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 * New 'rtpbindaddr' global setting. This allows a user to define which
21 ipaddress to bind the rtpengine too. For example, chan_sip might bind
22 to eth0 (10.0.0.2) but rtpengine to eth1 (192.168.1.10).
24 ------------------------------------------------------------------------------
25 --- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
26 ------------------------------------------------------------------------------
31 Asterisk 13 is the next Long Term Support (LTS) release of Asterisk. As such,
32 the focus of development for this release of Asterisk was on improving the
33 usability and features developed in the previous Standard release, Asterisk 12.
34 Beyond a general refinement of end user features, development focussed heavily
35 on the Asterisk APIs - the Asterisk Manager Interface (AMI) and the Asterisk
36 REST Interface (ARI) - and the PJSIP stack in Asterisk. Some highlights of the
39 * Asterisk security events are now provided via AMI, allowing end users to
40 monitor their Asterisk system in real time for security related issues.
41 * External control of Message Waiting Indicators (MWI) through both AMI and ARI.
42 * Reception/transmission of out of call text messages using any supported
43 channel driver/protocol stack through ARI.
44 * Resource List Server support in the PJSIP stack, providing subscriptions to
45 lists of resources and batched delivery of NOTIFY requests.
46 * Inter-Asterisk distributed device state and mailbox state using the PJSIP
49 It is important to note that Asterisk 13 is built on the architecture developed
50 during the previous Standard release, Asterisk 12. Users upgrading to
51 Asterisk 13 should read about the new features in Asterisk 12 later in this file
52 (see Functionality changes from Asterisk 11 to Asterisk 12), as well as the
53 UPGRADE-12.txt delivered with this release. In particular, users upgrading to
54 Asterisk 13 from a release prior to Asterisk 12 should read the specifications
55 on AMI, CDRs, and CEL on the Asterisk wiki:
56 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
57 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
58 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
60 Many new featuers in Asterisk 13 were introduced in point releases of
61 Asterisk 12. Following this section - which documents the changes from all
62 versions of Asterisk 12 to Asterisk 13 - users should examine the new features
63 that were introduced in the point releases of Asterisk 12, as they are also
64 included in Asterisk 13.
66 Finally, all users upgrading to Asterisk 13 should read the UPGRADE.txt file
67 delivered with this release.
72 * Sample config files have been moved from configs/ to a sub-folder of that
75 * The menuselect utility has been pulled into the Asterisk repository. As a
76 result, the libxml2 development library is now a required dependency for
79 * A new Compiler Flag, REF_DEBUG, has been added. When enabled, reference
80 counted objects will emit additional debug information to the refs log file
81 located in the standard Asterisk log file directory. This log file is useful
82 in tracking down object leaks and other reference counting issues. Prior to
83 this version, this option was only available by modifying the source code
84 directly. This change also includes a new script, refcounter.py, in the
85 contrib folder that will process the refs log file. Note that this replaces
86 the refcounter utility that could be built from the utils directory.
94 * This module was deprecated and has been removed. Users of app_dahdibarge
95 should use ChanSpy instead.
99 * New options to play a beep when starting a recording and stopping a recording
100 have been added. The option "p" will play a beep to the channel that starts
101 the recording. The option "P" will play a beep to the channel that stops the
106 * Queue rules can now be stored in a database table, queue_rules. Unlike other
107 RealTime tables, the queue_rules table is only examined on module load or
108 module reload. A new general setting has been added to queuerules.conf,
109 'realtime_rules', which, when set to 'yes', will cause app_queue to look in
110 RealTime for additional queue rules to parse. Note that both the file and
111 the database can be used as a provide of queue rules when 'realtime_rules'
114 When app_queue is reloaded, all rules are re-parsed and loaded into memory.
115 There is no caching of RealTime queue rules.
119 * This module was deprecated and has been removed. Users of app_readfile
120 should use func_env's FILE function instead.
124 * The 'say' family of dialplan applications now support the Japanese
125 language. The 'language' parameter in say.conf now recognizes a setting of
126 'ja', which will enable Japanese language specific mechanisms for playing
127 back numbers, dates, and other items.
131 * This module was deprecated and has been removed. Users of app_saycountpl
132 should use the Say family of applications.
136 * The SetMusicOnHold dialplan application was deprecated and has been removed.
137 Users of the application should use the CHANNEL function's musicclass
142 * The WaitMusicOnHold dialplan application was deprecated and has been
143 removed. Users of the application should use MusicOnHold with a duration
148 * VoiceMail and VoiceMailMain now support the Japanese language. The
149 'language' parameter in voicemail.conf now recognizes a setting of 'ja',
150 which will enable prompts to be played back using a Japanese grammatical
151 structure. Additional prompts are necessary for this functionality,
153 - jb-arimasu: there is
154 - jb-arimasen: there is not
155 - jb-oshitekudasai: please press
161 * Add the ability to specify multiple email addresses in configuration,
170 * This module was deprecated and has been removed. Users of cdr_sqlite
171 should use cdr_sqlite3_custom.
175 * Added the ability to support PostgreSQL application_name on connections.
176 This allows PostgreSQL to display the configured name in the
177 pg_stat_activity view and CSV log entries. This setting is configurable
178 for cdr_pgsql via the appname configuration setting in cdr_pgsql.conf.
186 * Added the ability to support PostgreSQL application_name on connections.
187 This allows PostgreSQL to display the configured name in the
188 pg_stat_activity view and CSV log entries. This setting is configurable
189 for cel_pgsql via the appname configuration setting in cel_pgsql.conf.
197 * SS7 support now requires libss7 v2.0 or later.
199 * Added SS7 support for connected line and redirecting.
201 * Most SS7 CLI commands are reworked as well as new SS7 commands added.
204 * Added several SS7 config option parameters described in
205 chan_dahdi.conf.sample.
209 * This module was deprecated and has been removed. Users of chan_gtalk
210 should use chan_motif.
214 * This module was deprecated and has been removed. Users of chan_h323
215 should use chan_ooh323.
219 * This module was deprecated and has been removed. Users of chan_jingle
220 should use chan_motif.
224 * The SIPPEER dialplan function no longer supports using a colon as a
225 delimiter for parameters. The parameters for the function should be
226 delimited using a comma.
228 * The SIPCHANINFO dialplan function was deprecated and has been removed. Users
229 of the function should use the CHANNEL function instead.
237 * Added functional peeraccount support. Except for Queue, the
238 accountcode propagation is now consistently propagated to outgoing
239 channels before dialing. The channel accountcode can change from its
240 original non-empty value on channel creation for the following specific
241 reasons. One, dialplan sets it using CHANNEL(accountcode). Two, an
242 originate method that can specify an accountcode value. Three, the
243 calling channel propagates its peeraccount or accountcode to the
244 outgoing channel's accountcode before dialing. The change has two
245 visible effects. One, local channels now cross accountcode and
246 peeraccount across the special bridge between the ;1 and ;2 channels
247 just like channels between normal bridges. Two, the
248 CHANNEL(peeraccount) value can now be set before Dial and FollowMe to
249 set the accountcode on the outgoing channel(s).
251 For Queue, an outgoing channel's non-empty accountcode will not change
252 unless explicitly set by CHANNEL(accountcode). The change has three
253 visible effects. One, local channels now cross accountcode and
254 peeraccount across the special bridge between the ;1 and ;2 channels
255 just like channels between normal bridges. Two, the queue member will
256 get an accountcode if it doesn't have one and one is available from the
257 calling channel's peeraccount. Three, accountcode propagation includes
258 local channel members where the accountcodes are propagated early
259 enough to be available on the ;2 channel.
263 * New DeviceStateChanged and PresenceStateChanged AMI events have been added.
264 These events are emitted whenever a device state or presence state change
265 occurs. The events are controlled by res_manager_device_state.so and
266 res_manager_presence_state.so. If the high frequency of these events is
267 problematic for you, do not load these modules.
269 * Added DialplanExtensionAdd and DialplanExtensionRemove AMI commands. They
270 work in basically the same way as the 'dialplan add extension' and
271 'dialplan remove extension' CLI commands respectively.
273 * New AMI action LoggerRotate reloads and rotates logger in the same manner
274 as CLI command 'logger rotate'
276 * New AMI Actions FAXSessions, FAXSession, and FAXStats replicate the
277 functionality of CLI commands 'fax show sessions', 'fax show session',
278 and fax show stats' respectively.
280 * New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset
281 enable manager control over PRI debugging levels and file output.
283 * AMI action PJSIPNotify may now send to a URI instead of only to a PJSIP
284 endpoint as long as a default outbound endpoint is set. This also applies
285 to the equivalent CLI command (pjsip send notify)
287 * The AMI action PJSIPShowEndpoint now includes ContactStatusDetail sections
288 that give information on Asterisk's attempts to qualify the endpoint.
290 * The DialEnd event will now contain a Forward header if the dial is ending
291 due to the call being forwarded. The contents of the Forward header is the
292 extension in the number to which the call is being forwarded.
296 * The "bridge_technology" extra field key has been added to BRIDGE_ENTER
297 and BRIDGE_EXIT events.
301 * Channel variables are now substituted in arguments passed to applications
302 run by using dynamic features.
306 * The TLS core in Asterisk now supports Perfect Forward Secrecy (PFS).
307 Enabling PFS is attempted by default, and is dependent on the configuration
308 of the module using TLS.
309 - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
310 specify a ECDHE cipher suite in sip.conf, for example:
311 tlscipher=AES128-SHA:DES-CBC3-SHA
312 - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
313 into the private key file, e.g., sip.conf tlsprivatekey. For example, the
314 default dh2048.pem - see
315 http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt
316 - Because clients expect the server to prefer PFS, and because OpenSSL sorts
317 its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
318 Consider re-ordering your cipher suites in the respective configuration
320 tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
321 will use PFS when offered by the client. Clients which do not offer PFS
322 fall-back to AES-128 (or even 3DES, as recommended by RFC 3261).
330 * The JACK_HOOK function now supports audio with a sample rate higher than
339 * Added the ability to support PostgreSQL application_name on connections.
340 This allows PostgreSQL to display the configured name in the
341 pg_stat_activity view and CSV log entries. This setting is configurable
342 for res_config_pgsql via the dbappname configuration setting in
345 res_pjsip_outbound_publish
347 * A new module, res_pjsip_outbound_publish provides the mechanisms for sending
348 PUBLISH requests for specific event packages to another SIP User Agent.
352 * The publish/subscribe core module has been updated to support RFC 4662
353 Resource Lists, allowing Asterisk to act as a Resource List Server (RLS).
354 Resource lists are configured in pjsip.conf under a new object type,
355 resource_list. Resource lists can contain either message-summary or presence
356 events, and can be composed of specific resources that provide the event or
357 other resource lists.
