blind transfer will result in a BLINDTRANSFER message with the destination
1247 context and extension. An attended transfer will result in an
1248 ATTENDEDTRANSFER message. This message will indicate the method by which
1249 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
1250 for running an application on a bridge or channel, or "LINK" for linking
1251 two bridges together with local channels. The queue log will also now detect
1252 externally initiated blind and attended transfers and record the transfer
1255 * When performing queue pause/unpause on an interface without specifying an
1256 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
1257 least one member of any queue exists for that interface.
1259 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
1260 for realtime queue log entries.
1264 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
1265 CDRs when they were previously disabled on a channel.
1267 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
1268 backends occurs on an as-needed basis in order to preserve linkedid
1269 propagation and other needed behavior.
1273 * A new application, this is similar to SayAlpha except that it supports
1274 case sensitive playback of the specified characters. For example,
1275 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
1279 * This application is deprecated in favor of CHANNEL(amaflags).
1283 * The SendDTMF application will now accept 'W' as valid input. This will cause
1284 the application to delay one second while streaming DTMF.
1288 * A new application in Asterisk 12, this hands control of the channel calling
1289 the application over to an external system. Currently, external systems
1290 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
1294 * UserEvent will now handle duplicate keys by overwriting the previous value
1295 assigned to the key.
1297 * In addition to AMI, UserEvent invocations will now be distributed to any
1298 interested Stasis applications.
1302 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1303 system as mailbox@context. The rest of the system cannot add @default
1304 to mailbox identifiers for app_voicemail that do not specify a context
1305 any longer. It is a mailbox identifier format that should only be
1306 interpreted by app_voicemail.
1308 * The voicemail.conf configuration file now has an 'alias' configuration
1309 parameter for use with the Directory application. The voicemail realtime
1310 database table schema has also been updated with an 'alias' column.
1315 * Pass through support has been added for both VP8 and Opus.
1317 * Added format attribute negotiation for the Opus codec. Format attribute
1318 negotiation is provided by the res_format_attr_opus module.
1323 * Masquerades as an operation inside Asterisk have been effectively hidden
1324 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
1325 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
1326 dropping of frame/audio hooks, and other internal implementation details
1327 that users had to deal with. This fundamental change has large implications
1328 throughout the changes documented for this version. For more information
1329 about the new core architecture of Asterisk, please see the Asterisk wiki.
1331 * Multiple parties in a bridge may now be transferred. If a participant in a
1332 multi-party bridge initiates a blind transfer, a Local channel will be used
1333 to execute the dialplan location that the transferer sent the parties to. If
1334 a participant in a multi-party bridge initiates an attended transfer,
1335 several options are possible. If the attended transfer results in a transfer
1336 to an application, a Local channel is used. If the attended transfer results
1337 in a transfer to another channel, the resulting channels will be merged into
1340 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
1341 driver specific. If the channel variable is set on the transferrer channel,
1342 the sound will be played to the target of an attended transfer.
1344 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
1345 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
1346 listed. Any more peers in the bridge will not be included in the list.
1347 BRIDGEPEER is not valid in holding bridges like parking since those channels
1348 do not talk to each other even though they are in a bridge.
1350 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
1351 and will contain a value if the BRIDGEPEER's channel driver supports it.
1353 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
1354 was responsible for an attended transfer in a similar fashion to
1357 * Modules using the Configuration Framework or Sorcery must have XML
1358 configuration documentation. This configuration documentation is included
1359 with the rest of Asterisk's XML documentation, and is accessible via CLI
1360 commands. See the CLI changes for more information.
1362 AMI (Asterisk Manager Interface)
1364 * Major changes were made to both the syntax as well as the semantics of the
1365 AMI protocol. In particular, AMI events have been substantially improved
1366 in this version of Asterisk. For more information, please see the AMI
1367 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
1369 * AMI events that reference a particular channel or bridge will now always
1370 contain a standard set of fields. When multiple channels or bridges are
1371 referenced in an event, fields for at least some subset of the channels
1372 and bridges in the event will be prefixed with a descriptive name to avoid
1373 name collisions. See the AMI event documentation on the Asterisk wiki for
1376 * The CLI command 'manager show commands' no longer truncates command names
1377 longer than 15 characters and no longer shows authorization requirement
1378 for commands. 'manager show command' now displays the privileges needed
1379 for using a given manager command instead.
1381 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
1382 peer in its response if the peer has a subscribe context set.
1384 * The SIPqualifypeer action now acknowledges the request once it has
1385 established that the request is against a known peer. It also issues a new
1386 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
1388 * The PlayDTMF action now supports an optional 'Duration' parameter. This
1389 specifies the duration of the digit to be played, in milliseconds.
1391 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
1392 updates when changes occur instead of requiring the use of pollmailboxes.
1394 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
1395 AMI client to manipulate audio currently being played back on a channel. The
1396 supported operations depend on the application being used to send audio to
1397 the channel. When the audio playback was initiated using the ControlPlayback
1398 application or CONTROL STREAM FILE AGI command, the audio can be paused,
1399 stopped, restarted, reversed, or skipped forward. When initiated by other
1400 mechanisms (such as the Playback application), the audio can be stopped,
1401 reversed, or skipped forward.
1403 * Channel related events now contain a snapshot of channel state, adding new
1404 fields to many of these events.
1406 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
1407 in a future release. Please use the common 'Exten' field instead.
1409 * The AMI event 'UserEvent' from app_userevent now contains the channel state
1410 fields. The channel state fields will come before the body fields.
1412 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
1413 'UnParkedCall' have changed significantly in the new res_parking module.
1415 The 'Channel' and 'From' headers are gone. For the channel that was parked
1416 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
1417 has a number of fields associated with it. The old 'Channel' header relayed
1418 the same data as the new 'ParkeeChannel' header.
1420 The 'From' field was ambiguous and changed meaning depending on the event.
1421 for most of these, it was the name of the channel that parked the call
1422 (the 'Parker'). There is no longer a header that provides this channel name,
1423 however the 'ParkerDialString' will contain a dialstring to redial the
1424 device that parked the call.
1426 On UnParkedCall events, the 'From' header would instead represent the
1427 channel responsible for retrieving the parkee. It receives a channel
1428 snapshot labeled 'Retriever'. The 'from' field is is replaced with
1431 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
1433 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
1434 fashion has changed the field names 'StartExten' and 'StopExten' to
1435 'StartSpace' and 'StopSpace' respectively.
1437 * The deprecated use of | (pipe) as a separator in the channelvars setting in
1438 manager.conf has been removed.
1440 * Channel Variables conveyed with a channel no longer contain the name of the
1441 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
1442 ChanVariable: bar=baz. When multiple channels are present in a single AMI
1443 event, the various ChanVariable fields will contain a suffix that specifies
1444 which channel they correspond to.
1446 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
1447 event always conveys the AMI event for a particular channel.
1449 * All 'Reload' events have been consolidated into a single event type. This
1450 event will always contain a Module field specifying the name of the module
1451 and a Status field denoting the result of the reload. All modules now issue
1452 this event when being reloaded.
1454 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
1455 fail to receive this event due to being connected after modules have loaded.
1456 AMI connections that want to know when Asterisk is ready should listen for
1457 the 'FullyBooted' event.
1459 * app_fax now sends the same send fax/receive fax events as res_fax. The
1460 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
1461 now the 'ReceiveFAX' event.
1463 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
1464 'MusicOnHoldStop'. The sub type field has been removed.
1466 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
1467 carrier for another protocol.
1469 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
1470 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
1471 to the specific channel. 'Both' may be specified to play a tone to both
1472 channels. The old 'yes' option is still accepted as a way of playing the
1473 tone to Channel2 only.
1475 * The AMI 'Status' response event to the AMI Status action replaces the
1476 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
1477 indicate what bridge the channel is currently in.
1479 * The AMI 'Hold' event has been moved out of individual channel drivers, into
1480 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
1483 * The AMI events in app_queue have been made more consistent with each other.
1484 Events that reference channels (QueueCaller* and Agent*) will show
1485 information about each channel. The (infamous) 'Join' and 'Leave' AMI
1486 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
1488 * The 'MCID' AMI event now publishes a channel snapshot when available and
1489 its non-channel-snapshot parameters now use either the "MCallerID" or
1490 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
1491 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
1492 parameters in the channel snapshot.
1494 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
1495 'AgentLogin' and 'AgentLogoff' respectively.
1497 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
1498 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
1500 * 'ChannelUpdate' events have been removed.
1502 * All AMI events now contain a 'SystemName' field, if available.
1504 * Local channel optimization is now conveyed in two events:
1505 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
1506 when the Local channel driver begins attempting to optimize itself out of
1507 the media path; the End event is sent after the channel halves have
1508 successfully optimized themselves out of the media path.
1510 * Local channel information in events is now prefixed with 'LocalOne' and
1511 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
1512 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
1513 and 'LocalOptimizationEnd' events.
1515 * The option 'allowmultiplelogin' can now be set or overriden in a particular
1516 account. When set in the general context, it will act as the default
1517 setting for defined accounts.
1519 * The 'BridgeAction' event was removed. It technically added no value, as the
1520 Bridge Action already receives confirmation of the bridge through a
1521 successful completion Event.
1523 * The 'BridgeExec' events were removed. These events duplicated the events that
1524 occur in the Briding API, and are conveyed now through BridgeCreate,
1525 BridgeEnter, and BridgeLeave events.
1527 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
1528 previous versions. They now report all SR/RR packets sent/received, and
1529 have been restructured to better reflect the data sent in a SR/RR. In
1530 particular, the event structure now supports multiple report blocks.
1532 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
1533 raised when a blind transfer/attended transfer completes successfully.
1534 They contain information about the transfer that just completed, including
1535 the location of the transfered channel.
1537 * Added a 'security' class to AMI which outputs the required fields for
1538 security messages similar to the log messages from res_security_log
1540 * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
1541 that describes the status value in a human readable string.
1543 CDR (Call Detail Records)
1545 * Significant changes have been made to the behavior of CDRs. The CDR engine
1546 was effectively rewritten and built on the Stasis message bus. For a full
1547 definition of CDR behavior in Asterisk 12, please read the specification
1548 on the Asterisk wiki (wiki.asterisk.org).
1550 * CDRs will now be created between all participants in a bridge. For each
1551 pair of channels in a bridge, a CDR is created to represent the path of
1552 communication between those two endpoints. This lets an end user choose who
1553 to bill for what during bridge operations with multiple parties.
1555 * The duration, billsec, start, answer, and end times now reflect the times
1556 associated with the current CDR for the channel, as opposed to a cumulative
1557 measurement of all CDRs for that channel.
1559 * When a CDR is dispatched, user defined CDR variables from both parties are
1560 included in the resulting CDR. If both parties have the same variable, only
1561 the Party A value is provided.
1563 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
1564 information regarding the CDR engine is logged as verbose messages. This
1565 option should only be used if the behavior of the CDR engine needs to be
1568 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
1569 normally configured in cdr.conf.
1571 * Added CLI command 'cdr show active {channel}'. When {channel} is not
1572 specified, this command provides a summary of the channels with CDR
1573 information and their statistics. When {channel} is specified, it shows
1574 detailed information about all records associated with {channel}.
1576 CEL (Channel Event Logging)
1578 * CEL has undergone significant rework in Asterisk 12, and is now built on the
1579 Stasis message bus. Please see the specification for CEL on the Asterisk
1580 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
1583 * The 'extra' field of all CEL events that use it now consists of a JSON blob
1584 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
1586 * BLINDTRANSFER events now report the transferee bridge unique
1587 identifier, extension, and context in a JSON blob as the extra string
1588 instead of the transferee channel name as the peer.
1590 * ATTENDEDTRANSFER events now report the peer as NULL and additional
1591 information in the 'extra' string as a JSON blob. For transfers that occur
1592 between two bridged channels, the 'extra' JSON blob contains the primary
1593 bridge unique identifier, the secondary channel name, and the secondary
1594 bridge unique identifier. For transfers that occur between a bridged channel
1595 and a channel running an app, the 'extra' JSON blob contains the primary
1596 bridge unique identifier, the secondary channel name, and the app name.
1598 * LOCAL_OPTIMIZE events have been added to convey local channel
1599 optimizations with the record occurring for the semi-one channel and
1600 the semi-two channel name in the peer field.