359 * Inbound publication support is provided by a new object, inbound-publication.
360 This configures res_pjsip_pubsub to accept PUBLISH requests from a particular
361 resource. Which events are accepted is constructed dynamically; see
362 res_pjsip_publish_asterisk for more information.
364 res_pjsip_publish_asterisk
366 * A new module, res_pjsip_publish_asterisk adds support for PUBLISH requests of
367 Asterisk information to other Asterisk servers. This module is intended only
368 for Asterisk to Asterisk exchanges of information. Currently, this includes
369 both mailbox state and device state information.
372 ------------------------------------------------------------------------------
373 --- Functionality changes from Asterisk 12.4.0 to Asterisk 12.5.0 ------------
374 ------------------------------------------------------------------------------
378 * Stored recordings now support a new operation, copy. This will take an
379 existing stored recording and copy it to a new location in the recordings
382 * LiveRecording objects now have three additional fields that can be reported
383 in a RecordingFinished ARI event:
384 - total_duration: the duration of the recording
385 - talking_duration: optional. The duration of talking detected in the
386 recording. This is only available if max_silence_seconds was specified
387 when the recording was started.
388 - silence_duration: optional. The duration of silence detected in the
389 recording. This is only available if max_silence_seconds was specified
390 when the recording was started.
391 Note that all duration values are reported in seconds.
393 * Users of ARI can now send and receive out of call text messages. Messages
394 can be sent directly to a particular endpoint, or can be sent to the
395 endpoints resource directly and inferred from the URI scheme. Text
396 messages are passed to ARI clients as TextMessageReceived events. ARI
397 clients can choose to receive text messages by subscribing to the particular
398 endpoint technology or endpoints that they are interested in.
400 * The applications resource now supports subscriptions to all endpoints of
401 a particular channel technology. For example, subscribing to an eventSource
402 of 'endpoint:PJSIP' will subscribe to all PJSIP endpoints.
406 * The endpoint configuration object now supports 'accountcode'. Any channel
407 created for an endpoint with this setting will have its accountcode set
408 to the specified value.
412 * A new module, res_hep_rtcp, has been added that will forward RTCP call
413 statistics to a HEP capture server. See res_hep for more information.
417 * Function AUDIOHOOK_INHERIT has been deprecated. Audiohooks are now
418 unconditionally inhereted through masquerades. As a side benefit, more
419 than one audiohook of a given type may persist through a masquerade now.
421 ------------------------------------------------------------------------------
422 --- Functionality changes from Asterisk 12.3.0 to Asterisk 12.4.0 ------------
423 ------------------------------------------------------------------------------
427 * Returns new AGENT_STATUS value "NOT_CONNECTED" if the agent fails to
428 connect with an incoming caller after being alerted to the presence
429 of the incoming caller. The most likely reason this would happen is
430 the agent did not acknowledge the call in time.
434 * New events have been added for the TALK_DETECT function. When the function
435 is used on a channel, ChannelTalkingStart/ChannelTalkingStop events will be
436 emitted to connected AMI clients indicating the start/stop of talking on
441 * New event models have been aded for the TALK_DETECT function. When the
442 function is used on a channel, ChannelTalkingStarted/ChannelTalkingFinished
443 events will be emitted to connected WebSockets subscribed to the channel,
444 indicating the start/stop of talking on the channel.
448 * A new function, TALK_DETECT, has been added. When set on a channel, this
449 fucntion causes events indicating the starting/stoping of talking on said
450 channel to be emitted to both AMI and ARI clients.
452 ------------------------------------------------------------------------------
453 --- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------
454 ------------------------------------------------------------------------------
458 * A new Playback URI 'tone' has been added. Tones are specified either as
459 an indication name (e.g. 'tone:busy') from indications.conf or as a tone
460 pattern (e.g. 'tone:240/250,0/250'). Tones differ from normal playback
461 URIs in that they must be stopped manually and will continue to occupy
462 a channel's ARI control queue until they are stopped. They also can not
463 be rewound or fastforwarded.
465 * User events can now be generated from ARI. Events can be signalled with
466 arbitrary json variables, and include one or more of channel, bridge, or
467 endpoint snapshots. An application must be specified which will receive
468 the event message (other applications can subscribe to it). The message
469 will also be delivered via AMI provided a channel is attached. Dialplan
470 generated user event messages are still transmitted via the channel, and
471 will only be received by a stasis application they are attached to or if
472 the channel is subscribed to.
476 * SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI
477 fields for prohibited callingpres information. Values are legacy, no, and
478 yes. By default, legacy is used.
479 trust_id_outbound=legacy - behavior remains the same as 1.8.26.1. When
480 dealing with prohibited callingpres and sendrpid=pai/rpid, RPID/PAI
481 headers are appended to outbound SIP messages just as they are with
482 allowed callingpres values, but data about the remote party's identity is
484 When sendrpid=rpid, only the remote party's domain is anonymized.
485 trust_id_outbound=no - when dealing with prohibited callingpres, RPID/PAI
486 headers are not sent.
487 trust_id_outbound=yes - RPID/PAI headers are applied with the full remote
488 party information in tact even for prohibited callingpres information.
489 In the case of PAI, a Privacy: id header will be appended for prohibited
490 calling information to communicate that the private information should
491 not be relayed to untrusted parties.
495 * Manager action 'Park' now takes an additional argument 'AnnounceChannel'
496 which can be used to announce the parked call's location to an arbitrary
497 channel in a bridge. If 'Channel' and 'TimeoutChannel' are now the two
498 parties in a one to one bridge, 'TimeoutChannel' is treated as having
499 parked 'Channel' like with the Park Call DTMF feature and will receive
500 announcements prior to being hung up.
502 ------------------------------------------------------------------------------
503 --- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
504 ------------------------------------------------------------------------------
508 * Record application now has an option 'o' which allows 0 to act as an exit
509 key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'
512 --------------------------
513 * ChanSpy now accepts a channel uniqueid or a fully specified channel name
514 as the chanprefix parameter if the 'u' option is specified.
517 --------------------------
518 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
519 conference user menus.
521 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
522 menus, bridge settings, and user settings that have been applied by the
523 CONFBRIDGE dialplan function.
525 * The ConfBridge dialplan application now sets a channel variable,
526 CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
527 how a channel exited the conference.
529 * Added conference user option 'announce_join_leave_review'. This option
530 implies 'announce_join_leave' with the added effect that the user will
531 be asked if they want to confirm or re-record the recording of their
532 name when entering the conference
535 --------------------------
536 * At exit, the Directory application now sets a channel variable
537 DIRECTORY_RESULT to one of the following based on the reason for exiting:
538 OPERATOR user requested operator by pressing '0' for operator
539 ASSISTANT user requested assistant by pressing '*' for assistant
540 TIMEOUT user pressed nothing and Directory stopped waiting
541 HANGUP user's channel hung up
542 SELECTED user selected a user from the directory and is routed
543 USEREXIT user pressed '#' from the selection prompt to exit
544 FAILED directory failed in a way that wasn't accounted for. Dang.
548 * Monitor() - A new option, B(), has been added that will turn on a periodic
549 beep while the call is being recorded.
552 --------------------------
553 * MusicOnHold streams (all modes other than "files") now support wide band
557 --------------------------
558 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
559 and for the channel executing Page respectively.
562 --------------------------
563 * PickupChan now accepts channel uniqueids of channels to pickup.
566 --------------------------
567 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
568 to 'true' (case insensitive), then any Say application (SayNumber,
569 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
570 anticipate DTMF. If DTMF is received, these applications will behave like
571 the background application and jump to the received extension once a match
572 is established or after a short period of inactivity.
575 -------------------------
576 * A new function, MIXMONITOR, has been added to allow access to individual
577 instances of MixMonitor on a channel.
579 * A new option, B(), has been added that will turn on a periodic beep while the
580 call is being recorded.
584 -------------------------
587 -------------------------
588 * TEL URI support for inbound INVITE requests has been added. chan_sip will
589 now handle TEL schemes in the Request and From URIs. The phone-context in
590 the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on
595 * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
596 the new AST_SORCERY diaplan function.
598 * Core Show Locks output now includes Thread/LWP ID if the platform
599 supports this feature.
601 * New "logger add channel" and "logger remove channel" CLI commands have
602 been added to allow creation and deletion of dynamic logger channels
603 without configuration changes. These dynamic logger channels will only
604 exist until the next restart of asterisk.
608 * The live recording object on recording events now contains a target_uri
609 field which contains the URI of what is being recorded.
611 * The bridge type used when creating a bridge is now a comma separated list of
612 bridge properties. Valid options are: mixing, holding, dtmf_events, and
615 * A channelId can now be provided when creating a channel, either in the
616 uri (POST channels/my-channel-id) or as query parameter. A local channel
617 will suffix the second channel id with ';2' unless provided as query
618 parameter otherChannelId.
620 * A bridgeId can now be provided when creating a bridge, either in the uri
621 (POST bridges/my-bridge-id) or as a query parameter.
623 * A playbackId can be provided when starting a playback, either in the uri
624 (POST channels/my-channel-id/play/my-playback-id /
625 POST bridges/my-bridge-id/play/my-playback-id) or as a query parameter.
627 * A snoop channel can be started with a snoopId, in the uri or query.
631 * Originate now takes optional parameters ChannelId and OtherChannelId,
632 used to set the UniqueId on creation. The other id is assigned to the
633 second channel when dialing LOCAL, or defaults to appending ;2 if only
634 the single Id is given.
636 * The Mixmonitor action now has a "Command" header that can be used to
637 indicate a post-process command to run once recording finishes.
641 * A new set of Alembic scripts has been added for CDR tables. This will create
642 a 'cdr' table with the default schema that Asterisk expects.
647 * A new function was added: PERIODIC_HOOK. This allows running a periodic
648 dialplan hook on a channel. Any audio generated by this hook will be
649 injected into the call.
657 * A new module, res_hep, has been added, that acts as a generic packet
658 capture agent for the Homer Encapsulation Protocol (HEP) version 3.
659 It can be configured via hep.conf. Other modules can use res_hep to send
660 message traffic to a HEP capture server.
664 * A new module, res_hep_pjsip, has been added that will forward PJSIP
665 message traffic to a HEP capture server. See res_hep for more
670 * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
671 be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
673 * Added the following new CLI commands:
674 - "pjsip show contacts" - list all current PJSIP contacts.
675 - "pjsip show contact" - show specific information about a current PJSIP
677 - "pjsip show channel" - show detailed information about a PJSIP channel.
681 * A new module, res_pjsip_multihomed handles situations where the system
682 Asterisk is running out has multiple interfaces. res_pjsip_multihomed
683 determines which interface should be used during message sending.
685 res_pjsip_pidf_digium_body_supplement
687 * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
688 request body formatting for presence support in Digium phones.
690 res_pjsip_send_to_voicemail
692 * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
693 particular headers to transfer a PJSIP channel directly to a particular
694 extension that has VoiceMail. This is intended to be used with Digium
695 phones that support this feature.
697 res_pjsip_outbound_registration
699 * A new CLI command has been added: "pjsip show registrations", which lists
700 all configured PJSIP registrations
703 ------------------------------------------------------------------------------
704 --- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
705 ------------------------------------------------------------------------------
709 * Added a new module that provides AMI control over MWI within Asterisk,
710 res_mwi_external_ami. Note that this module depends on res_mwi_external;
711 for more information on enabling this module, see res_mwi_external.
712 This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
713 the MWIGet/MWIGetComplete events.
715 * The DialStatus field in the DialEnd event can now contain additional
716 statuses that convey how the dial operation terminated. This includes
717 ABORT, CONTINUE, and GOTO.
719 * AMI will now emit security events. A new class authorization has been
720 added in manager.conf for the security events, 'security'. The new events
722 - FailedACL - raised when a request violates an ACL check
723 - InvalidAccountID - raised when a request fails an authentication
724 check due to an invalid account ID
725 - SessionLimit - raised when a request fails due to exceeding the
726 number of allowed concurrent sessions for a service
727 - MemoryLimit - raised when a request fails due to an internal memory
729 - LoadAverageLimit - raised when a request fails because a configured
730 load average limit has been reached
731 - RequestNotAllowed - raised when a request is not allowed by
733 - AuthMethodNotAllowed - raised when a request used an authentication
734 method not allowed by the service
735 - RequestBadFormat - raised when a request is received with bad formatting
736 - SuccessfulAuth - raised when a request successfully authenticates
737 - UnexpectedAddress - raised when a request has a different source address
738 then what is expected for a session already in progress with a service
739 - ChallengeResponseFailed - raised when a request's attempt to authenticate
740 has been challenged, and the request failed the authentication challenge
741 - InvalidPassword - raised when a request provides an invalid password
742 during an authentication attempt
743 - ChallengeSent - raised when an Asterisk service send an authentication
744 challenge to a request
745 - InvalidTransport - raised when a request attempts to use a transport not
746 allowed by the Asterisk service
748 * Bridge related events now have two additional fields: BridgeName and
749 BridgeCreator. BridgeName is a descriptive name for the bridge;
750 BridgeCreator is the name of the entity that created the bridge. This
751 affects the following events: ConfbridgeStart, ConfbridgeEnd,
752 ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
753 ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
754 AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
758 * The Bridge data model now contains the additional fields 'name' and
759 'creator'. The 'name' field conveys a descriptive name for the bridge;
760 the 'creator' field conveys the name of the entity that created the bridge.
761 This affects all responses to HTTP requests that return a Bridge data model
762 as well as all event derived data models that contain a Bridge data model.
763 The POST /bridges operation may now optionally specify a name to give to
764 the bridge being created.
766 * Added a new ARI resource 'mailboxes' which allows the creation and
767 modification of mailboxes managed by external MWI. Modules res_mwi_external
768 and res_stasis_mailbox must be enabled to use this resource. For more
769 information on external MWI control, see res_mwi_external.
771 * Added new events for externally initiated transfers. The event
772 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
773 of a bridge in the ARI controlled application to the dialplan; the
774 BridgeAttendedTransfer event is raised when a channel initiates an
775 attended transfer of a bridge in the ARI controlled application to the
778 * Channel variables may now be specified as a body parameter to the
779 POST /channels operation. The 'variables' key in the JSON is interpreted
780 as a sequence of key/value pairs that will be added to the created channel
781 as channel variables. Other parameters in the JSON body are treated as
782 query parameters of the same name.
786 * Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be
787 automatically handled by the HTTP server if a request is received with a
788 Transfer-Encoding type of "chunked".
792 * Path support has been added with the 'support_path' option in registration
795 * A 'debug' option has been added to the globals section that will allow
796 sip messages to be logged.
798 * A 'set_var' option has been added to endpoints that will automatically
799 set the desired variable(s) on a channel created for that endpoint.
801 * Several new tables and columns have been added to the realtime schema for
802 the res_pjsip related modules. See the UPGRADE.txt notes for updating
807 * A new module, res_mwi_external, has been added to Asterisk. This module
808 acts as a base framework that other modules can build on top of to allow
809 an external system to control MWI within Asterisk. For implementations
810 that make use of res_mwi_external, see res_mwi_external_ami and
811 res_ari_mailboxes. Note that res_mwi_external canflicts with other modules
812 that may produce MWI themselves, such as app_voicemail. res_mwi_external
813 and other modules that depend on it cannot be built or loaded with
814 app_voicemail present.
818 * DNS functionality will now automatically be enabled if the system configured
819 nameservers can be retrieved. If the system configured nameservers can not be
820 retrieved the functionality will resort to using system resolution. Functionalty
821 such as SRV records and failover will not be available if system resolution
824 ------------------------------------------------------------------------------
825 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
826 ------------------------------------------------------------------------------
831 Asterisk 12 is a standard release of the Asterisk project. As such, the
832 focus of development for this release was on core architectural changes and
833 major new features. This includes:
834 * A more flexible bridging core based on the Bridging API
835 * A new internal message bus, Stasis
836 * Major standardization and consistency improvements to AMI
837 * Addition of the Asterisk RESTful Interface (ARI)
838 * A new SIP channel driver, chan_pjsip
839 In addition, as the vast majority of bridging in Asterisk was migrated to the
840 Bridging API used by ConfBridge, major changes were made to most of the
841 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
843 Specifications have been written for the affected interfaces. These
844 specifications are available on the Asterisk wiki:
845 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
846 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
847 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
849 It is *highly* recommended that anyone migrating to Asterisk 12 read the
850 information regarding its release both in this file and in the accompanying
851 UPGRADE.txt file. More detailed information on the major changes can be found
852 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
857 * Added build option DISABLE_INLINE. This option can be used to work around a
858 bug in gcc. For more information, see
859 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
861 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
862 the CHANNEL_TRACE build option were incompatible with the new bridging
865 * Asterisk now optionally uses libxslt to improve XML documentation generation
866 and maintainability. If libxslt is not available on the system, some XML
867 documentation will be incomplete.
869 * Asterisk now depends on libjansson. If a package of libjansson is not
870 available on your distro, please see http://www.digip.org/jansson/.
872 * Asterisk now depends on libuuid and, optionally, uriparser. It is
873 recommended that you install uriparser, even if it is optional.
875 * The new SIP stack and channel driver uses a particular version of PJSIP.
876 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
877 configuring and installing PJSIP for usage with Asterisk.
879 * Optional API was re-implemented to be more portable, and no longer requires
880 weak reference support from the compiler. The build option OPTIONAL_API may
881 be disabled to disable Optional API support.
888 * Along with AgentRequest, this application has been modified to be a
889 replacement for chan_agent. The act of a channel calling the AgentLogin
890 application places the channel into a pool of agents that can be
891 requested by the AgentRequest application. Note that this application, as
892 well as all other agent related functionality, is now provided by the
893 app_agent_pool module. See chan_agent and AgentRequest for more information.
895 * This application no longer performs agent authentication. If authentication
896 is desired, the dialplan needs to perform this function using the
897 Authenticate or VMAuthenticate application or through an AGI script before
900 * If this application is called and the agent is already logged in, the
901 dialplan will continue exection with the AGENT_STATUS channel variable set
902 to ALREADY_LOGGED_IN.
904 * The agents.conf schema has changed. Rather than specifying agents on a
905 single line in comma delineated fashion, each agent is defined in a separate
906 context. This allows agents to use the power of context templates in their
909 * A number of parameters from agents.conf have been removed. This includes
910 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
911 urlprefix, and savecallsin. These options were obsoleted by the move from
912 a channel driver model to the bridging/application model provided by
917 * A new application, this will request a logged in agent from the pool and
918 bridge the requested channel with the channel calling this application.
919 Logged in agents are those channels that called the AgentLogin application.
920 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
921 application will be set with an appropriate error value.
925 * This application has been removed. It was a holdover from when
926 AgentCallbackLogin was removed.
930 * Added support for additional Ademco DTMF signalling formats, including
931 Express 4+1, Express 4+2, High Speed and Super Fast.
933 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
934 call time, in milliseconds, to run the application.
936 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
937 maximum number of times to retry the call.
939 * Added a new configuration option answait. If set, the AlarmReceiver
940 application will wait the number of milliseconds specified by answait
941 after the channel has answered. Valid values range between 500
942 milliseconds and 10000 milliseconds.
944 * Added configuration option no_group_meta. If enabled, grouping of metadata
945 information in the AlarmReceiver log file will be skipped.
949 * It is now no longer possible to bypass updating the CDR on the channel
950 when answering. CDRs reflect the state of the channel and will always
951 reflect the time they were Answered.
955 * A new application in Asterisk, this will place the calling channel
956 into a holding bridge, optionally entertaining them with some form of
957 media. Channels participating in a holding bridge do not interact with
958 other channels in the same holding bridge. Optionally, however, a channel
959 may join as an announcer. Any media passed from an announcer channel is
960 played to all channels in the holding bridge. Channels leave a holding
961 bridge either when an optional timer expires, or via the ChannelRedirect
962 application or AMI Redirect action.
966 * All participants in a bridge can now be kicked out of a conference room
967 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
968 command, i.e., 'confbridge kick <conference> all'
970 * CLI output for the 'confbridge list' command has been improved. When
971 displaying information about a particular bridge, flags will now be shown
972 for the participating users indicating properties of that user.
974 * The ConfbridgeList event now contains the following fields: WaitMarked,
975 EndMarked, and Waiting. This displays additional properties about the
976 user's profile, as well as whether or not the user is waiting for a
977 Marked user to enter the conference.
979 * Added a new option for conference recording, record_file_append. If enabled,
980 when the recording is stopped and then re-started, the existing recording
981 will be used and appended to.
983 * ConfBridge now has the ability to set the language of announcements to the
984 conference. The language can be set on a bridge profile in confbridge.conf
985 or by the dialplan function CONFBRIDGE(bridge,language)=en.
989 * The channel variable CPLAYBACKSTATUS may now return the value
990 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
991 such as AMI. See the AMI action ControlPlayback for more information.
995 * Added the 'a' option, which allows the caller to enter in an additional
996 alias for the user in the directory. This option must be used in conjunction
997 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
998 specified in voicemail.conf.
1002 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
1003 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
1004 containing the unique ID of the bridge that the channel happens to be in.
1008 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
1009 for more information.
1011 * Variables are no longer purged from the original CDR. See the 'v' option for
1014 * The 'A' option has been removed. The Answer time on a CDR is never updated
1017 * The 'd' option has been removed. The disposition on a CDR is a function of
1018 the state of the channel and cannot be altered.