1602 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
1603 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
1604 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
1605 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
1606 regardless of whether or not that bridge happens to contain multiple
1611 * When compiled with '--enable-dev-mode', the astobj2 library will now add
1612 several CLI commands that allow for inspection of ao2 containers that
1613 register themselves with astobj2. The CLI commands are 'astobj2 container
1614 dump', 'astobj2 container stats', and 'astobj2 container check'.
1616 * Added specific CLI commands for bridge inspection. This includes 'bridge
1617 show all', which lists all bridges in the system, and 'bridge show {id}',
1618 which provides specific information about a bridge.
1620 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
1621 ejecting the channels currently in the bridge. If the channels cannot
1622 continue in the dialplan or application that put them in the bridge, they
1625 * Added command 'bridge kick'. This will eject a single channel from a bridge.
1627 * Added commands to inspect and manipulate the registered bridge technologies.
1628 This include 'bridge technology show', which lists the registered bridge
1629 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
1630 which controls whether or not a registered bridge technology can be used
1631 during smart bridge operations. If a technology is suspended, it will not
1632 be used when a bridge technology is picked for channels; when unsuspended,
1633 it can be used again.
1635 * The command 'config show help {module} {type} {option}' will show
1636 configuration documentation for modules with XML configuration
1637 documentation. When {module}, {type}, and {option} are omitted, a listing
1638 of all modules with registered documentation is displayed. When {module}
1639 is specified, a listing of all configuration types for that module is
1640 displayed, along with their synopsis. When {module} and {type} are
1641 specified, a listing of all configuration options for that type are
1642 displayed along with their synopsis. When {module}, {type}, and {option}
1643 are specified, detailed information for that configuration option is
1646 * Added 'core show sounds' and 'core show sound' CLI commands. These display
1647 a listing of all installed media sounds available on the system and
1648 detailed information about a sound, respectively.
1650 * 'xmldoc dump' has been added. This CLI command will dump the XML
1651 documentation DOM as a string to the specified file. The Asterisk core
1652 will populate certain XML elements pulled from the source files with
1653 additional run-time information; this command lets a user produce the
1654 XML documentation with all information.
1658 * Parking has been pulled from core and placed into a separate module called
1659 res_parking. See Parking changes below for more details. Configuration for
1660 parking should now be performed in res_parking.conf. Configuration for
1661 parking in features.conf is now unsupported.
1663 * Core attended transfers now have several new options. While performing an
1664 attended transfer, the transferer now has the following options:
1665 - *1 - cancel the attended transfer (configurable via atxferabort)
1666 - *2 - complete the attended transfer, dropping out of the call
1667 (configurable via atxfercomplete)
1668 - *3 - complete the attended transfer, but stay in the call. This will turn
1669 the call into a multi-party bridge (configurable via atxferthreeway)
1670 - *4 - swap to the other party. Once an attended transfer has begun, this
1671 options may be used multiple times (configurable via atxferswap)
1673 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
1674 must be on the channel initiating the transfer to have any effect.
1676 * The BRIDGE_FEATURES channel variable would previously only set features for
1677 the calling party and would set this feature regardless of whether the
1678 feature was in caps or in lowercase. Use of a caps feature for a letter
1679 will now apply the feature to the calling party while use of a lowercase
1680 letter will apply that feature to the called party.
1682 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
1684 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
1685 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
1686 activated the dynamic feature.
1688 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
1689 only on the channel executing the dynamic feature. Executing a dynamic
1690 feature on the bridge peer in a multi-party bridge will execute it on all
1691 peers of the activating channel.
1693 * You can now have the settings for a channel updated using the FEATURE()
1694 and FEATUREMAP() functions inherited to child channels by setting
1695 FEATURE(inherit)=yes.
1697 * automixmon now supports additional channel variables from automon including:
1698 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
1699 and TOUCH_MIXMONITOR_MESSAGE_STOP
1701 * A new general features.conf option 'recordingfailsound' has been added which
1702 allowssetting a failure sound for a user tries to invoke a recording feature
1703 such as automon or automixmon and it fails.
1705 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
1706 features.c for atxferdropcall=no to work properly. This option now just
1711 * Added log rotation strategy 'none'. If set, no log rotation strategy will
1712 be used. Given that this can cause the Asterisk log files to grow quickly,
1713 this option should only be used if an external mechanism for log management
1718 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
1719 will store the path information for that peer when it registers. Realtime
1720 tables can also use the 'supportpath' field to enable Path header support.
1722 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
1723 objectIdentifier. This maps to the supportpath option in sip.conf.
1727 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
1728 provides modules a useful abstraction on top of the many storage mechanisms
1729 in Asterisk, including the Asterisk Database, static configuration files,
1730 static Realtime, and dynamic Realtime. It also provides a caching service.
1731 Users can configure a hierarchy of data storage layers for specific modules
1734 * All future modules which utilize Sorcery for object persistence must have a
1735 column named "id" within their schema when using the Sorcery realtime module.
1736 This column must be able to contain a string of up to 128 characters in length.
1738 Security Events Framework
1740 * Security Event timestamps now use ISO 8601 formatted date/time instead of
1741 the "seconds-microseconds" format that it was using previously.
1745 * The Stasis message bus is a publish/subscribe message bus internal to
1746 Asterisk. Many services in Asterisk are built on the Stasis message bus,
1747 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
1748 Stasis can be configured in stasis.conf. Note that these parameters operate
1749 at a very low level in Asterisk, and generally will not require changes.
1753 * When a channel driver is configured to enable jiterbuffers, they are now
1754 applied unconditionally when a channel joins a bridge. If a jitterbuffer
1755 is already set for that channel when it enters, such as by the JITTERBUFFER
1756 function, then the existing jitterbuffer will be used and the one set by
1757 the channel driver will not be applied.
1761 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
1762 dialplan applications provided by the app_agent_pool module. Agents are
1763 connected with callers using the new AgentRequest dialplan application.
1764 The Agents:<agent-id> device state is available to monitor the status of an
1765 agent. See agents.conf.sample for valid configuration options.
1767 * The updatecdr option has been removed. Altering the names of channels on a
1768 CDR is not supported - the name of the channel is the name of the channel,
1769 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
1770 has also been removed, for the same reason.
1772 * The endcall and enddtmf configuration options are removed. Use the
1773 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
1774 channel before calling AgentLogin.
1778 * chan_bridge has been removed. Its functionality has been incorporated
1779 directly into the ConfBridge application itself.
1783 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
1784 of the specified span and its B-channels. Note that this command should
1785 only be used if you understand the risks it entails.
1787 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
1788 A range of channels can be specified to be destroyed. Note that this command
1789 should only be used if you understand the risks it entails.
1791 * Added the CLI command 'dahdi create channels'. A range of channels can be
1792 specified to be created, or the keyword 'new' can be used to add channels
1795 * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
1796 the exact configured mailbox name. For app_voicemail mailboxes this is
1799 * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
1803 * IPv6 support has been added. We are now able to bind to and
1804 communicate using IPv6 addresses.
1808 * The /b option has been removed.
1810 * chan_local moved into the system core and is no longer a loadable module.
1814 * Added general support for busy detection.
1816 * Added ECAM command support for Sony Ericsson phones.
1820 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
1821 SIP stack. A collection of resource modules provides the bulk of the SIP
1822 functionality. For more information on the new SIP channel driver, see
1823 https://wiki.asterisk.org/wiki/x/JYGLAQ
1827 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
1828 using the 'supportpath' setting, either on a global basis or on a peer basis.
1829 This setting enables Asterisk to route outgoing out-of-dialog requests via a
1830 set of proxies by using a pre-loaded route-set defined by the Path headers in
1831 the REGISTER request. See Realtime updates for more configuration information.
1833 * The SIP_CODEC family of variables may now specify more than one codec. Each
1834 codec must be separated by a comma. The first codec specified is the
1835 preferred codec for the offer. This allows a dialplan writer to specify both
1836 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
1838 * The 'callevents' parameter has been removed. Hold AMI events are now raised
1839 in the core, and can be filtered out using the 'eventfilter' parameter
1842 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
1843 codecs configured for a peer instead of the requested codec.
1845 * The option "register_retry_403" has been added to chan_sip to work around
1846 servers that are known to erroneously send 403 in response to valid
1847 REGISTER requests and allows Asterisk to continue attepmting to connect.
1851 * Added the 'immeddialkey' parameter. If set, when the user presses the
1852 configured key the already entered number will be immediately dialed. This
1853 is useful when the dialplan allows for variable length pattern matching.
1854 Valid options are '*' and '#'.
1856 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
1857 milliseconds) before a call forward is considered to not be answered.
1859 * The 'serviceurl' parameter allows Service URLs to be attached to line
1868 * The password option has been disabled, as the AgentLogin application no
1869 longer provides authentication.
1873 * Due to changes in the Asterisk core, this function is no longer needed to
1874 preserve a MixMonitor on a channel during transfer operations and dialplan
1875 execution. It is effectively obsolete.
1879 * The 'amaflags' and 'accountcode' attributes for the CDR function are
1880 deprecated. Use the CHANNEL function instead to access these attributes.
1882 * The 'l' option has been removed. When reading a CDR attribute, the most
1883 recent record is always used. When writing a CDR attribute, all non-finalized
1886 * The 'r' option has been removed, for the same reason as the 'l' option.
1888 * The 's' option has been removed, as LOCKED semantics no longer exist in the
1893 * A new function CDR_PROP has been added. This function lets you set properties
1894 on a channel's active CDRs. This function is write-only. Properties accept
1895 boolean values to set/clear them on the channel's CDRs. Valid properties
1897 - 'party_a' - make this channel the preferred Party A in any CDR between two
1898 channels. If two channels have this property set, the creation time of the
1899 channel is used to determine who is Party A. Note that dialed channels are
1900 never Party A in a CDR.
1901 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
1902 application when set to True, and analogous to the 'e' option in ResetCDR
1907 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
1908 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
1909 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
1912 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
1913 string, i.e., [[context],extension],priority. If set on a channel, if a
1914 channel leaves a bridge but is not hung up it will resume dialplan execution
1919 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
1920 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
1921 The value of this setting is ignored when disabled is used for the argument.
1925 * A new function provided by chan_pjsip, this function can be used in
1926 conjunction with the Dial application to construct a dial string that will
1927 dial all contacts on an Address of Record associated with a chan_pjsip
1932 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
1933 outbound channel prior to dialing.
1937 * Redirecting reasons can now be set to arbitrary strings. This means
1938 that the REDIRECTING dialplan function can be used to set the redirecting
1939 reason to any string. It also allows for custom strings to be read as the
1940 redirecting reason from SIP Diversion headers.
1944 * The SPEECH_ENGINE function now supports read operations. When read from, it
1945 will return the current value of the requested attribute.
1949 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1950 system as mailbox@context. The rest of the system cannot add @default
1951 to mailbox identifiers for app_voicemail that do not specify a context
1952 any longer. It is a mailbox identifier format that should only be
1953 interpreted by app_voicemail.
1959 res_agi (Asterisk Gateway Interface)
1961 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
1963 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
1966 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
1967 will start the playback of the audio at the position specified. It will
1968 also return the final position of the file in 'endpos'.
1970 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
1971 channel variable if the user stopped the file playback or if a remote
1972 entity stopped the playback. If neither stopped the playback, it will
1973 indicate the overall success/failure of the playback. If stopped early,
1974 the final offset of the file will be set in the CPLAYBACKOFFSET channel
1977 * The SAY ALPHA command now accepts an additional parameter to control
1978 whether it specifies the case of uppercase, lowercase, or all letters to
1979 provide functionality similar to SayAlphaCase.
1981 res_ari (Asterisk RESTful Interface) (and others)
1983 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
1984 control telephony primitives in Asterisk by remote client. This includes
1985 channels, bridges, endpoints, media, and other fundamental concepts. Users
1986 of ARI can develop their own communications applications, controlling
1987 multiple channels using an HTTP RESTful interface and receiving JSON events
1988 about the objects via a WebSocket connection. ARI can be configured in
1989 Asterisk via ari.conf. For more information on ARI, see
1990 https://wiki.asterisk.org/wiki/x/0YCLAQ
1994 * Parking has been extracted from the Asterisk core as a loadable module,
1995 res_parking. Configuration for parking is now provided by res_parking.conf.
1996 Configuration through features.conf is no longer supported.
1998 * res_parking uses the configuration framework. If an invalid configuration is
1999 supplied, res_parking will fail to load or fail to reload. Previously,
2000 invalid configurations would generally be accepted, with certain errors
2001 resulting in individually disabled parking lots.
2003 * Parked calls are now placed in bridges. While this is largely an
2004 architectural change, it does have implications on how channels in a parking
2005 lot are viewed. For example, commands that display channels in bridges will
2006 now also display the channels in a parking lot.
2008 * The order of arguments for the new parking applications have been modified.
2009 Timeout and return context/exten/priority are now implemented as options,
2010 while the name of the parking lot is now the first parameter. See the
2011 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
2012 in-depth information as well as syntax.
2014 * Extensions are by default no longer automatically created in the dialplan to
2015 park calls or pickup parked calls. Generation of dialplan extensions can be
2016 enabled using the 'parkext' configuration option.
2018 * ADSI functionality for parking is no longer supported. The 'adsipark'
2019 configuration option has been removed as a result.
2021 * The PARKINGSLOT channel variable has been deprecated in favor of
2022 PARKING_SPACE to match the naming scheme of the new system.
2024 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
2025 channel even when the configuration option 'comebactoorigin' is enabled.
2027 * A new CLI command 'parking show' has been added. This allows a user to
2028 inspect the parking lots that are currently in use.
2029 'parking show <parkinglot>' will also show the parked calls in a specific
2032 * The CLI command 'parkedcalls' is now deprecated in favor of
2033 'parking show <parkinglot>'.
2035 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
2036 can be used to get a list of parked calls for a specific parking lot.
2038 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
2039 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
2040 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
2041 longer a required argument.
2043 * The ParkAndAnnounce application is now provided through res_parking instead
2044 of through the separate app_parkandannounce module.
2046 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
2047 by default. Instead, it will follow the timeout rules of the parking lot. The
2048 old behavior can be reproduced by using the 'c' option.
2050 * Dynamic parking lots will now fail to be created under the following
2052 - if the parking lot specified by PARKINGDYNAMIC does not exist
2053 - if they require exclusive park and parkedcall extensions which overlap
2054 with existing parking lots.
2056 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
2057 currently contain no calls. Dynamic parking lots containing parked calls
2058 will persist through the reloads without alteration.
2060 * If 'parkext_exclusive' is set for a parking lot and that extension is
2061 already in use when that parking lot tries to register it, this is now
2062 considered a parking system configuration error. Configurations which do
2063 this will be rejected.
2065 * Added channel variable PARKER_FLAT. This contains the name of the extension
2066 that would be used if 'comebacktoorigin' is enabled. This can be useful when
2067 comebacktoorigin is disabled, but the dialplan or an external control
2068 mechanism wants to use the extension in the park-dial context that was
2069 generated to re-dial the parker on timeout.
2071 res_pjsip (and many others)
2073 * A large number of resource modules make up the SIP stack based on pjsip.
2074 The chan_pjsip channel driver users these resource modules to provide
2075 various SIP functionality in Asterisk. The majority of configuration for
2076 these modules is performed in pjsip.conf. Other modules may use their
2077 own configuration files.
2079 * Added 'set_var' option for an endpoint. For each variable specified that
2080 variable gets set upon creation of a channel involving the endpoint.
2084 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
2085 them, an Asterisk-specific version of PJSIP needs to be installed.
2086 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
2088 res_statsd/res_chan_stats
2090 * A new resource module, res_statsd, has been added, which acts as a statsd
2091 client. This module allows Asterisk to publish statistics to a statsd
2092 server. In conjunction with res_chan_stats, it will publish statistics about
2093 channels to the statsd server. It can be configured via res_statsd.conf.
2097 * Device state for XMPP buddies is now available using the following format:
2098 XMPP/<client name>/<buddy address>
2099 If any resource is available the device state is considered to be not in use.
2100 If no resources exist or all are unavailable the device state is considered
2107 Realtime/Database Scripts
2109 * Asterisk previously included example db schemas in the contrib/realtime/
2110 directory of the source tree. This has been replaced by a set of database
2111 migrations using the Alembic framework. This allows you to use alembic to
2112 initialize the database for you. It will also serve as a database migration
2113 tool when upgrading Asterisk in the future.
2115 See contrib/ast-db-manage/README.md for more details.
2119 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
2120 This python script will convert an existing sip.conf file to a
2121 pjsip.conf file, for use with the chan_pjsip channel driver. This script
2122 is meant to be an aid in converting an existing chan_sip configuration to
2123 a chan_pjsip configuration, but it is expected that configuration beyond
2124 what the script provides will be needed.
2126 ------------------------------------------------------------------------------
2127 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
2128 ------------------------------------------------------------------------------
2132 * The Asterisk build system will now build and install a shared library
2133 (libasteriskssl.so) used to wrap various initialization and shutdown functions
2134 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
2135 that Asterisk can ensure that these functions do *not* get called by any
2136 modules that are loaded into Asterisk, since they should only be called once
2137 in any single process. If desired, this feature can be disabled by supplying
2138 the "--disable-asteriskssl" option to the configure script.
2140 * A new make target, 'full', has been added to the Makefile. This performs
2141 the same compilation actions as make all, but will also scan the entirety of
2142 each source file for documentation. This option is needed to generate AMI
2143 event documentation. Note that your system must have Python in order for
2144 this make target to succeed.
2146 * The optimization portion of the build system has been reworked to avoid
2147 broken builds on certain architectures. All architecture-specific
2148 optimization has been removed in favor of using -march=native to allow gcc
2149 to detect the environment in which it is running when possible. This can
2150 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
2152 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
2153 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
2155 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
2156 previously parsed the header file to obtain the version of Asterisk, you
2157 will now have to go through Asterisk to get the version information.
2165 * Added 'F()' option. Similar to the dial option, this can be supplied with
2166 arguments indicating where the callee should go after the caller is hung up,
2167 or without options specified, the priority after the Queue will be used.
2172 * Added menu action admin_toggle_mute_participants. This will mute / unmute
2173 all non-admin participants on a conference. The confbridge configuration
2174 file also allows for the default sounds played to all conference users when
2175 this occurs to be overriden using sound_participants_unmuted and
2176 sound_participants_muted.
2178 * Added menu action participant_count. This will playback the number of
2179 current participants in a conference.
2181 * Added announcement configuration option to user profile. If set the sound
2182 file will be played to the user, and only the user, upon joining the
2185 * Added record_file_append option that defaults to "yes", but if set to no
2186 will create a new file between each start/stop recording.
2191 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
2192 channels respectively before the callee channels are called.
2197 * Added support for IPv6.
2199 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
2200 external process will cause the current playlist to be cleared, including
2201 stopping any audio file that is currently playing. This is useful when you
2202 want to interrupt audio playback only when specific DTMF is entered by the
2208 * A new option, 'I' has been added to app_followme. By setting this option,
2209 Asterisk will not update the caller with connected line changes when they
2210 occur. This is similar to app_dial and app_queue.
2212 * The 'N' option is now ignored if the call is already answered.
2214 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
2215 and caller channels respectively before the callee channels are called.
2217 * The winning FollowMe outgoing call is now put on hold if the caller put it on
2223 * MixMonitor hooks now have IDs associated with them which can be used to
2224 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
2225 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
2226 now accepts that ID as an argument.
2228 * Added 'm' option, which stores a copy of the recording as a voicemail in the
2229 indicated mailboxes.
2234 * The connect action in app_mysql now allows you to specify a port number to
2235 connect to. This is useful if you run a MySQL server on a non-standard
2241 * Increased the default number of allowed destinations from 5 to 12.
2246 * The app_page application now no longer depends on DAHDI or app_meetme. It
2247 has been re-architected to use app_confbridge internally.
2252 * Added queue options autopausebusy and autopauseunavail for automatically
2253 pausing a queue member when their device reports busy or congestion.
2255 * The 'ignorebusy' option for queue members has been deprecated in favor of
2256 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
2257 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
2258 per interface basis. Individual ringinuse values can now be set in
2259 queues.conf via an argument to member definitions. Lastly, the queue
2260 'ringinuse' setting now only determines defaults for the per member
2261 'ringinuse' setting and does not override per member settings like it does
2262 in earlier versions.
2264 * Added 'F()' option. Similar to the dial option, this can be supplied with
2265 arguments indicating where the callee should go after the caller is hung up,
2266 or without options specified, the priority after the Queue will be used.
2268 * Added new option log_member_name_as_agent, which will cause the membername to
2269 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
2270 state_interface has been set.
2272 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
2274 * App_queue will now play periodic announcements for the caller that
2275 holds the first position in the queue while waiting for answer.
2279 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
2280 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
2281 changed arguments to SayUnixTime so that every option is truly optional even
2282 when using multiple options (so that j option could be used without having to
2283 manually specify timezone and format) There are other benefits, e.g., format
2284 can now be used without specifying time zone as well.
2289 * Addition of the VM_INFO function - see Function changes.
2291 * The imapserver, imapport, and imapflags configuration options can now be
2292 overriden on a user by user basis.
2294 * When voicemail plays a message's envelope with saycid set to yes, when
2295 reaching the caller id field it will play a recording of a file with the same
2296 base name as the sender's callerid if there is a similarly named file in
2297 <astspooldir>/recordings/callerids/
2299 * Voicemails now contains a unique message identifier "msg_id", which is stored
2300 in the message envelope with the sound files. IMAP backends will now store
2301 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
2302 backends will store the message identifier in a "msg_id" column. See
2303 UPGRADE.txt for more information.
2305 * Added VoiceMailPlayMsg application. This application will play a single
2306 voicemail message from a mailbox. The result of the application, SUCCESS or
2307 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
2312 * Hangup handlers can be attached to channels using the CHANNEL() function.
2313 Hangup handlers will run when the channel is hung up similar to the h
2314 extension. The hangup_handler_push option will push a GoSub compatible
2315 location in the dialplan onto the channel's hangup handler stack. The
2316 hangup_handler_pop option will remove the last added location, and optionally
2317 replace it with a new GoSub compatible location. The hangup_handler_wipe
2318 option will remove all locations on the stack, and optionally add a new
2321 * The expression parser now recognizes the ABS() absolute value function,
2322 which will convert negative floating point values to positive values.
2324 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
2325 control of faxdetect.
2327 * Addition of the VM_INFO function that can be used to retrieve voicemail
2328 user information, such as the email address and full name.
2329 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
2332 * The REDIRECTING function now supports the redirecting original party id
2335 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
2336 lets you set some of the configuration options from the [general] section
2337 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
2338 the key sequence used to activate built-in features, such as blindxfer,
2339 and automon. See the built-in documentation for details.
2341 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
2342 instead of simply the uri. This is the format that MessageSend() can use
2343 in the from parameter for outgoing SIP messages.
2345 * Added the PRESENCE_STATE function. This allows retrieving presence state
2346 information from any presence state provider. It also allows setting
2347 presence state information from a CustomPresence presence state provider.
2348 See AMI/CLI changes for related commands.
2350 * Added the AMI_CLIENT function to make manager account attributes available
2351 to the dialplan. It currently supports returning the current number of
2352 active sessions for a given account.
2354 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
2355 and the REDIRECTING functions.
2363 * Added a manager event "LocalBridge" for local channel call bridges between
2364 the two pseudo-channels created.
2369 * Added dialtone_detect option for analog ports to disconnect incoming
2370 calls when dialtone is detected.
2372 * Added option colp_send to send ISDN connected line information. Allowed
2373 settings are block, to not send any connected line information; connect, to
2374 send connected line information on initial connect; and update, to send
2375 information on any update during a call. Default is update.
2377 * Add options namedcallgroup and namedpickupgroup to support installations
2378 where a higher number of groups (>64) is required.
2380 * Added support to use private party ID information with PRI calls.
2385 * A new channel driver named chan_motif has been added which provides support for
2386 Google Talk and Jingle in a single channel driver. This new channel driver includes
2387 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
2388 hold, unhold, and ringing notification. It is also compliant with the current Jingle
2389 specification, current Google Jingle specification, and the original Google Talk
2395 * Added NAT support for RTP. Setting in config is 'nat', which can be set
2396 globally and overriden on a peer by peer basis.
2398 * Direct media functionality has been added. Options in config are:
2399 directmedia (directrtp) and directrtpsetup (earlydirect)
2401 * ChannelUpdate events now contain a CallRef header.