1020 * The 'D' option has been removed. Who the Party B is on a CDR is a function
1021 of the state of the respective channels involved in the CDR and cannot be
1024 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
1025 such that the start time and, if applicable, the answer time was updated.
1026 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
1027 'r' option now triggers the Reset, setting the start time (and answer time
1028 if applicable) to the current time. Note that the 'a' option still sets
1029 the answer time to the current time if the channel was already answered.
1031 * The 's' option has been removed. A variable can be set on the original CDR
1032 if desired using the CDR function, and removed from a forked CDR using the
1035 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
1036 longer applies in the CDR engine.
1038 * The 'v' option now prevents the copy of the variables from the original CDR
1039 to the forked CDR. Previously the variables were always copied but were
1040 removed from the original. This was changed as removing variables from a CDR
1041 can have unintended side effects - this option allows the user to prevent
1042 propagation of variables from the original to the forked without modifying
1047 * Added the 'n' option to MeetMe to prevent application of the DENOISE
1048 function to a channel joining a conference. Some channel drivers that vary
1049 the number of audio samples in a voice frame will experience significant
1050 quality problems if a denoiser is attached to the channel; this option gives
1051 them the ability to remove the denoiser without having to unload func_speex.
1055 * The 'b' option now includes conferences as well as sounds played to the
1058 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
1059 running during a transfer. If a MixMonitor is started on a channel,
1060 the MixMonitor will continue to record the audio passing through the
1061 channel even in the presence of transfers.
1065 * The NoCDR application is deprecated. Please use the CDR_PROP function to
1068 * While the NoCDR application will prevent CDRs for a channel from being
1069 propagated to registered CDR backends, it will not prevent that data from
1070 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
1071 function that enables CDRs on a channel will restore those records that have
1072 not yet been finalized.
1076 * The app_parkandannounce module has been removed. The application
1077 ParkAndAnnounce is now provided by the res_parking module. See the
1078 res_parking changes for more information.
1082 * Added queue available hint. The hint can be added to the dialplan using the
1083 following syntax: exten,hint,Queue:{queue_name}_avail
1084 For example, if the name of the queue is 'markq':
1085 exten => 8501,hint,Queue:markq_avail
1086 This will report 'InUse' if there are no logged in agents or no free agents.
1087 It will report 'Idle' when an agent is free.
1089 * Queues now support a hint for member paused state. The hint uses the form
1090 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
1091 are the name of the queue and the name of the member to subscribe to,
1092 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
1093 Members will show as In Use when paused.
1095 * The configuration options eventwhencalled and eventmemberstatus have been
1096 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
1097 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
1098 sent. The "Variable" fields will also no longer exist on the Agent* events.
1099 These events can be filtered out from a connected AMI client using the
1100 eventfilter setting in manager.conf.
1102 * The queue log now differentiates between blind and attended transfers. A
1103 blind transfer will result in a BLINDTRANSFER message with the destination
1104 context and extension. An attended transfer will result in an
1105 ATTENDEDTRANSFER message. This message will indicate the method by which
1106 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
1107 for running an application on a bridge or channel, or "LINK" for linking
1108 two bridges together with local channels. The queue log will also now detect
1109 externally initiated blind and attended transfers and record the transfer
1112 * When performing queue pause/unpause on an interface without specifying an
1113 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
1114 least one member of any queue exists for that interface.
1116 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
1117 for realtime queue log entries.
1121 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
1122 CDRs when they were previously disabled on a channel.
1124 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
1125 backends occurs on an as-needed basis in order to preserve linkedid
1126 propagation and other needed behavior.
1130 * A new application, this is similar to SayAlpha except that it supports
1131 case sensitive playback of the specified characters. For example,
1132 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
1136 * This application is deprecated in favor of CHANNEL(amaflags).
1140 * The SendDTMF application will now accept 'W' as valid input. This will cause
1141 the application to delay one second while streaming DTMF.
1145 * A new application in Asterisk 12, this hands control of the channel calling
1146 the application over to an external system. Currently, external systems
1147 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
1151 * UserEvent will now handle duplicate keys by overwriting the previous value
1152 assigned to the key.
1154 * In addition to AMI, UserEvent invocations will now be distributed to any
1155 interested Stasis applications.
1159 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1160 system as mailbox@context. The rest of the system cannot add @default
1161 to mailbox identifiers for app_voicemail that do not specify a context
1162 any longer. It is a mailbox identifier format that should only be
1163 interpreted by app_voicemail.
1165 * The voicemail.conf configuration file now has an 'alias' configuration
1166 parameter for use with the Directory application. The voicemail realtime
1167 database table schema has also been updated with an 'alias' column.
1172 * Pass through support has been added for both VP8 and Opus.
1174 * Added format attribute negotiation for the Opus codec. Format attribute
1175 negotiation is provided by the res_format_attr_opus module.
1180 * Masquerades as an operation inside Asterisk have been effectively hidden
1181 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
1182 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
1183 dropping of frame/audio hooks, and other internal implementation details
1184 that users had to deal with. This fundamental change has large implications
1185 throughout the changes documented for this version. For more information
1186 about the new core architecture of Asterisk, please see the Asterisk wiki.
1188 * Multiple parties in a bridge may now be transferred. If a participant in a
1189 multi-party bridge initiates a blind transfer, a Local channel will be used
1190 to execute the dialplan location that the transferer sent the parties to. If
1191 a participant in a multi-party bridge initiates an attended transfer,
1192 several options are possible. If the attended transfer results in a transfer
1193 to an application, a Local channel is used. If the attended transfer results
1194 in a transfer to another channel, the resulting channels will be merged into
1197 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
1198 driver specific. If the channel variable is set on the transferrer channel,
1199 the sound will be played to the target of an attended transfer.
1201 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
1202 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
1203 listed. Any more peers in the bridge will not be included in the list.
1204 BRIDGEPEER is not valid in holding bridges like parking since those channels
1205 do not talk to each other even though they are in a bridge.
1207 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
1208 and will contain a value if the BRIDGEPEER's channel driver supports it.
1210 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
1211 was responsible for an attended transfer in a similar fashion to
1214 * Modules using the Configuration Framework or Sorcery must have XML
1215 configuration documentation. This configuration documentation is included
1216 with the rest of Asterisk's XML documentation, and is accessible via CLI
1217 commands. See the CLI changes for more information.
1219 AMI (Asterisk Manager Interface)
1221 * Major changes were made to both the syntax as well as the semantics of the
1222 AMI protocol. In particular, AMI events have been substantially improved
1223 in this version of Asterisk. For more information, please see the AMI
1224 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
1226 * AMI events that reference a particular channel or bridge will now always
1227 contain a standard set of fields. When multiple channels or bridges are
1228 referenced in an event, fields for at least some subset of the channels
1229 and bridges in the event will be prefixed with a descriptive name to avoid
1230 name collisions. See the AMI event documentation on the Asterisk wiki for
1233 * The CLI command 'manager show commands' no longer truncates command names
1234 longer than 15 characters and no longer shows authorization requirement
1235 for commands. 'manager show command' now displays the privileges needed
1236 for using a given manager command instead.
1238 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
1239 peer in its response if the peer has a subscribe context set.
1241 * The SIPqualifypeer action now acknowledges the request once it has
1242 established that the request is against a known peer. It also issues a new
1243 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
1245 * The PlayDTMF action now supports an optional 'Duration' parameter. This
1246 specifies the duration of the digit to be played, in milliseconds.
1248 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
1249 updates when changes occur instead of requiring the use of pollmailboxes.
1251 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
1252 AMI client to manipulate audio currently being played back on a channel. The
1253 supported operations depend on the application being used to send audio to
1254 the channel. When the audio playback was initiated using the ControlPlayback
1255 application or CONTROL STREAM FILE AGI command, the audio can be paused,
1256 stopped, restarted, reversed, or skipped forward. When initiated by other
1257 mechanisms (such as the Playback application), the audio can be stopped,
1258 reversed, or skipped forward.
1260 * Channel related events now contain a snapshot of channel state, adding new
1261 fields to many of these events.
1263 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
1264 in a future release. Please use the common 'Exten' field instead.
1266 * The AMI event 'UserEvent' from app_userevent now contains the channel state
1267 fields. The channel state fields will come before the body fields.
1269 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
1270 'UnParkedCall' have changed significantly in the new res_parking module.
1272 The 'Channel' and 'From' headers are gone. For the channel that was parked
1273 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
1274 has a number of fields associated with it. The old 'Channel' header relayed
1275 the same data as the new 'ParkeeChannel' header.
1277 The 'From' field was ambiguous and changed meaning depending on the event.
1278 for most of these, it was the name of the channel that parked the call
1279 (the 'Parker'). There is no longer a header that provides this channel name,
1280 however the 'ParkerDialString' will contain a dialstring to redial the
1281 device that parked the call.
1283 On UnParkedCall events, the 'From' header would instead represent the
1284 channel responsible for retrieving the parkee. It receives a channel
1285 snapshot labeled 'Retriever'. The 'from' field is is replaced with
1288 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
1290 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
1291 fashion has changed the field names 'StartExten' and 'StopExten' to
1292 'StartSpace' and 'StopSpace' respectively.
1294 * The deprecated use of | (pipe) as a separator in the channelvars setting in
1295 manager.conf has been removed.
1297 * Channel Variables conveyed with a channel no longer contain the name of the
1298 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
1299 ChanVariable: bar=baz. When multiple channels are present in a single AMI
1300 event, the various ChanVariable fields will contain a suffix that specifies
1301 which channel they correspond to.
1303 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
1304 event always conveys the AMI event for a particular channel.
1306 * All 'Reload' events have been consolidated into a single event type. This
1307 event will always contain a Module field specifying the name of the module
1308 and a Status field denoting the result of the reload. All modules now issue
1309 this event when being reloaded.
1311 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
1312 fail to receive this event due to being connected after modules have loaded.
1313 AMI connections that want to know when Asterisk is ready should listen for
1314 the 'FullyBooted' event.
1316 * app_fax now sends the same send fax/receive fax events as res_fax. The
1317 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
1318 now the 'ReceiveFAX' event.
1320 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
1321 'MusicOnHoldStop'. The sub type field has been removed.
1323 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
1324 carrier for another protocol.
1326 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
1327 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
1328 to the specific channel. 'Both' may be specified to play a tone to both
1329 channels. The old 'yes' option is still accepted as a way of playing the
1330 tone to Channel2 only.
1332 * The AMI 'Status' response event to the AMI Status action replaces the
1333 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
1334 indicate what bridge the channel is currently in.
1336 * The AMI 'Hold' event has been moved out of individual channel drivers, into
1337 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
1340 * The AMI events in app_queue have been made more consistent with each other.