2406 * Asterisk will no longer substitute CID number for CID name in the display
2407 name field if CID number exists without a CID name. This change improves
2408 compatibility with certain device features such as Avaya IP500's directory
2411 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
2412 created using that setting to not be removed during SIP reload.
2414 * Added settings recordonfeature and recordofffeature. When receiving an INFO
2415 request with a "Record:" header, this will turn the requested feature on/off.
2416 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
2417 dynamic features must be enabled and configured properly on the requesting
2418 channel for this to function properly.
2420 * Add support to realtime for the 'callbackextension' option.
2422 * When multiple peers exist with the same address, but differing
2423 callbackextension options, incoming requests that are matched by address
2424 will be matched to the peer with the matching callbackextension if it is
2427 * Two new NAT options, auto_force_rport and auto_comedia, have been added
2428 which set the force_rport and comedia options automatically if Asterisk
2429 detects that an incoming SIP request crossed a NAT after being sent by
2430 the remote endpoint.
2432 * The default global nat setting in sip.conf has been changed from force_rport
2433 to auto_force_rport.
2435 * NAT settings are now a combinable list of options. The equivalent of the
2436 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
2438 * Adds an option send_diversion which can be disabled to prevent
2439 diversion headers from automatically being added to INVITE requests.
2441 * Add support for lightweight NAT keepalive. If enabled a blank packet will
2442 be sent to the remote host at a given interval to keep the NAT mapping open.
2443 This can be enabled using the keepalive configuration option.
2445 * Add option 'tonezone' to specify country code for indications. This option
2446 can be set both globally and overridden for specific peers.
2448 * The SIP Security Events Framework now supports IPv6.
2450 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
2451 between multiple user agents. When set, for directmedia reinvites,
2452 Asterisk will not send an immediate reinvite on an incoming call leg. This
2453 option is useful when peered with another SIP user agent that is known to
2454 send immediate direct media reinvites upon call establishment.
2456 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
2459 * Add options subminexpiry and submaxexpiry to set limits of subscription
2460 timer independently from registration timer settings. The setting of the
2461 registration timer limits still is done by options minexpiry, maxexpiry
2462 and defaultexpiry. For backwards compatibility the setting of minexpiry
2463 and maxexpiry also is used to configure the subscription timer limits if
2464 subminexpiry and submaxexpiry are not set in sip.conf.
2466 * Set registration timer limits to default values when reloading sip
2467 configuration and values are not set by configuration.
2469 * Add options namedcallgroup and namedpickupgroup to support installations
2470 where a higher number of groups (>64) is required.
2472 * When a MESSAGE request is received, the address the request was received from
2473 is now saved in the SIP_RECVADDR variable.
2475 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
2476 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
2477 the ANI2/OLI information is set on the channel, which can be retrieved using
2478 the CALLERID function.
2480 * Peers can now be configured to support negotiation of ICE candidates using
2481 the setting icesupport. See res_rtp_asterisk changes for more information.
2483 * Added support for format attribute negotiation. See the Codecs changes for
2486 * Extra headers specified with SIPAddHeader are sent with the REFER message
2487 when using Transfer application. See refer_addheaders in sip.conf.sample.
2489 * Added support to use private party ID information with calls.
2491 * Adds an option discard_remote_hold_retrieval that when set stops telling
2492 the peer to start music on hold.
2497 * Added skinny version 17 protocol support.
2501 --------------------
2502 * Added option 'dtmf_duration' allowing playback time of DTMF tones to be set
2504 * Modified option 'date_format' to allow options to display date in 31Jan and Jan31
2505 formats as options 0 and 1. The previous options 0 and 1 now map to options 2 and 3
2506 as per the UNISTIM protocol.
2508 * Fixed issues with dialtone not matching indications.conf and mute stopping rx
2509 as well as tx. Also fixed issue with call "Timer" displaying as French "Dur\E9e"
2511 * Added ability to use multiple lines for a single phone. This allows multiple
2512 calls to occur on a single phone, using callwaiting and switching between calls.
2514 * Added option 'sharpdial' allowing end dialing by pressing # key
2516 * Added option 'interdigit_timer' to control phone dial timeout
2518 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
2520 * Added global 'debug' option, that enables debug in channel driver
2522 * Added ability to translate on-screen menu in multiple languages. Tested on
2523 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
2524 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
2527 * In addition to English added French and Russian languages for on-screen menus
2529 * Reworked dialing number input: added dialing by timeout, immediate dial on
2530 on dialplan compare, phone number length now not limited by screen size
2532 * Added ability to pickup a call using features.conf defined value and
2538 * Add options namedcallgroup and namedpickupgroup to support installations
2539 where a higher number of groups (>64) is required.
2541 * Added support to use private party ID information with calls.
2546 * The minimum DTMF duration can now be configured in asterisk.conf
2547 as "mindtmfduration". The default value is (as before) set to 80 ms.
2548 (previously it was only available in source code)
2550 * Named ACLs can now be specified in acl.conf and used in configurations that
2551 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
2552 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
2553 working ACL. In addition, some CLI commands have been added to provide
2554 show information and allow for module reloading - see CLI Changes.
2556 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
2557 items (separated by commas), and items in the rule can be negated by prefixing
2558 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
2559 longer necessray to control the order that the 'permit' and 'deny' columns are
2560 returned from queries.
2562 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
2563 be used within the dynamic weight attribute when specifying a mapping.
2565 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
2566 header, instead of putting the user defined event name there. When enabled
2567 the UserDefType header is added for user defined events. This feature is
2568 enabled with the setting show_user_defined.
2570 * Macro has been deprecated in favor of GoSub. For redirecting and connected
2571 line purposes use the following variables instead of their macro equivalents:
2572 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
2573 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
2574 cc_callback_macro in channel configurations.
2576 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
2579 * Call files now support the "early_media" option to connect with an outgoing
2580 extension when early media is received.
2582 * Added support to use private party ID information with calls.
2587 * A new channel variable, AGIEXITONHANGUP, has been added which allows
2588 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
2589 AGI application would exit immediately after a channel hangup is detected.
2591 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
2592 are resolved and each address is attempted in turn until one succeeds or
2596 AMI (Asterisk Manager Interface)
2598 * The originate action now has an option "EarlyMedia" that enables the
2599 call to bridge when we get early media in the call. Previously,
2600 early media was disregarded always when originating calls using AMI.
2602 * Added setvar= option to manager accounts (much like sip.conf)
2604 * Originate now generates an error response if the extension given is not found
2607 * MixMonitor will now show IDs associated with the mixmonitor upon creating
2608 them if the i(variable) option is used. StopMixMonitor will accept
2609 MixMonitorID as an option to close specific MixMonitors.
2611 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
2612 updated to include information about peers configured with
2613 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
2614 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
2615 returned if auto_force_rport is not enabled.
2617 * Added SIPpeerstatus manager command which will generate PeerStatus events
2618 similar to the existing PeerStatus events found in chan_sip on demand.
2620 * Hangup now can take a regular expression as the Channel option. If you want
2621 to hangup multiple channels, use /regex/ as the Channel option. Existing
2622 behavior to hanging up a single channel is unchanged, but if you pass a regex,
2623 the manager will send you a list of channels back that were hung up.
2625 * Support for IPv6 addresses has been added.
2627 * AMI Events can now be documented in the Asterisk source. Note that AMI event
2628 documentation is only generated when Asterisk is compiled using 'make full'.
2629 See the CLI section for commands to display AMI event information.
2631 * The AMI Hangup event now includes the AccountCode header so you can easily
2632 correlate with AMI Newchannel events.
2634 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
2635 the StateInterface of the queue member.
2637 * Added AMI event SessionTimeout in the Call category that is issued when a
2638 call is terminated due to either RTP stream inactivity or SIP session timer
2641 * CEL events can now contain a user defined header UserDefType. See core
2642 changes for more information.
2644 * OOH323 ChannelUpdate events now contain a CallRef header.
2646 * Added PresenceState command. This command will report the presence state for
2647 the given presence provider.
2649 * Added Parkinglots command. This will list all parking lots as a series of
2650 AMI Parkinglot events.
2652 * Added MessageSend command. This behaves in the same manner as the
2653 MessageSend application, and is a technolgoy agnostic mechanism to send out
2654 of call text messages.
2656 * Added "message" class authorization. This grants an account permission to
2657 send out of call messages. Write-only.
2662 * The "dialplan add include" command has been modified to create context a context
2663 if one does not already exist. For instance, "dialplan add include foo into bar"
2664 will create context "bar" if it does not already exist.
2666 * A "dialplan remove context" command has been added to remove a context from
2669 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
2670 filenames of all running mixmonitors on a channel.
2672 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
2673 numeric instead of 0, 1, or 2.
2675 * "stun show status" will show a table describing how the STUN client is
2678 * "acl show [named acl]" will show information regarding a Named ACL. The
2679 acl module can be reloaded with "reload acl".
2681 * Added CLI command to display AMI event information - "manager show events",
2682 which shows a list of all known and documented AMI events, and "manager show
2683 event [event name]", which shows detail information about a specific AMI
2686 * The result of the CLI command "queue show" now includes the state interface
2687 information of the queue member.
2689 * The command "core set verbose" will now set a separate level of logging for
2690 each remote console without affecting any other console.
2692 * Added command "cdr show pgsql status" to check connection status
2694 * "sip show channel" will now display the complete route set.
2696 * Added "presencestate list" command. This command will list all custom
2697 presence states that have been set by using the PRESENCE_STATE dialplan
2700 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
2701 command. This changes a custom presence to a new state.
2706 * Codec lists may now be modified by the '!' character, to allow succinct
2707 specification of a list of codecs allowed and disallowed, without the
2708 requirement to use two different keywords. For example, to specify all
2709 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
2711 * Add support for parsing SDP attributes, generating SDP attributes, and
2712 passing it through. This support includes codecs such as H.263, H.264, SILK,
2713 and CELT. You are able to set up a call and have attribute information pass.
2714 This should help considerably with video calls.
2716 * The iLBC codec can now use a system-provided iLBC library if one is installed,
2717 just like the GSM codec.
2721 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
2722 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
2726 * Asterisk version and build information is now logged at the beginning of a
2729 * Threads belonging to a particular call are now linked with callids which get
2730 added to any log messages produced by those threads. Log messages can now be
2731 easily identified as involved with a certain call by looking at their call id.
2732 Call ids may also be attached to log messages for just about any case where
2733 it can be determined to be related to a particular call.
2735 * Each logging destination and console now have an independent notion of the
2736 current verbosity level. Logger.conf now allows an optional argument to
2737 the 'verbose' specifier, indicating the level of verbosity sent to that
2738 particular logging destination. Additionally, remote consoles now each
2739 have their own verbosity level. The command 'core set verbose' will now set
2740 a separate level for each remote console without affecting any other
2746 * Added 'announcement' option which will play at the start of MOH and between
2747 songs in modes of MOH that can detect transitions between songs (eg.
2753 * New per parking lot options: comebackcontext and comebackdialtime. See
2754 configs/features.conf.sample for more details.
2756 * Channel variable PARKER is now set when comebacktoorigin is disabled in
2759 * Channel variable PARKEDCALL is now set with the name of the parking lot
2760 when a timeout occurs.
2766 CDR Postgresql Driver
2768 * Added command "cdr show pgsql status" to check connection status
2771 CDR Adaptive ODBC Driver
2773 * Added schema option for databases that support specifying a schema.
2781 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
2782 CALENDAR_WRITE has completed successfully.
2787 * A new option, 'probation' has been added to rtp.conf
2788 RTP in strictrtp mode can now require more than 1 packet to exit learning
2789 mode with a new source (and by default requires 4). The probation option
2790 allows the user to change the required number of packets in sequence to any
2791 desired value. Use a value of 1 to essentially restore the old behavior.
2792 Also, with strictrtp on, Asterisk will now drop all packets until learning
2793 mode has successfully exited. These changes are based on how pjmedia handles
2794 media sources and source changes.
2796 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
2797 enabled or disabled using the icesupport setting. A variety of other
2798 settings have been introduced to configure STUN/TURN connections.
2803 * A new module, res_corosync, has been introduced. This module uses the
2804 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
2805 of Asterisk servers to both Message Waiting Indication (MWI) and/or
2806 Device State (presence) information. This module is very similar to, and
2807 is a replacement for the res_ais module that was in previous releases of
2813 * This module adds a cleaned up, drop-in replacement for res_jabber called
2814 res_xmpp. This provides the same externally facing functionality but is
2815 implemented differently internally. res_jabber has been deprecated in favor
2816 of res_xmpp; please see the UPGRADE.txt file for more information.
2821 * The safe_asterisk script has been updated to allow several of its parameters
2822 to be set from environment variables. This also enables a custom run
2823 directory of Asterisk to be specified, instead of defaulting to /tmp.
2825 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
2826 its value to determine the directory to assume is the top-level directory of
2827 the source tree. If the variable is not set, it defaults to the current
2828 behavior and uses the current working directory.
2830 ------------------------------------------------------------------------------
2831 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
2832 ------------------------------------------------------------------------------
2836 * Asterisk now has protocol independent support for processing text messages
2837 outside of a call. Messages are routed through the Asterisk dialplan.
2838 SIP MESSAGE and XMPP are currently supported. There are options in
2839 jabber.conf and sip.conf to allow enabling these features.
2840 -> jabber.conf: see the "sendtodialplan" and "context" options.
2841 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
2842 and "outofcall_message_context" options.
2843 The MESSAGE() dialplan function and MessageSend() application have been
2844 added to go along with this functionality. More detailed usage information
2845 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
2846 * If real-time text support (T.140) is negotiated, it will be preferred for
2847 sending text via the SendText application. For example, via SIP, messages
2848 that were once sent via the SIP MESSAGE request would be sent via RTP if
2849 T.140 text is negotiated for a call.
2853 * parkedmusicclass can now be set for non-default parking lots.
2855 Asterisk Manager Interface
2856 --------------------------
2857 * PeerStatus now includes Address and Port.
2858 * Added Hold events for when the remote party puts the call on and off hold
2859 for chan_dahdi ISDN channels.
2860 * Added new action MeetmeListRooms to list active conferences (shows same
2861 data as "meetme list" at the CLI).
2862 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
2863 Description field that is set by 'description' in the channel configuration
2865 * Added Uniqueid header to UserEvent.
2866 * Added new action FilterAdd to control event filters for the current session.
2867 This requires the system permission and uses the same filter syntax as
2868 filters that can be defined in manager.conf
2869 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
2870 versions had some instances of the event converted, but others were left
2871 as-is. All Unlink events should now be converted to Bridge events. The AMI
2872 protocol version number was incremented to 1.2 as a result of this change.
2874 Asterisk HTTP Server
2875 --------------------------
2876 * The HTTP Server can bind to IPv6 addresses.
2879 --------------------------
2880 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
2881 with busydetect. usage example: busypattern=200,200,200,600
2884 --------------------------
2885 * New 'gtalk show settings' command showing the current settings loaded from
2887 * The 'logger reload' command now supports an optional argument, specifying an
2888 alternate configuration file to use.
2889 * 'dialplan add extension' command will now automatically create a context if
2890 the specified context does not exist with a message indicated it did so.
2891 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
2892 Description field which can be populated with 'description' in the channel
2893 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
2896 --------------------------
2897 * The filter option in cdr_adaptive_odbc now supports negating the argument,
2898 thus allowing records which do NOT match the specified filter.
2899 * Added ability to log CONGESTION calls to CDR
2902 --------------------------
2903 * Ability to define custom SILK formats in codecs.conf.
2904 * Addition of speex32 audio format with translation.
2905 * CELT codec pass-through support and ability to define
2906 custom CELT formats in codecs.conf.
2907 * Ability to read raw signed linear files with sample rates
2908 ranging from 8khz - 192khz. The new file extensions introduced
2909 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
2910 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
2911 Skinny, H.323, etc) can still only support the following codecs:
2912 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
2913 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
2914 Video: h261, h263, h263p, h264, mpeg4
2919 --------------------------
2920 * New highly optimized and customizable ConfBridge application capable of
2921 mixing audio at sample rates ranging from 8khz-96khz.
2922 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
2923 and bridge profiles on a channel.
2924 * CONFBRIDGE_INFO dialplan function capable of retrieving information
2925 about a conference such as locked status and number of parties, admins,
2927 * Addition of video_mode option in confbridge.conf for adding video support
2928 into a bridge profile.
2929 * Addition of the follow_talker video_mode in confbridge.conf. This video
2930 mode dynamically switches the video feed to always display the loudest talker
2931 supplying video in the conference.
2935 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
2936 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
2937 variables from asterisk.conf.
2941 * Addition of the JITTERBUFFER dialplan function. This function allows
2942 for jitterbuffering to occur on the read side of a channel. By using
2943 this function conference applications such as ConfBridge and MeetMe can
2944 have the rx streams jitterbuffered before conference mixing occurs.
2945 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
2947 * Added STRREPLACE function. This function let's the user search a variable
2948 for a given string to replace with another string as many times as the
2949 user specifies or just throughout the whole string.
2950 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
2951 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
2952 * Added extensions to chan_ooh323 in function CHANNEL()
2954 libpri channel driver (chan_dahdi) DAHDI changes
2955 --------------------------
2956 * Added moh_signaling option to specify what to do when the channel's bridged
2957 peer puts the ISDN channel on hold.
2958 * Added display_send and display_receive options to control how the display ie
2959 is handled. To send display text from the dialplan use the SendText()
2960 application when the option is enabled.
2961 * Added mcid_send option to allow sending a MCID request on a span.
2964 --------------------------
2965 * Added setvar option to calendar.conf to allow setting channel variables on
2966 notification channels.
2967 * Added "calendar show types" CLI command to list registered calendar
2971 --------------------------
2972 * Added two new options, r and t with file name arguments to record
2973 single direction (unmixed) audio recording separate from the bidirectional
2974 (mixed) recording. The mixed file name argument is optional now as long
2975 as at least one recording option is used.
2978 --------------------------
2979 * Added a new option, l, which will disable local call optimization for
2980 channels involved with the FollowMe thread. Use this option to improve
2981 compatability for a FollowMe call with certain dialplan apps, options, and
2985 --------------------------
2986 * Added option "k" that will automatically close the conference when there's
2987 only one person left when a user exits the conference.
2990 --------------------------
2991 * cel_pgsql now supports the 'extra' column for data added using the
2992 CELGenUserEvent() application.
2995 --------------------------
2996 * Support for defining hints has been added to pbx_lua. See the 'hints' table
2997 in the sample extensions.lua file for syntax details.
2998 * Applications that perform jumps in the dialplan such as Goto will now
2999 execute properly. When pbx_lua detects that the context, extension, or
3000 priority we are executing on has changed it will immediately return control
3001 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
3002 the priority after the currently executing priority.
3003 * An autoservice is now started by default for pbx_lua channels. It can be
3004 stopped and restarted using the autoservice_stop() and autoservice_start()
3008 --------------------------
3009 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
3010 into a FAXStatus event with an 'Operation' header that will be either
3011 'send', 'receive', and 'gateway'.
3012 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
3013 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
3014 feature will handle converting a fax call between an audio T.30 fax terminal
3015 and an IFP T.38 fax terminal.
3019 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
3020 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
3021 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
3025 * Added general option negative_penalty_invalid default off. when set
3026 members are seen as invalid/logged out when there penalty is negative.
3027 for realtime members when set remove from queue will set penalty to -1.
3028 * Added queue option autopausedelay when autopause is enabled it will be
3029 delayed for this number of seconds since last successful call if there
3030 was no prior call the agent will be autopaused immediately.
3031 * Added member option ignorebusy this when set and ringinuse is not
3032 will allow per member control of multiple calls as ringinuse does for
3037 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
3039 * Added 'k' option to MeetMe to automatically kill the conference when there's only
3040 one participant left (much like a normal call bridge)
3041 * Added extra argument to Originate to set timeout.
3045 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
3046 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
3047 utility in the UTILS section of menuselect. If an existing astdb is found and no
3048 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
3049 convert an existing astdb to the SQLite3 version automatically at runtime.
3053 * Modules marked as deprecated are no longer marked as building by default. Enabling
3054 these modules is still available via menuselect.
3058 * authdebug is now disabled by default. To enable this functionaility again
3059 set authdebug = yes in iax.conf.
3063 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
3064 releases it was disabled.
3068 * The PBX core previously made a call with a non-existing extension test for
3069 extension s@default and jump there if the extension existed.
3070 This was a bad default behaviour and violated the principle of least surprise.
3071 It has therefore been changed in this release. It may affect some
3072 applications and configurations that rely on this behaviour. Most channel
3073 drivers have avoided this for many releases by testing whether the extension
3074 called exists before starting the PBX and generating a local error.
3075 This behaviour still exists and works as before.
3077 Extension "s" is used when no extension is given in a channel driver,
3078 like immediate answer in DAHDI or calling to a domain with no user part
3081 ------------------------------------------------------------------------------
3082 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
3083 ------------------------------------------------------------------------------
3087 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
3088 now defaults to force_rport. It is very important that phones requiring nat=no be
3089 specifically set as such instead of relying on the default setting. If at all
3090 possible, all devices should have nat settings configured in the general section as
3091 opposed to configuring nat per-device.
3092 * Added preferred_codec_only option in sip.conf. This feature limits the joint
3093 codecs sent in response to an INVITE to the single most preferred codec.
3094 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
3095 to be used for the outgoing call. It must be one of the codecs configured
3097 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
3098 to be used for holding a private key. If tlsprivatekey is not specified,
3099 tlscertfile is searched for both public and private key.
3100 * Added tlsclientmethod option to sip.conf. This allows the protocol for
3101 outbound client connections to be specified.
3102 * The sendrpid parameter has been expanded to include the options
3103 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
3104 header to be sent (equivalent to setting sendrpid=yes) and setting
3105 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
3106 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
3107 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
3108 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
3109 will accept the SDP even if the SDP version number is not properly incremented,
3110 but will generate a warning in the log indicating that the SIP peer that sent
3111 the SDP should have the 'ignoresdpversion' option set.
3112 * The 'nat' option has now been been changed to have yes, no, force_rport, and
3113 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
3114 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
3115 remote side requests it and disables symmetric RTP support. Setting it to
3116 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
3117 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
3118 and enables symmetric RTP support.
3119 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
3120 response. This permits the master channel to know how each channel dialled
3121 in a multi-channel setup resolved in an individual way. This carries a
3122 performance penalty and can be disabled in sip.conf using the
3123 'storesipcause' option.
3124 * Added 'externtcpport' and 'externtlsport' options to allow custom port
3125 configuration for the externip and externhost options when tcp or tls is used.
3126 * Added support for message body (stored in content variable) to SIP NOTIFY message
3127 accessible via AMI and CLI.
3128 * Added 'media_address' configuration option which can be used to explicitly specify
3129 the IP address to use in the SDP for media (audio, video, and text) streams.
3130 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
3131 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
3133 * Added 'use_q850_reason' configuration option for generating and parsing
3134 if available Reason: Q.850;cause=<cause code> header. It is implemented
3135 in some gateways for better passing PRI/SS7 cause codes via SIP.
3136 * When dialing SIP peers, a new component may be added to the end of the dialstring
3137 to indicate that a specific remote IP address or host should be used when dialing
3138 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
3139 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
3140 ability to selectively force bridged channels to also be encrypted is also
3141 implemented. Branching in the dialplan can be done based on whether or not
3142 a channel has secure media and/or signaling.
3143 * Added directmediapermit/directmediadeny to limit which peers can send direct media
3145 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
3146 Charge messages to snom phones.
3147 * Added support for G.719 media streams.
3148 * Added support for 16khz signed linear media streams.
3149 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
3150 RTP has been outfitted with the same abilities.
3151 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
3152 available in device configurations as well as in the dial plan.
3153 * Addition of the 'subscribe_network_change' option for turning on and off
3154 res_stun_monitor module support in chan_sip.
3155 * Addition of the 'auth_options_requests' option for turning on and off
3156 authentication for OPTIONS requests in chan_sip.
3160 * Add #tryinclude statement for config files. This provides the same
3161 functionality as the #include statement however an asterisk module will
3162 still load if the filename does not exist. Using the #include statement
3163 Asterisk will not allow the module to load.
3167 * Added rtsavesysname option into iax.conf to allow the systname to be saved
3168 on realtime updates.
3169 * Added the ability for chan_iax2 to inform the dialplan whether or not
3170 encryption is being used. This interoperates with the SIP SRTP implementation
3171 so that a secure SIP call can be bridged to a secure IAX call when the
3172 dialplan requires bridged channels to be "secure".