1341 Events that reference channels (QueueCaller* and Agent*) will show
1342 information about each channel. The (infamous) 'Join' and 'Leave' AMI
1343 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
1345 * The 'MCID' AMI event now publishes a channel snapshot when available and
1346 its non-channel-snapshot parameters now use either the "MCallerID" or
1347 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
1348 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
1349 parameters in the channel snapshot.
1351 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
1352 'AgentLogin' and 'AgentLogoff' respectively.
1354 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
1355 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
1357 * 'ChannelUpdate' events have been removed.
1359 * All AMI events now contain a 'SystemName' field, if available.
1361 * Local channel optimization is now conveyed in two events:
1362 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
1363 when the Local channel driver begins attempting to optimize itself out of
1364 the media path; the End event is sent after the channel halves have
1365 successfully optimized themselves out of the media path.
1367 * Local channel information in events is now prefixed with 'LocalOne' and
1368 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
1369 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
1370 and 'LocalOptimizationEnd' events.
1372 * The option 'allowmultiplelogin' can now be set or overriden in a particular
1373 account. When set in the general context, it will act as the default
1374 setting for defined accounts.
1376 * The 'BridgeAction' event was removed. It technically added no value, as the
1377 Bridge Action already receives confirmation of the bridge through a
1378 successful completion Event.
1380 * The 'BridgeExec' events were removed. These events duplicated the events that
1381 occur in the Briding API, and are conveyed now through BridgeCreate,
1382 BridgeEnter, and BridgeLeave events.
1384 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
1385 previous versions. They now report all SR/RR packets sent/received, and
1386 have been restructured to better reflect the data sent in a SR/RR. In
1387 particular, the event structure now supports multiple report blocks.
1389 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
1390 raised when a blind transfer/attended transfer completes successfully.
1391 They contain information about the transfer that just completed, including
1392 the location of the transfered channel.
1394 * Added a 'security' class to AMI which outputs the required fields for
1395 security messages similar to the log messages from res_security_log
1397 * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
1398 that describes the status value in a human readable string.
1400 CDR (Call Detail Records)
1402 * Significant changes have been made to the behavior of CDRs. The CDR engine
1403 was effectively rewritten and built on the Stasis message bus. For a full
1404 definition of CDR behavior in Asterisk 12, please read the specification
1405 on the Asterisk wiki (wiki.asterisk.org).
1407 * CDRs will now be created between all participants in a bridge. For each
1408 pair of channels in a bridge, a CDR is created to represent the path of
1409 communication between those two endpoints. This lets an end user choose who
1410 to bill for what during bridge operations with multiple parties.
1412 * The duration, billsec, start, answer, and end times now reflect the times
1413 associated with the current CDR for the channel, as opposed to a cumulative
1414 measurement of all CDRs for that channel.
1416 * When a CDR is dispatched, user defined CDR variables from both parties are
1417 included in the resulting CDR. If both parties have the same variable, only
1418 the Party A value is provided.
1420 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
1421 information regarding the CDR engine is logged as verbose messages. This
1422 option should only be used if the behavior of the CDR engine needs to be
1425 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
1426 normally configured in cdr.conf.
1428 * Added CLI command 'cdr show active {channel}'. When {channel} is not
1429 specified, this command provides a summary of the channels with CDR
1430 information and their statistics. When {channel} is specified, it shows
1431 detailed information about all records associated with {channel}.
1433 CEL (Channel Event Logging)
1435 * CEL has undergone significant rework in Asterisk 12, and is now built on the
1436 Stasis message bus. Please see the specification for CEL on the Asterisk
1437 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
1440 * The 'extra' field of all CEL events that use it now consists of a JSON blob
1441 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
1443 * BLINDTRANSFER events now report the transferee bridge unique
1444 identifier, extension, and context in a JSON blob as the extra string
1445 instead of the transferee channel name as the peer.
1447 * ATTENDEDTRANSFER events now report the peer as NULL and additional
1448 information in the 'extra' string as a JSON blob. For transfers that occur
1449 between two bridged channels, the 'extra' JSON blob contains the primary
1450 bridge unique identifier, the secondary channel name, and the secondary
1451 bridge unique identifier. For transfers that occur between a bridged channel
1452 and a channel running an app, the 'extra' JSON blob contains the primary
1453 bridge unique identifier, the secondary channel name, and the app name.
1455 * LOCAL_OPTIMIZE events have been added to convey local channel
1456 optimizations with the record occurring for the semi-one channel and
1457 the semi-two channel name in the peer field.
1459 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
1460 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
1461 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
1462 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
1463 regardless of whether or not that bridge happens to contain multiple
1468 * When compiled with '--enable-dev-mode', the astobj2 library will now add
1469 several CLI commands that allow for inspection of ao2 containers that
1470 register themselves with astobj2. The CLI commands are 'astobj2 container
1471 dump', 'astobj2 container stats', and 'astobj2 container check'.
1473 * Added specific CLI commands for bridge inspection. This includes 'bridge
1474 show all', which lists all bridges in the system, and 'bridge show {id}',
1475 which provides specific information about a bridge.
1477 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
1478 ejecting the channels currently in the bridge. If the channels cannot
1479 continue in the dialplan or application that put them in the bridge, they
1482 * Added command 'bridge kick'. This will eject a single channel from a bridge.
1484 * Added commands to inspect and manipulate the registered bridge technologies.
1485 This include 'bridge technology show', which lists the registered bridge
1486 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
1487 which controls whether or not a registered bridge technology can be used
1488 during smart bridge operations. If a technology is suspended, it will not
1489 be used when a bridge technology is picked for channels; when unsuspended,
1490 it can be used again.
1492 * The command 'config show help {module} {type} {option}' will show
1493 configuration documentation for modules with XML configuration
1494 documentation. When {module}, {type}, and {option} are omitted, a listing
1495 of all modules with registered documentation is displayed. When {module}
1496 is specified, a listing of all configuration types for that module is
1497 displayed, along with their synopsis. When {module} and {type} are
1498 specified, a listing of all configuration options for that type are
1499 displayed along with their synopsis. When {module}, {type}, and {option}
1500 are specified, detailed information for that configuration option is
1503 * Added 'core show sounds' and 'core show sound' CLI commands. These display
1504 a listing of all installed media sounds available on the system and
1505 detailed information about a sound, respectively.
1507 * 'xmldoc dump' has been added. This CLI command will dump the XML
1508 documentation DOM as a string to the specified file. The Asterisk core
1509 will populate certain XML elements pulled from the source files with
1510 additional run-time information; this command lets a user produce the
1511 XML documentation with all information.
1515 * Parking has been pulled from core and placed into a separate module called
1516 res_parking. See Parking changes below for more details. Configuration for
1517 parking should now be performed in res_parking.conf. Configuration for
1518 parking in features.conf is now unsupported.
1520 * Core attended transfers now have several new options. While performing an
1521 attended transfer, the transferer now has the following options:
1522 - *1 - cancel the attended transfer (configurable via atxferabort)
1523 - *2 - complete the attended transfer, dropping out of the call
1524 (configurable via atxfercomplete)
1525 - *3 - complete the attended transfer, but stay in the call. This will turn
1526 the call into a multi-party bridge (configurable via atxferthreeway)
1527 - *4 - swap to the other party. Once an attended transfer has begun, this
1528 options may be used multiple times (configurable via atxferswap)
1530 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
1531 must be on the channel initiating the transfer to have any effect.
1533 * The BRIDGE_FEATURES channel variable would previously only set features for
1534 the calling party and would set this feature regardless of whether the
1535 feature was in caps or in lowercase. Use of a caps feature for a letter
1536 will now apply the feature to the calling party while use of a lowercase
1537 letter will apply that feature to the called party.
1539 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
1541 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
1542 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
1543 activated the dynamic feature.
1545 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
1546 only on the channel executing the dynamic feature. Executing a dynamic
1547 feature on the bridge peer in a multi-party bridge will execute it on all
1548 peers of the activating channel.
1550 * You can now have the settings for a channel updated using the FEATURE()
1551 and FEATUREMAP() functions inherited to child channels by setting
1552 FEATURE(inherit)=yes.
1554 * automixmon now supports additional channel variables from automon including:
1555 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
1556 and TOUCH_MIXMONITOR_MESSAGE_STOP
1558 * A new general features.conf option 'recordingfailsound' has been added which
1559 allowssetting a failure sound for a user tries to invoke a recording feature
1560 such as automon or automixmon and it fails.
1562 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
1563 features.c for atxferdropcall=no to work properly. This option now just
1568 * Added log rotation strategy 'none'. If set, no log rotation strategy will
1569 be used. Given that this can cause the Asterisk log files to grow quickly,
1570 this option should only be used if an external mechanism for log management
1575 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
1576 will store the path information for that peer when it registers. Realtime
1577 tables can also use the 'supportpath' field to enable Path header support.
1579 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
1580 objectIdentifier. This maps to the supportpath option in sip.conf.
1584 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
1585 provides modules a useful abstraction on top of the many storage mechanisms
1586 in Asterisk, including the Asterisk Database, static configuration files,
1587 static Realtime, and dynamic Realtime. It also provides a caching service.
1588 Users can configure a hierarchy of data storage layers for specific modules
1591 * All future modules which utilize Sorcery for object persistence must have a
1592 column named "id" within their schema when using the Sorcery realtime module.
1593 This column must be able to contain a string of up to 128 characters in length.
1595 Security Events Framework
1597 * Security Event timestamps now use ISO 8601 formatted date/time instead of
1598 the "seconds-microseconds" format that it was using previously.
1602 * The Stasis message bus is a publish/subscribe message bus internal to
1603 Asterisk. Many services in Asterisk are built on the Stasis message bus,
1604 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
1605 Stasis can be configured in stasis.conf. Note that these parameters operate
1606 at a very low level in Asterisk, and generally will not require changes.
1610 * When a channel driver is configured to enable jiterbuffers, they are now
1611 applied unconditionally when a channel joins a bridge. If a jitterbuffer
1612 is already set for that channel when it enters, such as by the JITTERBUFFER
1613 function, then the existing jitterbuffer will be used and the one set by
1614 the channel driver will not be applied.
1618 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
1619 dialplan applications provided by the app_agent_pool module. Agents are
1620 connected with callers using the new AgentRequest dialplan application.
1621 The Agents:<agent-id> device state is available to monitor the status of an
1622 agent. See agents.conf.sample for valid configuration options.
1624 * The updatecdr option has been removed. Altering the names of channels on a
1625 CDR is not supported - the name of the channel is the name of the channel,
1626 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
1627 has also been removed, for the same reason.
1629 * The endcall and enddtmf configuration options are removed. Use the
1630 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
1631 channel before calling AgentLogin.