3173 * Addition of the 'subscribe_network_change' option for turning on and off
3174 res_stun_monitor module support in chan_iax.
3179 * Added ability to preset channel variables on indicated lines with the setvar
3180 configuration option. Also, clearvars=all resets the list of variables back
3182 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
3183 See configs/res_pktccops.conf for more information.
3185 XMPP Google Talk/Jingle changes
3186 -------------------------------
3187 * Added the externip option to gtalk.conf.
3188 * Added the stunaddr option to gtalk.conf which allows for the automatic
3189 retrieval of the external ip from a stun server.
3193 * Added 'p' option to PickupChan() to allow for picking up channel by the first
3194 match to a partial channel name.
3195 * Added .m3u support for Mp3Player application.
3196 * Added progress option to the app_dial D() option. When progress DTMF is
3197 present, those values are sent immediately upon receiving a PROGRESS message
3198 regardless if the call has been answered or not.
3199 * Added functionality to the app_dial F() option to continue with execution
3200 at the current location when no parameters are provided.
3201 * Added the 'a' option to app_dial to answer the calling channel before any
3202 announcements or macros are executed.
3203 * Modified app_dial to set answertime when the called channel answers even if
3204 the called channel hangs up during playback of an announcement.
3205 * Modified app_dial 'r' option to support an additional parameter to play an
3206 indication tone from indications.conf
3207 * Added c() option to app_chanspy. This option allows custom DTMF to be set
3208 to cycle through the next available channel. By default this is still '*'.
3209 * Added x() option to app_chanspy. This option allows DTMF to be set to
3210 exit the application.
3211 * The Voicemail application has been improved to automatically ignore messages
3212 that only contain silence.
3213 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
3214 associated mailbox(es) to be greetings-only.
3215 * The ChanSpy application now has the 'S' option, which makes the application
3216 automatically exit once it hits a point where no more channels are available
3218 * The ChanSpy application also now has the 'E' option, which spies on a single
3219 channel and exits when that channel hangs up.
3220 * The MeetMe application now turns on the DENOISE() function by default, for
3221 each participant. In our tests, this has significantly decreased background
3222 noise (especially noisy data centers).
3223 * Voicemail now permits storage of secrets in a separate file, located in the
3224 spool directory of each individual user. The control for this is located in
3225 the "passwordlocation" option in voicemail.conf. Please see the sample
3226 configuration for more information.
3227 * The ChanIsAvail application now exposes the returned cause code using a separate
3228 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
3229 * Added 'd' option to app_followme. This option disables the "Please hold"
3231 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
3232 received will terminate recording.
3233 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
3234 Previously the folder could only be set per context, but has now been extended
3235 using the imapfolder option.
3236 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
3237 * Voicemail now allows the pager date format to be specified separately from the
3239 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
3240 to allow joining, leaving, and sending text to group chats.
3241 * MeetMe has a new option 'G' to play an announcement before joining a conference.
3242 * Page has a new option 'A(x)' which will playback an announcement simultaneously
3243 to all paged phones (and optionally excluding the caller's one using the new
3244 option 'n') before the call is bridged.
3245 * The 'f' option to Dial has been augmented to take an optional argument. If no
3246 argument is provided, the 'f' option works as it always has. If an argument is
3247 provided, then the connected party information of all outgoing channels created
3248 during the Dial will be set to the argument passed to the 'f' option.
3249 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
3251 * The OSP lookup application adds in/outbound network ID, optional security,
3252 number portability, QoS reporting, destination IP port, custom info and service
3254 * Added new application VMSayName that will play the recorded name of the voicemail
3255 user if it exists, otherwise will play the mailbox number.
3256 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
3257 retrieve state for a particular bridge, where <name> is the conference name
3258 * app_directory now allows exiting at any time using the operator or pound key.
3259 * Voicemail now supports setting a locale per-mailbox.
3260 * Two new applications are provided for declining counting phrases in multiple
3261 languages. See the application notes for SayCountedNoun and SayCountedAdj for
3263 * Voicemail now runs the externnotify script when pollmailboxes is activated and
3265 * Voicemail now includes rdnis within msgXXXX.txt file.
3266 * ExternalIVR now supports IPv6 addresses.
3267 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
3268 at https://wiki.asterisk.org/wiki/x/oQBB
3269 * ParkedCall and Park can now specify the parking lot to use.
3273 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
3274 over SRV records associated with a specific service. From the CLI, type
3275 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
3276 details on how these may be used.
3277 * PITCH_SHIFT dialplan function added. This function can be used to modify the
3278 pitch of a channel's tx and rx audio streams.
3279 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
3280 setting various connected line and redirecting party information.
3281 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
3282 support ISDN subaddressing.
3283 * The CHANNEL() function now supports the "name" and "checkhangup" options.
3284 * For DAHDI channels, the CHANNEL() dialplan function now allows
3285 the dialplan to request changes in the configuration of the active
3286 echo canceller on the channel (if any), for the current call only.
3289 exten => s,n,Set(CHANNEL(echocan_mode)=off)
3291 The possible values are:
3293 on - normal mode (the echo canceller is actually reinitialized)
3295 fax - FAX/data mode (NLP disabled if possible, otherwise completely
3297 voice - voice mode (returns from FAX mode, reverting the changes that
3298 were made when FAX mode was requested)
3299 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
3300 and setting variables on the channel which created the current channel.
3301 Administrators should take care to avoid naming conflicts, when multiple
3302 channels are dialled at once, especially when used with the Local channel
3303 construct (which all could set variables on the master channel). Usage
3304 of the HASH() dialplan function, with the key set to the name of the slave
3305 channel, is one approach that will avoid conflicts.
3306 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
3308 * func_odbc now allows multiple row results to be retrieved without using
3309 mode=multirow. If rowlimit is set, then additional rows may be retrieved
3310 from the same query by using the name of the function which retrieved the
3311 first row as an argument to ODBC_FETCH().
3312 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
3313 dialplan. This function returns the content of the received message.
3314 * Added REPLACE, which searches a given variable name for a set of characters,
3315 then either replaces them with a single character or deletes them.
3316 * Added PASSTHRU, which literally passes the same argument back as its return
3317 value. The intent is to be able to use a literal string argument to
3318 functions that currently require a variable name as an argument.
3319 * HASH-associated variables now can be inherited across channel creation, by
3320 prefixing the name of the hash at assignment with the appropriate number of
3321 underscores, just like variables.
3322 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
3323 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
3324 whether or not channels that are bridged to the current channel will be
3325 required to have secure signaling and/or media.
3326 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
3327 the current channel has secure signaling and/or media.
3328 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
3329 "no_media_path" option.
3330 Returns "0" if there is a B channel associated with the call.
3331 Returns "1" if no B channel is associated with the call. The call is either
3332 on hold or is a call waiting call.
3333 * Added option to dialplan function CDR(), the 'f' option
3334 allows for high resolution times for billsec and duration fields.
3335 * FILE() now supports line-mode and writing.
3336 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
3337 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
3341 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
3342 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
3343 and is set when a dynamic feature is triggered.
3344 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
3345 to dynamically create a new parking lot matching the value this varible is
3347 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
3348 features.conf that should be the base for dynamic parkinglots.
3349 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
3350 parkinglot should have.
3351 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
3352 parkinglot should have.
3353 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
3358 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
3359 timeout has expired.
3360 * Added 'R' option to app_queue. This option stops moh and indicates ringing
3361 to the caller when an Agent's phone is ringing. This can be used to indicate
3362 to the caller that their call is about to be picked up, which is nice when
3363 one has been on hold for an extened period of time.
3364 * A new config option, penaltymemberslimit, has been added to queues.conf.
3365 When set this option will disregard penalty settings when a queue has too
3367 * A new option, 'I' has been added to both app_queue and app_dial.
3368 By setting this option, Asterisk will not update the caller with
3369 connected line changes or redirecting party changes when they occur.
3370 * A 'relative-periodic-announce' option has been added to queues.conf. When
3371 enabled, this option will cause periodic announce times to be calculated
3372 from the end of announcements rather than from the beginning.
3373 * The autopause option in queues.conf can be passed a new value, "all." The
3374 result is that if a member becomes auto-paused, he will be paused in all
3375 queues for which he is a member, not just the queue that failed to reach
3377 * Added dialplan function QUEUE_EXISTS to check if a queue exists
3378 * The queue logger now allows events to optionally propagate to a file,
3379 even when realtime logging is turned on. Additionally, realtime logging
3380 supports sending the event arguments to 5 individual fields, although it
3381 will fallback to the previous data definition, if the new table layout is
3384 mISDN channel driver (chan_misdn) changes
3385 ----------------------------------------
3386 * Added display_connected parameter to misdn.conf to put a display string
3387 in the CONNECT message containing the connected name and/or number if
3388 the presentation setting permits it.
3389 * Added display_setup parameter to misdn.conf to put a display string
3390 in the SETUP message containing the caller name and/or number if the
3391 presentation setting permits it.
3392 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
3393 indicate the dialplan settings are to be obtained from the asterisk
3395 * Made misdn.conf parameter callerid accept the "name" <number> format
3396 used by the rest of the system.
3397 * Made use the nationalprefix and internationalprefix misdn.conf
3398 parameters to prefix any received number from the ISDN link if that
3399 number has the corresponding Type-Of-Number. NOTE: This includes
3400 comparing the incoming call's dialed number against the MSN list.
3401 * Added the following new parameters: unknownprefix, netspecificprefix,
3402 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
3403 received number from the ISDN link if that number has the corresponding
3405 * Added new dialplan application misdn_command which permits controlling
3406 the CCBS/CCNR functionality.
3407 * Added new dialplan function mISDN_CC which permits retrieval of various
3408 values from an active call completion record.
3409 * For PTP, you should manually send the COLR of the redirected-to party
3410 for an incomming redirected call if the incoming call could experience
3411 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
3412 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
3413 if the REDIRECTING(from-num) is not empty.
3414 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
3415 option on all of the REDIRECTING statements before dialing the
3416 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
3417 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
3418 redirecting-to presentation (COLR) when it becomes available.
3419 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
3422 thirdparty mISDN enhancements
3423 -----------------------------
3424 mISDN has been modified by Digium, Inc. to greatly expand facility message
3426 * Enhanced COLP support for call diversion and transfer.
3427 * CCBS/CCNR support.
3429 The latest modified mISDN v1.1.x based version is available at:
3430 http://svn.digium.com/svn/thirdparty/mISDN/trunk
3431 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
3433 Tagged versions of the modified mISDN code are available under:
3434 http://svn.digium.com/svn/thirdparty/mISDN/tags
3435 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
3437 libpri channel driver (chan_dahdi) DAHDI changes
3438 -------------------------------------------
3439 * The channel variable PRIREDIRECTREASON is now just a status variable
3440 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
3441 to read and alter the reason.
3442 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
3443 redirected-to party for an incomming redirected call if the incoming call
3444 could experience further redirects. Just set the
3445 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
3446 to the COLR. A call has been redirected if the REDIRECTING(count) is not
3448 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
3449 use the inhibit(i) option on all of the REDIRECTING statements before
3450 dialing the redirected-to party. You still have to set the
3451 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
3452 will update the redirecting-to presentation (COLR) when it becomes available.
3453 * Added the ability to ignore calls that are not in a Multiple Subscriber
3454 Number (MSN) list for PTMP CPE interfaces.
3455 * Added dynamic range compression support for dahdi channels. It is
3456 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
3457 * Added support for ISDN calling and called subaddress with partial support
3458 for connected line subaddress.
3459 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
3460 * Added handling of received HOLD/RETRIEVE messages and the optional ability
3461 to transfer a held call on disconnect similar to an analog phone.
3462 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
3463 Will reroute/deflect an outgoing call when receive the message.
3464 Can use the DAHDISendCallreroutingFacility to send the message for the
3466 * Added standard location to add options to chan_dahdi dialing:
3467 Dial(DAHDI/g1[/extension[/options]])
3470 R Reverse charging indication
3471 * Added Reverse Charging Indication (Collect calls) send/receive option.
3472 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
3473 Dial(DAHDI/g1/extension/R)
3474 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
3475 (requires latest LibPRI)
3476 * Added ability to send/receive keypad digits in the SETUP message.
3477 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
3478 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
3479 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
3480 (requires latest LibPRI)
3481 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
3482 to eliminate tromboned calls. A tromboned call goes out an interface and comes
3483 back into the same interface. Tromboned calls happen because of call routing,
3484 call deflection, call forwarding, and call transfer.
3485 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
3486 * Added the ability to support call waiting calls. (The SETUP has no B channel
3488 * Added Malicious Call ID (MCID) event to the AMI call event class.
3489 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
3491 Asterisk Manager Interface
3492 --------------------------
3493 * The Hangup action now accepts a Cause header which may be used to
3494 set the channel's hangup cause.
3495 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
3496 to specify a separate .pem file to hold a private key. By default sslcert
3497 is used to hold both the public and private key.
3498 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
3499 for options containing the 'tls' prefix. For example, 'sslenable' is now
3500 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
3501 across all .conf files. All affected sample.conf files have been modified to
3502 reflect this change. Previous options such as 'sslenable' still work,
3503 but options with the 'tls' prefix are preferred.
3504 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
3505 in a channel. (res_mutestream.so)
3506 * The configuration file manager.conf now supports a channelvars option, which
3507 specifies a list of channel variables to include in each channel-oriented
3509 * The redirect command now has new parameters ExtraContext, ExtraExtension,
3510 and ExtraPriority to allow redirecting the second channel to a different
3511 location than the first.
3512 * Added new event "JabberStatus" in the Jabber module to monitor buddies
3514 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
3515 in a MixMonitor recording.
3516 * The 'iax2 show peers' output is now similar to the expected output of
3518 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
3520 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
3521 AOC-E messages on a channel.
3522 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
3523 conform more closely to similar events.
3524 * Added a new eventfilter option per user to allow whitelisting and blacklisting
3526 * Added optional parkinglot variable for park command.
3527 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
3528 if CallerIDNum and CallerIDName headers are also present.
3530 Channel Event Logging
3531 ---------------------
3532 * A new interface, CEL, is introduced here. CEL logs single events, much like
3533 the AMI, but it differs from the AMI in that it logs to db backends much
3534 like CDR does; is based on the event subsystem introduced by Russell, and
3535 can share in all its benefits; allows multiple backends to operate like CDR;
3536 is specialized to event data that would be of concern to billing sytems,
3537 like CDR. Backends for logging and accounting calls have been produced,
3538 but a new CDR backend is still in development.
3542 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
3543 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
3544 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
3545 * Multiple files and formats can now be specified in cdr_custom.conf.
3546 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
3547 See configs/cdr_syslog.conf.sample for more information.
3548 * A 'sequence' field has been added to CDRs which can be combined with
3549 linkedid or uniqueid to uniquely identify a CDR.
3550 * Handling of billsec and duration field has changed. If your table definition
3551 specifies those fields as float,double or similar they will now be logged with
3552 microsecond accuracy instead of a whole integer.
3554 Calendaring for Asterisk
3555 ------------------------
3556 * A new set of modules were added supporing calendar integration with Asterisk.
3557 Dialplan functions for reading from and writing to calendars are included,
3558 as well as the ability to execute dialplan logic upon calendar event notifications.
3559 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
3560 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
3561 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
3562 2003 support does not support forms-based authentication).
3564 Call Completion Supplementary Services for Asterisk
3565 ---------------------------------------------------
3566 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
3567 DAHDI/ISDN supports call completion for the following switch types:
3568 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
3569 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
3571 Multicast RTP Support
3572 ---------------------
3573 * A new RTP engine and channel driver have been added which supports Multicast RTP.
3574 The channel driver can be used with the Page application to perform multicast RTP
3575 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
3576 Type can be either basic or linksys.
3577 Destination is the IP address and port for the RTP packets.
3578 Control address is specific to the linksys type and is used for sending the control
3579 packets unique to them.
3581 Security Events Framework
3582 -------------------------
3583 * Asterisk has a new C API for reporting security events. The module res_security_log
3584 sends these events to the "security" logger level. Currently, AMI is the only
3585 Asterisk component that reports security events. However, SIP support will be
3586 coming soon. For more information on the security events framework, see the
3587 "Asterisk Security Framework" section of the Asterisk wiki at
3588 https://wiki.asterisk.org/wiki/x/wgBQ
3589 * SIP support was added in Asterisk 10
3590 * This API now supports IPv6 addresses
3594 * A technology independent fax frontend (res_fax) has been added to Asterisk.
3595 * A spandsp based fax backend (res_fax_spandsp) has been added.
3596 * The app_fax module has been deprecated in favor of the res_fax module and
3597 the new res_fax_spandsp backend.
3598 * The SendFAX and ReceiveFAX applications now send their log messages to a
3599 'fax' logger level, instead of to the generic logger levels. To see these
3600 messages, the system's logger.conf file will need to direct the 'fax' logger
3601 level to one or more destinations; the logger.conf.sample file includes an
3602 example of how to do this. Note that if the 'fax' logger level is *not*
3603 directed to at least one destination, log messages generated by these
3604 applications will be lost, and that if the 'fax' logger level is directed to
3605 the console, the 'core set verbose' and 'core set debug' CLI commands will
3606 have no effect on whether the messages appear on the console or not.
3610 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
3611 Now, in order to enable transmitting silence during record the transmit_silence
3612 option should be used. transmit_silence_during_record remains a valid option, but
3613 defaults to the behavior of the transmit_silence option.
3614 * Addition of the Unit Test Framework API for managing registration and execution
3615 of unit tests with the purpose of verifying the operation of C functions.
3616 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
3617 XMPP text messages to the remote JID.
3618 * Modules.conf has a new option - "require" - that marks a module as critical for
3619 the execution of Asterisk.
3620 If one of the required modules fail to load, Asterisk will exit with a return
3622 * An 'X' option has been added to the asterisk application which enables #exec support.
3623 This allows #exec to be used in asterisk.conf.
3624 * jabber.conf supports a new option auth_policy that toggles auto user registration.
3625 * A new lockconfdir option has been added to asterisk.conf to protect the
3626 configuration directory (/etc/asterisk by default) during reloads.
3627 * The parkeddynamic option has been added to features.conf to enable the creation
3628 of dynamic parkinglots.
3629 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
3630 the reportalarms config option.
3631 * chan_dahdi supports dialing configuring and dialing by device file name.
3632 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
3633 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
3634 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
3635 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
3636 Handy for the above name-based syntax as it does not depend on
3637 initialization order.
3638 * The Realtime dialplan switch now caches entries for 1 second. This provides a
3639 significant increase in performance (about 3X) for installations using this switchtype.
3640 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
3641 AIS. For more information, please see the Distributed Device State section of the
3642 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
3643 * The addition of G.719 pass-through support.
3644 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
3645 during device configuration.
3646 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
3647 have less than 3 lines on the LCD.
3648 * Realtime now supports database failover. See the sample extconfig.conf for details.
3649 * The addition of improved translation path building for wideband codecs. Sample
3650 rate changes during translation are now avoided unless absolutely necessary.
3651 * The addition of the res_stun_monitor module for monitoring and reacting to network
3652 changes while behind a NAT.
3653 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
3654 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
3655 These allow support for any Administration. Default is AT&T values.
3659 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
3660 optionally accept a filename, to apply the setting only to the code generated from
3661 that source file when Asterisk was built. However, there are some modules in Asterisk
3662 that are composed of multiple source files, so this did not result in the behavior
3663 that users expected. In this version, 'core set debug' and 'core set verbose'
3664 can optionally accept *module* names instead (with or without the .so extension),
3665 which applies the setting to the entire module specified, regardless of which source
3666 files it was built from.
3667 * New 'manager show settings' command showing the current settings loaded from
3669 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
3670 the channel hangup request to all channels.
3671 * Added a "core reload" CLI command that executes a global reload of Asterisk.
3673 ------------------------------------------------------------------------------
3674 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
3675 ------------------------------------------------------------------------------
3679 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
3680 Snom phones use this for call pickup of extensions that the phone is
3682 * Added support for setting the domain in the URI for caller of an
3683 outbound call by using the SIPFROMDOMAIN channel variable.
3684 * Added a new configuration option "remotesecret" for authentication to
3685 remote services. For backwards compatibility, "secret" still has the
3686 same function as before, but now you can configure both a remote secret and a
3687 local secret for mutual authentication.
3688 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
3689 the sound will be played to the target of an attended transfer
3690 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
3691 finer control over how many peers Asterisk will qualify and the gap between them
3692 when all peers need to be qualified at the same time.
3693 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
3694 (either globally or for a specific peer), chan_sip will treat any SDP data
3695 it receives as new data and update the media stream accordingly. By
3696 default, Asterisk will only modify the media stream if the SDP session
3697 version received is different from the current SDP session version. This
3698 option is required to interoperate with devices that have non-standard SDP
3699 session version implementations (observed with Microsoft OCS). This option
3700 is disabled by default.
3701 * The parsing of register => lines in sip.conf has been modified to allow a port
3702 to be present in the "user" portion. Please see the sip.conf.sample file for more
3704 * Added support for subscribing to MWI on a remote server and making the status available
3705 as a mailbox. Please see the sip.conf.sample file for more information.
3706 * Added a function to remove SIP headers added in the dialplan before the
3707 first INVITE is generated - SIPRemoveHeader()
3708 * Channel variables set with setvar= in a device configuration is now
3709 set both for inbound and outbound calls.
3710 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
3714 * Added immediate option to iax.conf
3715 * Added forceencryption option to iax.conf
3716 * Added Encryption and Trunk status to manager command "iaxpeers"
3720 * The configuration file now holds separate sections for devices and lines.
3721 Please have a look at configs/skinny.conf.sample and change your skinny.conf
3726 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
3727 support for LibOpenR2. http://www.libopenr2.org/
3728 * The UK option waitfordialtone has been added for use with BT analog
3730 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
3731 is used in conjunction with the 'faxdetect' configuration option. When
3732 'faxbuffers' is used and fax tones are detected, the channel will dynamically
3733 switch to the configured faxbuffers policy. For example, to use 6 buffers
3734 and a 'full' buffer policy for a fax transmission, add:
3736 The faxbuffers configuration will be in affect until the call is torn down.
3737 * Added service message support for 4ESS/5ESS switches.
3741 * For DAHDI channels, the CHANNEL() dialplan function now
3742 supports changing the channel's buffer policy (for the current
3743 call only), using this syntax:
3745 exten => s,n,Set(CHANNEL(buffers)=6,full)
3747 This would change the channel to the 'full' buffer policy and
3748 6 (six) buffers. Possible options for this setting are the same
3749 as those in chan_dahdi.conf.
3750 * Added a new dialplan function, CURLOPT, which permits setting various
3751 options that may be useful with the CURL dialplan function, such as
3752 cookies, proxies, connection timeouts, passwords, etc.
3753 * Permit the syntax and synopsis fields of the corresponding dialplan
3754 functions to be individually set from func_odbc.conf.
3755 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
3756 * func_odbc now may specify an insert query to execute, when the write query
3757 affects 0 rows (usually indicating that no such row exists).
3758 * Added a new dialplan function, LISTFILTER, which permits removing elements
3759 from a set list, by name. Uses the same general syntax as the existing CUT
3760 and FIELDQTY dialplan functions, which also manage lists.
3761 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
3762 obtaining realtime data from the dialplan.
3763 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
3764 a subroutine when using the GoSub() and Return() applications.
3765 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
3766 of "core show function AUDIOHOOK_INHERIT" from the CLI
3767 * Added AES_ENCRYPT. For information on its use, please see the output
3768 of "core show function AES_ENCRYPT" from the CLI
3769 * Added AES_DECRYPT. For information on its use, please see the output
3770 of "core show function AES_DECRYPT" from the CLI
3771 * func_odbc now supports database transactions across multiple queries.
3775 * Scheduled meetme conferences may now have their end times extended by
3777 * app_authenticate now gives the ability to select a prompt other than
3779 * app_directory now pays attention to the searchcontexts setting in
3780 voicemail.conf and will look through all contexts, if no context is
3781 specified in the initial argument.
3782 * A new application, Originate, has been introduced, that allows asynchronous
3783 call origination from the dialplan.
3784 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
3785 in addition to the setting in the "general" context.
3786 * Added ConfBridge dialplan application which does conference bridges without
3787 DAHDI. For information on its use, please see the output of
3788 "core show application ConfBridge" from the CLI.
3792 * The Asterisk CLI has a new command, "channel redirect", which is similar in
3793 operation to the AMI Redirect action.