1635 * chan_bridge has been removed. Its functionality has been incorporated
1636 directly into the ConfBridge application itself.
1640 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
1641 of the specified span and its B-channels. Note that this command should
1642 only be used if you understand the risks it entails.
1644 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
1645 A range of channels can be specified to be destroyed. Note that this command
1646 should only be used if you understand the risks it entails.
1648 * Added the CLI command 'dahdi create channels'. A range of channels can be
1649 specified to be created, or the keyword 'new' can be used to add channels
1652 * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
1653 the exact configured mailbox name. For app_voicemail mailboxes this is
1656 * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
1660 * IPv6 support has been added. We are now able to bind to and
1661 communicate using IPv6 addresses.
1665 * The /b option has been removed.
1667 * chan_local moved into the system core and is no longer a loadable module.
1671 * Added general support for busy detection.
1673 * Added ECAM command support for Sony Ericsson phones.
1677 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
1678 SIP stack. A collection of resource modules provides the bulk of the SIP
1679 functionality. For more information on the new SIP channel driver, see
1680 https://wiki.asterisk.org/wiki/x/JYGLAQ
1684 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
1685 using the 'supportpath' setting, either on a global basis or on a peer basis.
1686 This setting enables Asterisk to route outgoing out-of-dialog requests via a
1687 set of proxies by using a pre-loaded route-set defined by the Path headers in
1688 the REGISTER request. See Realtime updates for more configuration information.
1690 * The SIP_CODEC family of variables may now specify more than one codec. Each
1691 codec must be separated by a comma. The first codec specified is the
1692 preferred codec for the offer. This allows a dialplan writer to specify both
1693 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
1695 * The 'callevents' parameter has been removed. Hold AMI events are now raised
1696 in the core, and can be filtered out using the 'eventfilter' parameter
1699 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
1700 codecs configured for a peer instead of the requested codec.
1702 * The option "register_retry_403" has been added to chan_sip to work around
1703 servers that are known to erroneously send 403 in response to valid
1704 REGISTER requests and allows Asterisk to continue attepmting to connect.
1708 * Added the 'immeddialkey' parameter. If set, when the user presses the
1709 configured key the already entered number will be immediately dialed. This
1710 is useful when the dialplan allows for variable length pattern matching.
1711 Valid options are '*' and '#'.
1713 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
1714 milliseconds) before a call forward is considered to not be answered.
1716 * The 'serviceurl' parameter allows Service URLs to be attached to line
1725 * The password option has been disabled, as the AgentLogin application no
1726 longer provides authentication.
1730 * Due to changes in the Asterisk core, this function is no longer needed to
1731 preserve a MixMonitor on a channel during transfer operations and dialplan
1732 execution. It is effectively obsolete.
1736 * The 'amaflags' and 'accountcode' attributes for the CDR function are
1737 deprecated. Use the CHANNEL function instead to access these attributes.
1739 * The 'l' option has been removed. When reading a CDR attribute, the most
1740 recent record is always used. When writing a CDR attribute, all non-finalized
1743 * The 'r' option has been removed, for the same reason as the 'l' option.
1745 * The 's' option has been removed, as LOCKED semantics no longer exist in the
1750 * A new function CDR_PROP has been added. This function lets you set properties
1751 on a channel's active CDRs. This function is write-only. Properties accept
1752 boolean values to set/clear them on the channel's CDRs. Valid properties
1754 - 'party_a' - make this channel the preferred Party A in any CDR between two
1755 channels. If two channels have this property set, the creation time of the
1756 channel is used to determine who is Party A. Note that dialed channels are
1757 never Party A in a CDR.
1758 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
1759 application when set to True, and analogous to the 'e' option in ResetCDR
1764 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
1765 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
1766 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
1769 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
1770 string, i.e., [[context],extension],priority. If set on a channel, if a
1771 channel leaves a bridge but is not hung up it will resume dialplan execution
1776 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
1777 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
1778 The value of this setting is ignored when disabled is used for the argument.
1782 * A new function provided by chan_pjsip, this function can be used in
1783 conjunction with the Dial application to construct a dial string that will
1784 dial all contacts on an Address of Record associated with a chan_pjsip
1789 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
1790 outbound channel prior to dialing.
1794 * Redirecting reasons can now be set to arbitrary strings. This means
1795 that the REDIRECTING dialplan function can be used to set the redirecting
1796 reason to any string. It also allows for custom strings to be read as the
1797 redirecting reason from SIP Diversion headers.
1801 * The SPEECH_ENGINE function now supports read operations. When read from, it
1802 will return the current value of the requested attribute.
1806 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1807 system as mailbox@context. The rest of the system cannot add @default
1808 to mailbox identifiers for app_voicemail that do not specify a context
1809 any longer. It is a mailbox identifier format that should only be
1810 interpreted by app_voicemail.
1816 res_agi (Asterisk Gateway Interface)
1818 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
1820 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
1823 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
1824 will start the playback of the audio at the position specified. It will
1825 also return the final position of the file in 'endpos'.
1827 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
1828 channel variable if the user stopped the file playback or if a remote
1829 entity stopped the playback. If neither stopped the playback, it will
1830 indicate the overall success/failure of the playback. If stopped early,
1831 the final offset of the file will be set in the CPLAYBACKOFFSET channel
1834 * The SAY ALPHA command now accepts an additional parameter to control
1835 whether it specifies the case of uppercase, lowercase, or all letters to
1836 provide functionality similar to SayAlphaCase.
1838 res_ari (Asterisk RESTful Interface) (and others)
1840 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
1841 control telephony primitives in Asterisk by remote client. This includes
1842 channels, bridges, endpoints, media, and other fundamental concepts. Users
1843 of ARI can develop their own communications applications, controlling
1844 multiple channels using an HTTP RESTful interface and receiving JSON events
1845 about the objects via a WebSocket connection. ARI can be configured in
1846 Asterisk via ari.conf. For more information on ARI, see
1847 https://wiki.asterisk.org/wiki/x/0YCLAQ
1851 * Parking has been extracted from the Asterisk core as a loadable module,
1852 res_parking. Configuration for parking is now provided by res_parking.conf.
1853 Configuration through features.conf is no longer supported.
1855 * res_parking uses the configuration framework. If an invalid configuration is
1856 supplied, res_parking will fail to load or fail to reload. Previously,
1857 invalid configurations would generally be accepted, with certain errors
1858 resulting in individually disabled parking lots.
1860 * Parked calls are now placed in bridges. While this is largely an
1861 architectural change, it does have implications on how channels in a parking
1862 lot are viewed. For example, commands that display channels in bridges will
1863 now also display the channels in a parking lot.
1865 * The order of arguments for the new parking applications have been modified.
1866 Timeout and return context/exten/priority are now implemented as options,
1867 while the name of the parking lot is now the first parameter. See the
1868 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
1869 in-depth information as well as syntax.
1871 * Extensions are by default no longer automatically created in the dialplan to
1872 park calls or pickup parked calls. Generation of dialplan extensions can be
1873 enabled using the 'parkext' configuration option.
1875 * ADSI functionality for parking is no longer supported. The 'adsipark'
1876 configuration option has been removed as a result.
1878 * The PARKINGSLOT channel variable has been deprecated in favor of
1879 PARKING_SPACE to match the naming scheme of the new system.
1881 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
1882 channel even when the configuration option 'comebactoorigin' is enabled.
1884 * A new CLI command 'parking show' has been added. This allows a user to
1885 inspect the parking lots that are currently in use.
1886 'parking show <parkinglot>' will also show the parked calls in a specific
1889 * The CLI command 'parkedcalls' is now deprecated in favor of
1890 'parking show <parkinglot>'.
1892 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
1893 can be used to get a list of parked calls for a specific parking lot.
1895 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
1896 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
1897 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
1898 longer a required argument.
1900 * The ParkAndAnnounce application is now provided through res_parking instead
1901 of through the separate app_parkandannounce module.
1903 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
1904 by default. Instead, it will follow the timeout rules of the parking lot. The
1905 old behavior can be reproduced by using the 'c' option.
1907 * Dynamic parking lots will now fail to be created under the following
1909 - if the parking lot specified by PARKINGDYNAMIC does not exist
1910 - if they require exclusive park and parkedcall extensions which overlap
1911 with existing parking lots.
1913 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
1914 currently contain no calls. Dynamic parking lots containing parked calls
1915 will persist through the reloads without alteration.
1917 * If 'parkext_exclusive' is set for a parking lot and that extension is
1918 already in use when that parking lot tries to register it, this is now
1919 considered a parking system configuration error. Configurations which do
1920 this will be rejected.
1922 * Added channel variable PARKER_FLAT. This contains the name of the extension
1923 that would be used if 'comebacktoorigin' is enabled. This can be useful when
1924 comebacktoorigin is disabled, but the dialplan or an external control
1925 mechanism wants to use the extension in the park-dial context that was
1926 generated to re-dial the parker on timeout.
1928 res_pjsip (and many others)
1930 * A large number of resource modules make up the SIP stack based on pjsip.
1931 The chan_pjsip channel driver users these resource modules to provide
1932 various SIP functionality in Asterisk. The majority of configuration for
1933 these modules is performed in pjsip.conf. Other modules may use their
1934 own configuration files.
1936 * Added 'set_var' option for an endpoint. For each variable specified that
1937 variable gets set upon creation of a channel involving the endpoint.
1941 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
1942 them, an Asterisk-specific version of PJSIP needs to be installed.
1943 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
1945 res_statsd/res_chan_stats
1947 * A new resource module, res_statsd, has been added, which acts as a statsd
1948 client. This module allows Asterisk to publish statistics to a statsd
1949 server. In conjunction with res_chan_stats, it will publish statistics about
1950 channels to the statsd server. It can be configured via res_statsd.conf.
1954 * Device state for XMPP buddies is now available using the following format:
1955 XMPP/<client name>/<buddy address>
1956 If any resource is available the device state is considered to be not in use.
1957 If no resources exist or all are unavailable the device state is considered
1964 Realtime/Database Scripts
1966 * Asterisk previously included example db schemas in the contrib/realtime/
1967 directory of the source tree. This has been replaced by a set of database
1968 migrations using the Alembic framework. This allows you to use alembic to
1969 initialize the database for you. It will also serve as a database migration
1970 tool when upgrading Asterisk in the future.
1972 See contrib/ast-db-manage/README.md for more details.
1976 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
1977 This python script will convert an existing sip.conf file to a
1978 pjsip.conf file, for use with the chan_pjsip channel driver. This script
1979 is meant to be an aid in converting an existing chan_sip configuration to
1980 a chan_pjsip configuration, but it is expected that configuration beyond
1981 what the script provides will be needed.