3794 * extensions.conf now allows you to use keyword "same" to define an extension
3795 without actually specifying an extension. It uses exactly the same pattern
3796 as previously used on the last "exten" line. For example:
3797 exten => 123,1,NoOp(something)
3798 same => n,SomethingElse()
3799 * musiconhold.conf classes of type 'files' can now use relative directory paths,
3800 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
3801 * All deprecated CLI commands are removed from the sourcecode. They are now handled
3802 by the new clialiases module. See cli_aliases.conf.sample file.
3803 * Times within timespecs are now accurate down to the minute. This is a change
3804 from historical Asterisk, which only provided timespecs rounded to the nearest
3805 even (read: evenly divisible by 2) minute mark.
3806 * The realtime switch now supports an option flag, 'p', which disables searches for
3808 * In addition to a time range and date range, timespecs now accept a 5th optional
3809 argument, timezone. This allows you to perform time checks on alternate
3810 timezones, especially if those daylight savings time ranges vary from your
3811 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
3813 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
3814 give you the correct output for an asterisk box behind nat. It will give you the
3815 externhost and localnet settings.
3816 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
3817 can connect calls in passthrough mode, as well as record and play back files.
3818 * Successful and unsuccessful call pickup can now be alerted through sounds, by
3819 using pickupsound and pickupfailsound in features.conf.
3820 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
3821 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
3822 instead of the /var/run/asterisk.pid where it used to be. This will make
3823 installs as non-root easier to manage.
3828 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
3829 be written; they will no longer be explicitly written.
3831 Asterisk Manager Interface
3832 --------------------------
3833 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
3834 a non-empty value) in your request. If you do this, any pending AMI events will
3835 *not* be included in the response to your request as they would normally, but
3836 will be left in the event queue for the next request you make to retrieve. For
3837 some applications, this will allow you to guarantee that you will only see
3838 events in responses to 'WaitEvent' actions, and can better know when to expect them.
3839 To know whether the Asterisk server supports this header or not, your client can
3840 inspect the first response back from the server to see if it includes this header:
3842 Pragma: SuppressEvents
3844 If this is included, the server supports event suppression.
3846 * Added 4 new Actions to list skinny device(s) and line(s)
3852 LDAP Schema File Additions
3853 --------------------------
3854 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
3855 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
3857 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
3858 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
3859 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
3860 * Removed redundant IPaddr (there's already IPAddress)
3861 - Gives more configuration Flags for SIP-Users available (tested)
3862 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
3863 without extensibleObject (which really should be the last resort); gives
3864 also additional possibilities for LDAP-filter
3866 ------------------------------------------------------------------------------
3867 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
3868 ------------------------------------------------------------------------------
3870 Device State Handling
3871 ---------------------
3872 * The event infrastructure in Asterisk got another big update to help support
3873 distributed events. It currently supports distributed device state and
3874 distributed Voicemail MWI (Message Waiting Indication). A new module has
3875 been merged, res_ais, which facilitates communicating events between servers.
3876 It uses the SAForum AIS (Service Availability Forum Application Interface
3877 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
3878 a cluster of Asterisk servers, and to share events between them. For more
3879 information on setting this up, refer to the Distributed Device State section
3880 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
3884 * Added a new dialplan function, AST_CONFIG(), which allows you to access
3885 variables from an Asterisk configuration file.
3886 * The JACK_HOOK function now has a c() option to supply a custom client name.
3887 * Added two new dialplan functions from libspeex for audio gain control and
3888 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
3889 rx directions of a channel from the dialplan.
3890 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
3891 based on other parameters. The default is still to search based on the
3892 forwarding station ID. However, there are new options that allow you to search
3893 based on the message desk terminal ID, or the message desk number.
3894 * TIMEOUT() has been modified to be accurate down to the millisecond.
3895 * ENUM*() functions now include the following new options:
3896 - 'u' returns the full URI and does not strip off the URI-scheme.
3897 - 's' triggers ISN specific rewriting
3898 - 'i' looks for branches into an Infrastructure ENUM tree
3899 - 'd' for a direct DNS lookup without any flipping of digits.
3900 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
3901 * CHANNEL() now has options for the maximum, minimum, and standard or normal
3902 deviation of jitter, rtt, and loss for a call using chan_sip.
3904 DAHDI channel driver (chan_dahdi) Changes
3905 ----------------------------------------
3906 * Channels can now be configured using named sections in chan_dahdi.conf, just
3907 like other channel drivers, including the use of templates.
3908 * The default for pridialplan has changed from 'national' to 'unknown'.
3912 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
3913 to something that matches the pattern a hint will be created using the contents
3914 and variables evaluated.
3915 * Dialplan matching has been extended to allow an extension to return to the
3916 PBX core to wait for more digits. This is done by using the new dialplan
3917 application called "Incomplete". This will permit a whole new level of
3918 extension control, by giving the administrator more control over early
3919 matches employing one of the short-circuit pattern match operators. Note
3920 that custom applications can trigger this same behavior by returning the
3921 special value AST_PBX_INCOMPLETE.
3925 * Directory now permits both first and last names to be matched at the same
3926 time. In addition, the number of digits to enter of the name can be set in
3927 the arguments to Directory; previously, you could enter only 3, regardless
3928 of how many names are in your company. For large companies, this should be
3930 * Voicemail now permits a mailbox setting to wrap around from first to last
3931 messages, if the "messagewrap" option is set to a true value.
3932 * Voicemail now permits an external script to be run, for password validation.
3933 The script should output "VALID" or "INVALID" on stdout, depending upon the
3934 wish to validate or invalidate the password given. Arguments are:
3935 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
3937 * Dial has a new option: F(context^extension^pri), which permits a callee to
3938 continue in the dialplan, at the specified label, if the caller hangs up.
3939 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
3940 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
3941 * The Jack application now has a c() option to supply a custom client name.
3942 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
3943 like the pre-existing whisper mode, except that the spy can also talk to the
3944 participant on the bridged channel as well.
3945 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
3946 to be spoken instead of the channel name or number. For more information on the
3947 use of this option, issue the command "core show application ChanSpy" from the
3949 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
3950 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
3951 words, if using the 'd' option, it is not possible to enter a number to append to
3952 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
3953 change to whisper mode, and pressing 6 will change to barge mode.
3954 * ExternalIVR now takes several options that affect the way it performs, as
3955 well as having several new commands. Please see the External IVR page on the Asterisk
3956 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
3957 * Added ability to communicate over a TCP socket instead of forking a child process for the
3958 ExternalIVR application.
3959 * ChanIsAvail has a new option, 'a', which will return all available channels instead
3960 of just the first one if you give the function more then one channel to check.
3961 * PrivacyManager now takes an option where you can specify a context where the
3962 given number will be matched. This way you have more control over who is allowed
3963 and it stops the people who blindly enter 10 digits.
3964 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
3965 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
3966 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
3967 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
3968 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
3969 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
3970 * The Dial() application no longer copies the language used by the caller to the callee's
3971 channel. If you desire for the caller's channel's language to be used for file playback
3972 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
3973 * SendImage() no longer hangs up the channel on error; instead, it sets the
3974 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
3975 'UNSUPPORTED'. This change makes SendImage() more consistent with other
3977 * Park has a new option, 's', which silences the announcement of the parking space number.
3978 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
3979 invalid input and will be assumed to mean that no timeout is desired.
3983 * Added DNS manager support to registrations for peers referencing peer entries.
3984 DNS manager runs in the background which allows DNS lookups to be run asynchronously
3985 as well as periodically updating the IP address. These properties allow for
3986 better performance as well as recovery in the event of an IP change.
3987 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
3988 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
3989 These changes also provide performance improvements for call setup and tear down.
3990 * Added ability to specify registration expiry time on a per registration basis in
3992 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
3994 * Added t38pt_usertpsource option. See sip.conf.sample for details.
3995 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
3996 * 'sip show peers' and 'sip show users' display their entries sorted in
3997 alphabetical order, as opposed to the order they were in, in the config
3999 * Videosupport now supports an additional option, "always", which always sets
4000 up video RTP ports, even on clients that don't support it. This helps with
4001 callfiles and certain transfers to ensure that if two video phones are
4002 connected, they will always share video feeds.
4006 * Existing DNS manager lookups extended to check for SRV records.
4007 * IAX2 encryption support has been improved to support periodic key rotation
4008 within a call for enhanced security. The option "keyrotate" has been
4009 provided to disable this functionality to preserve backwards compatibility
4010 with older versions of IAX2 that do not support key rotation.
4014 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
4015 data tree based on the given <path>.
4016 * New CLI command "data show providers" that will display all the registered
4018 * New CLI command, "config reload <file.conf>" which reloads any module that
4019 references that particular configuration file. Also added "config list"
4020 which shows which configuration files are in use.
4021 * New CLI commands, "pri show version" and "ss7 show version" that will
4022 display which version of libpri and libss7 are being used, respectively.
4023 A new API call was added so trunk will now have to be compiled against
4024 a versions of libpri and libss7 that have them or it will not know that
4025 these libraries exist.
4026 * The commands "core show globals", "core set global" and "core set chanvar" has
4027 been deprecated in favor of the more semanticly correct "dialplan show globals",
4028 "dialplan set chanvar" and "dialplan set global".
4029 * New CLI command "dialplan show chanvar" to list all variables associated
4030 with a given channel.
4034 * Addresses managed by DNS manager now can check to see if there is a DNS
4035 SRV record for a given domain and will use that hostname/port if present.
4037 AMI - The manager (TCP/TLS/HTTP)
4038 --------------------------------
4039 * The Status command now takes an optional list of variables to display
4040 along with channel status.
4041 * The QueueEntry event now also includes the channel's uniqueid
4045 * res_odbc no longer has a limit of 1023 total possible unshared connections,
4046 as some people were running into this limit. This limit has been increased
4051 * The TRANSFER queue log entry now includes the the caller's original
4052 position in the transferred-from queue.
4053 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
4054 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
4055 as well as an explanation about timeout options in general
4056 * Added a new option - C - for forcing the "answered elsewhere" flag on
4057 cancellation of calls in to members of the queue. This is to avoid the
4058 call to a member of a queue having the call listed as a "missed call".
4062 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
4063 adaptive capabilities. What this means in practical terms is that if your
4064 realtime table lacks critical fields, Asterisk will now emit warnings to
4065 that effect. Also, some of the realtime drivers have the ability (if
4066 configured) to automatically add those columns to the table with the
4067 correct type and length.
4071 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
4072 the 'setvar' option to cause a given audio file to be played upon completion
4073 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
4074 Skinny channels only.
4075 * You can now compile Asterisk against the Hoard Memory Allocator, see the
4076 Hoard page on the Asterisk wiki for more information:
4077 https://wiki.asterisk.org/wiki/x/pQBB
4078 * Config file variables may now be appended to, by using the '+=' append
4079 operator. This is most helpful when working with long SQL queries in
4080 func_odbc.conf, as the queries no longer need to be specified on a single
4082 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
4083 which will add a second to the billsec when the ending
4084 time is set, if the number in the microseconds field of the end time is
4085 greater than the number of microseconds in the answer time. This allows
4086 users to count the 'initiated' seconds in their billing records.
4088 ------------------------------------------------------------------------------
4089 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
4090 ------------------------------------------------------------------------------
4092 AMI - The manager (TCP/TLS/HTTP)
4093 --------------------------------
4094 * Manager has undergone a lot of changes, all of them documented
4095 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
4096 * Manager version has changed to 1.1
4097 * Added a new action 'CoreShowChannels' to list currently defined channels
4098 and some information about them.
4099 * Added a new action 'SIPshowregistry' to list SIP registrations.
4100 * Added TLS support for the manager interface and HTTP server
4101 * Added the URI redirect option for the built-in HTTP server
4102 * The output of CallerID in Manager events is now more consistent.
4103 CallerIDNum is used for number and CallerIDName for name.
4104 * Enable https support for builtin web server.
4105 See configs/http.conf.sample for details.
4106 * Added a new action, GetConfigJSON, which can return the contents of an
4107 Asterisk configuration file in JSON format. This is intended to help
4108 improve the performance of AJAX applications using the manager interface
4110 * SIP and IAX manager events now use "ChannelType" in all cases where we
4111 indicate channel driver. Previously, we used a mixture of "Channel"
4112 and "ChannelDriver" headers.
4113 * Added a "Bridge" action which allows you to bridge any two channels that
4114 are currently active on the system.
4115 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
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