1983 ------------------------------------------------------------------------------
1984 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
1985 ------------------------------------------------------------------------------
1989 * The Asterisk build system will now build and install a shared library
1990 (libasteriskssl.so) used to wrap various initialization and shutdown functions
1991 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
1992 that Asterisk can ensure that these functions do *not* get called by any
1993 modules that are loaded into Asterisk, since they should only be called once
1994 in any single process. If desired, this feature can be disabled by supplying
1995 the "--disable-asteriskssl" option to the configure script.
1997 * A new make target, 'full', has been added to the Makefile. This performs
1998 the same compilation actions as make all, but will also scan the entirety of
1999 each source file for documentation. This option is needed to generate AMI
2000 event documentation. Note that your system must have Python in order for
2001 this make target to succeed.
2003 * The optimization portion of the build system has been reworked to avoid
2004 broken builds on certain architectures. All architecture-specific
2005 optimization has been removed in favor of using -march=native to allow gcc
2006 to detect the environment in which it is running when possible. This can
2007 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
2009 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
2010 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
2012 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
2013 previously parsed the header file to obtain the version of Asterisk, you
2014 will now have to go through Asterisk to get the version information.
2022 * Added 'F()' option. Similar to the dial option, this can be supplied with
2023 arguments indicating where the callee should go after the caller is hung up,
2024 or without options specified, the priority after the Queue will be used.
2029 * Added menu action admin_toggle_mute_participants. This will mute / unmute
2030 all non-admin participants on a conference. The confbridge configuration
2031 file also allows for the default sounds played to all conference users when
2032 this occurs to be overriden using sound_participants_unmuted and
2033 sound_participants_muted.
2035 * Added menu action participant_count. This will playback the number of
2036 current participants in a conference.
2038 * Added announcement configuration option to user profile. If set the sound
2039 file will be played to the user, and only the user, upon joining the
2042 * Added record_file_append option that defaults to "yes", but if set to no
2043 will create a new file between each start/stop recording.
2048 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
2049 channels respectively before the callee channels are called.
2054 * Added support for IPv6.
2056 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
2057 external process will cause the current playlist to be cleared, including
2058 stopping any audio file that is currently playing. This is useful when you
2059 want to interrupt audio playback only when specific DTMF is entered by the
2065 * A new option, 'I' has been added to app_followme. By setting this option,
2066 Asterisk will not update the caller with connected line changes when they
2067 occur. This is similar to app_dial and app_queue.
2069 * The 'N' option is now ignored if the call is already answered.
2071 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
2072 and caller channels respectively before the callee channels are called.
2074 * The winning FollowMe outgoing call is now put on hold if the caller put it on
2080 * MixMonitor hooks now have IDs associated with them which can be used to
2081 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
2082 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
2083 now accepts that ID as an argument.
2085 * Added 'm' option, which stores a copy of the recording as a voicemail in the
2086 indicated mailboxes.
2091 * The connect action in app_mysql now allows you to specify a port number to
2092 connect to. This is useful if you run a MySQL server on a non-standard
2098 * Increased the default number of allowed destinations from 5 to 12.
2103 * The app_page application now no longer depends on DAHDI or app_meetme. It
2104 has been re-architected to use app_confbridge internally.
2109 * Added queue options autopausebusy and autopauseunavail for automatically
2110 pausing a queue member when their device reports busy or congestion.
2112 * The 'ignorebusy' option for queue members has been deprecated in favor of
2113 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
2114 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
2115 per interface basis. Individual ringinuse values can now be set in
2116 queues.conf via an argument to member definitions. Lastly, the queue
2117 'ringinuse' setting now only determines defaults for the per member
2118 'ringinuse' setting and does not override per member settings like it does
2119 in earlier versions.
2121 * Added 'F()' option. Similar to the dial option, this can be supplied with
2122 arguments indicating where the callee should go after the caller is hung up,
2123 or without options specified, the priority after the Queue will be used.
2125 * Added new option log_member_name_as_agent, which will cause the membername to
2126 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
2127 state_interface has been set.
2129 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
2131 * App_queue will now play periodic announcements for the caller that
2132 holds the first position in the queue while waiting for answer.
2136 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
2137 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
2138 changed arguments to SayUnixTime so that every option is truly optional even
2139 when using multiple options (so that j option could be used without having to
2140 manually specify timezone and format) There are other benefits, e.g., format
2141 can now be used without specifying time zone as well.
2146 * Addition of the VM_INFO function - see Function changes.
2148 * The imapserver, imapport, and imapflags configuration options can now be
2149 overriden on a user by user basis.
2151 * When voicemail plays a message's envelope with saycid set to yes, when
2152 reaching the caller id field it will play a recording of a file with the same
2153 base name as the sender's callerid if there is a similarly named file in
2154 <astspooldir>/recordings/callerids/
2156 * Voicemails now contains a unique message identifier "msg_id", which is stored
2157 in the message envelope with the sound files. IMAP backends will now store
2158 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
2159 backends will store the message identifier in a "msg_id" column. See
2160 UPGRADE.txt for more information.
2162 * Added VoiceMailPlayMsg application. This application will play a single
2163 voicemail message from a mailbox. The result of the application, SUCCESS or
2164 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
2169 * Hangup handlers can be attached to channels using the CHANNEL() function.
2170 Hangup handlers will run when the channel is hung up similar to the h
2171 extension. The hangup_handler_push option will push a GoSub compatible
2172 location in the dialplan onto the channel's hangup handler stack. The
2173 hangup_handler_pop option will remove the last added location, and optionally
2174 replace it with a new GoSub compatible location. The hangup_handler_wipe
2175 option will remove all locations on the stack, and optionally add a new
2178 * The expression parser now recognizes the ABS() absolute value function,
2179 which will convert negative floating point values to positive values.
2181 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
2182 control of faxdetect.
2184 * Addition of the VM_INFO function that can be used to retrieve voicemail
2185 user information, such as the email address and full name.
2186 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
2189 * The REDIRECTING function now supports the redirecting original party id
2192 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
2193 lets you set some of the configuration options from the [general] section
2194 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
2195 the key sequence used to activate built-in features, such as blindxfer,
2196 and automon. See the built-in documentation for details.
2198 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
2199 instead of simply the uri. This is the format that MessageSend() can use
2200 in the from parameter for outgoing SIP messages.
2202 * Added the PRESENCE_STATE function. This allows retrieving presence state
2203 information from any presence state provider. It also allows setting
2204 presence state information from a CustomPresence presence state provider.
2205 See AMI/CLI changes for related commands.
2207 * Added the AMI_CLIENT function to make manager account attributes available
2208 to the dialplan. It currently supports returning the current number of
2209 active sessions for a given account.
2211 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
2212 and the REDIRECTING functions.
2220 * Added a manager event "LocalBridge" for local channel call bridges between
2221 the two pseudo-channels created.
2226 * Added dialtone_detect option for analog ports to disconnect incoming
2227 calls when dialtone is detected.
2229 * Added option colp_send to send ISDN connected line information. Allowed
2230 settings are block, to not send any connected line information; connect, to
2231 send connected line information on initial connect; and update, to send
2232 information on any update during a call. Default is update.
2234 * Add options namedcallgroup and namedpickupgroup to support installations
2235 where a higher number of groups (>64) is required.
2237 * Added support to use private party ID information with PRI calls.
2242 * A new channel driver named chan_motif has been added which provides support for
2243 Google Talk and Jingle in a single channel driver. This new channel driver includes
2244 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
2245 hold, unhold, and ringing notification. It is also compliant with the current Jingle
2246 specification, current Google Jingle specification, and the original Google Talk
2252 * Added NAT support for RTP. Setting in config is 'nat', which can be set
2253 globally and overriden on a peer by peer basis.
2255 * Direct media functionality has been added. Options in config are:
2256 directmedia (directrtp) and directrtpsetup (earlydirect)
2258 * ChannelUpdate events now contain a CallRef header.
2263 * Asterisk will no longer substitute CID number for CID name in the display
2264 name field if CID number exists without a CID name. This change improves
2265 compatibility with certain device features such as Avaya IP500's directory
2268 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
2269 created using that setting to not be removed during SIP reload.
2271 * Added settings recordonfeature and recordofffeature. When receiving an INFO
2272 request with a "Record:" header, this will turn the requested feature on/off.
2273 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
2274 dynamic features must be enabled and configured properly on the requesting
2275 channel for this to function properly.
2277 * Add support to realtime for the 'callbackextension' option.
2279 * When multiple peers exist with the same address, but differing
2280 callbackextension options, incoming requests that are matched by address
2281 will be matched to the peer with the matching callbackextension if it is
2284 * Two new NAT options, auto_force_rport and auto_comedia, have been added
2285 which set the force_rport and comedia options automatically if Asterisk
2286 detects that an incoming SIP request crossed a NAT after being sent by
2287 the remote endpoint.
2289 * The default global nat setting in sip.conf has been changed from force_rport
2290 to auto_force_rport.
2292 * NAT settings are now a combinable list of options. The equivalent of the
2293 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
2295 * Adds an option send_diversion which can be disabled to prevent
2296 diversion headers from automatically being added to INVITE requests.
2298 * Add support for lightweight NAT keepalive. If enabled a blank packet will
2299 be sent to the remote host at a given interval to keep the NAT mapping open.
2300 This can be enabled using the keepalive configuration option.
2302 * Add option 'tonezone' to specify country code for indications. This option
2303 can be set both globally and overridden for specific peers.
2305 * The SIP Security Events Framework now supports IPv6.
2307 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
2308 between multiple user agents. When set, for directmedia reinvites,
2309 Asterisk will not send an immediate reinvite on an incoming call leg. This
2310 option is useful when peered with another SIP user agent that is known to
2311 send immediate direct media reinvites upon call establishment.
2313 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
2316 * Add options subminexpiry and submaxexpiry to set limits of subscription
2317 timer independently from registration timer settings. The setting of the
2318 registration timer limits still is done by options minexpiry, maxexpiry
2319 and defaultexpiry. For backwards compatibility the setting of minexpiry
2320 and maxexpiry also is used to configure the subscription timer limits if
2321 subminexpiry and submaxexpiry are not set in sip.conf.
2323 * Set registration timer limits to default values when reloading sip
2324 configuration and values are not set by configuration.
2326 * Add options namedcallgroup and namedpickupgroup to support installations
2327 where a higher number of groups (>64) is required.
2329 * When a MESSAGE request is received, the address the request was received from
2330 is now saved in the SIP_RECVADDR variable.
2332 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
2333 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
2334 the ANI2/OLI information is set on the channel, which can be retrieved using
2335 the CALLERID function.
2337 * Peers can now be configured to support negotiation of ICE candidates using
2338 the setting icesupport. See res_rtp_asterisk changes for more information.
2340 * Added support for format attribute negotiation. See the Codecs changes for
2343 * Extra headers specified with SIPAddHeader are sent with the REFER message
2344 when using Transfer application. See refer_addheaders in sip.conf.sample.
2346 * Added support to use private party ID information with calls.
2348 * Adds an option discard_remote_hold_retrieval that when set stops telling
2349 the peer to start music on hold.
2354 * Added skinny version 17 protocol support.
2358 --------------------
2359 * Added option 'dtmf_duration' allowing playback time of DTMF tones to be set
2361 * Modified option 'date_format' to allow options to display date in 31Jan and Jan31
2362 formats as options 0 and 1. The previous options 0 and 1 now map to options 2 and 3
2363 as per the UNISTIM protocol.
2365 * Fixed issues with dialtone not matching indications.conf and mute stopping rx
2366 as well as tx. Also fixed issue with call "Timer" displaying as French "Dur\E9e"
2368 * Added ability to use multiple lines for a single phone. This allows multiple
2369 calls to occur on a single phone, using callwaiting and switching between calls.
2371 * Added option 'sharpdial' allowing end dialing by pressing # key
2373 * Added option 'interdigit_timer' to control phone dial timeout
2375 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
2377 * Added global 'debug' option, that enables debug in channel driver
2379 * Added ability to translate on-screen menu in multiple languages. Tested on
2380 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
2381 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
2384 * In addition to English added French and Russian languages for on-screen menus
2386 * Reworked dialing number input: added dialing by timeout, immediate dial on
2387 on dialplan compare, phone number length now not limited by screen size
2389 * Added ability to pickup a call using features.conf defined value and
2395 * Add options namedcallgroup and namedpickupgroup to support installations
2396 where a higher number of groups (>64) is required.
2398 * Added support to use private party ID information with calls.
2403 * The minimum DTMF duration can now be configured in asterisk.conf
2404 as "mindtmfduration". The default value is (as before) set to 80 ms.
2405 (previously it was only available in source code)
2407 * Named ACLs can now be specified in acl.conf and used in configurations that
2408 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
2409 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
2410 working ACL. In addition, some CLI commands have been added to provide
2411 show information and allow for module reloading - see CLI Changes.
2413 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
2414 items (separated by commas), and items in the rule can be negated by prefixing
2415 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
2416 longer necessray to control the order that the 'permit' and 'deny' columns are
2417 returned from queries.
2419 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
2420 be used within the dynamic weight attribute when specifying a mapping.
2422 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
2423 header, instead of putting the user defined event name there. When enabled
2424 the UserDefType header is added for user defined events. This feature is
2425 enabled with the setting show_user_defined.
2427 * Macro has been deprecated in favor of GoSub. For redirecting and connected
2428 line purposes use the following variables instead of their macro equivalents:
2429 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
2430 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
2431 cc_callback_macro in channel configurations.
2433 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
2436 * Call files now support the "early_media" option to connect with an outgoing
2437 extension when early media is received.
2439 * Added support to use private party ID information with calls.
2444 * A new channel variable, AGIEXITONHANGUP, has been added which allows
2445 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
2446 AGI application would exit immediately after a channel hangup is detected.
2448 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
2449 are resolved and each address is attempted in turn until one succeeds or
2453 AMI (Asterisk Manager Interface)
2455 * The originate action now has an option "EarlyMedia" that enables the
2456 call to bridge when we get early media in the call. Previously,
2457 early media was disregarded always when originating calls using AMI.
2459 * Added setvar= option to manager accounts (much like sip.conf)
2461 * Originate now generates an error response if the extension given is not found
2464 * MixMonitor will now show IDs associated with the mixmonitor upon creating
2465 them if the i(variable) option is used. StopMixMonitor will accept
2466 MixMonitorID as an option to close specific MixMonitors.
2468 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
2469 updated to include information about peers configured with
2470 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
2471 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
2472 returned if auto_force_rport is not enabled.
2474 * Added SIPpeerstatus manager command which will generate PeerStatus events
2475 similar to the existing PeerStatus events found in chan_sip on demand.
2477 * Hangup now can take a regular expression as the Channel option. If you want
2478 to hangup multiple channels, use /regex/ as the Channel option. Existing
2479 behavior to hanging up a single channel is unchanged, but if you pass a regex,
2480 the manager will send you a list of channels back that were hung up.
2482 * Support for IPv6 addresses has been added.
2484 * AMI Events can now be documented in the Asterisk source. Note that AMI event
2485 documentation is only generated when Asterisk is compiled using 'make full'.
2486 See the CLI section for commands to display AMI event information.
2488 * The AMI Hangup event now includes the AccountCode header so you can easily
2489 correlate with AMI Newchannel events.
2491 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
2492 the StateInterface of the queue member.
2494 * Added AMI event SessionTimeout in the Call category that is issued when a
2495 call is terminated due to either RTP stream inactivity or SIP session timer
2498 * CEL events can now contain a user defined header UserDefType. See core
2499 changes for more information.
2501 * OOH323 ChannelUpdate events now contain a CallRef header.
2503 * Added PresenceState command. This command will report the presence state for
2504 the given presence provider.
2506 * Added Parkinglots command. This will list all parking lots as a series of
2507 AMI Parkinglot events.
2509 * Added MessageSend command. This behaves in the same manner as the
2510 MessageSend application, and is a technolgoy agnostic mechanism to send out
2511 of call text messages.
2513 * Added "message" class authorization. This grants an account permission to
2514 send out of call messages. Write-only.
2519 * The "dialplan add include" command has been modified to create context a context
2520 if one does not already exist. For instance, "dialplan add include foo into bar"
2521 will create context "bar" if it does not already exist.
2523 * A "dialplan remove context" command has been added to remove a context from
2526 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
2527 filenames of all running mixmonitors on a channel.
2529 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
2530 numeric instead of 0, 1, or 2.
2532 * "stun show status" will show a table describing how the STUN client is
2535 * "acl show [named acl]" will show information regarding a Named ACL. The
2536 acl module can be reloaded with "reload acl".
2538 * Added CLI command to display AMI event information - "manager show events",
2539 which shows a list of all known and documented AMI events, and "manager show
2540 event [event name]", which shows detail information about a specific AMI
2543 * The result of the CLI command "queue show" now includes the state interface
2544 information of the queue member.
2546 * The command "core set verbose" will now set a separate level of logging for
2547 each remote console without affecting any other console.
2549 * Added command "cdr show pgsql status" to check connection status
2551 * "sip show channel" will now display the complete route set.
2553 * Added "presencestate list" command. This command will list all custom
2554 presence states that have been set by using the PRESENCE_STATE dialplan
2557 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
2558 command. This changes a custom presence to a new state.
2563 * Codec lists may now be modified by the '!' character, to allow succinct
2564 specification of a list of codecs allowed and disallowed, without the
2565 requirement to use two different keywords. For example, to specify all
2566 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
2568 * Add support for parsing SDP attributes, generating SDP attributes, and
2569 passing it through. This support includes codecs such as H.263, H.264, SILK,
2570 and CELT. You are able to set up a call and have attribute information pass.
2571 This should help considerably with video calls.
2573 * The iLBC codec can now use a system-provided iLBC library if one is installed,
2574 just like the GSM codec.
2578 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
2579 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
2583 * Asterisk version and build information is now logged at the beginning of a
2586 * Threads belonging to a particular call are now linked with callids which get
2587 added to any log messages produced by those threads. Log messages can now be
2588 easily identified as involved with a certain call by looking at their call id.
2589 Call ids may also be attached to log messages for just about any case where
2590 it can be determined to be related to a particular call.
2592 * Each logging destination and console now have an independent notion of the
2593 current verbosity level. Logger.conf now allows an optional argument to
2594 the 'verbose' specifier, indicating the level of verbosity sent to that
2595 particular logging destination. Additionally, remote consoles now each
2596 have their own verbosity level. The command 'core set verbose' will now set
2597 a separate level for each remote console without affecting any other
2603 * Added 'announcement' option which will play at the start of MOH and between
2604 songs in modes of MOH that can detect transitions between songs (eg.
2610 * New per parking lot options: comebackcontext and comebackdialtime. See
2611 configs/features.conf.sample for more details.
2613 * Channel variable PARKER is now set when comebacktoorigin is disabled in
2616 * Channel variable PARKEDCALL is now set with the name of the parking lot
2617 when a timeout occurs.
2623 CDR Postgresql Driver
2625 * Added command "cdr show pgsql status" to check connection status
2628 CDR Adaptive ODBC Driver
2630 * Added schema option for databases that support specifying a schema.
2638 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
2639 CALENDAR_WRITE has completed successfully.
2644 * A new option, 'probation' has been added to rtp.conf
2645 RTP in strictrtp mode can now require more than 1 packet to exit learning
2646 mode with a new source (and by default requires 4). The probation option
2647 allows the user to change the required number of packets in sequence to any
2648 desired value. Use a value of 1 to essentially restore the old behavior.
2649 Also, with strictrtp on, Asterisk will now drop all packets until learning
2650 mode has successfully exited. These changes are based on how pjmedia handles
2651 media sources and source changes.
2653 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
2654 enabled or disabled using the icesupport setting. A variety of other
2655 settings have been introduced to configure STUN/TURN connections.
2660 * A new module, res_corosync, has been introduced. This module uses the
2661 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
2662 of Asterisk servers to both Message Waiting Indication (MWI) and/or
2663 Device State (presence) information. This module is very similar to, and
2664 is a replacement for the res_ais module that was in previous releases of
2670 * This module adds a cleaned up, drop-in replacement for res_jabber called
2671 res_xmpp. This provides the same externally facing functionality but is
2672 implemented differently internally. res_jabber has been deprecated in favor
2673 of res_xmpp; please see the UPGRADE.txt file for more information.
2678 * The safe_asterisk script has been updated to allow several of its parameters
2679 to be set from environment variables. This also enables a custom run
2680 directory of Asterisk to be specified, instead of defaulting to /tmp.
2682 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
2683 its value to determine the directory to assume is the top-level directory of
2684 the source tree. If the variable is not set, it defaults to the current
2685 behavior and uses the current working directory.
2687 ------------------------------------------------------------------------------
2688 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
2689 ------------------------------------------------------------------------------
2693 * Asterisk now has protocol independent support for processing text messages
2694 outside of a call. Messages are routed through the Asterisk dialplan.
2695 SIP MESSAGE and XMPP are currently supported. There are options in
2696 jabber.conf and sip.conf to allow enabling these features.
2697 -> jabber.conf: see the "sendtodialplan" and "context" options.
2698 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
2699 and "outofcall_message_context" options.
2700 The MESSAGE() dialplan function and MessageSend() application have been
2701 added to go along with this functionality. More detailed usage information
2702 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
2703 * If real-time text support (T.140) is negotiated, it will be preferred for
2704 sending text via the SendText application. For example, via SIP, messages
2705 that were once sent via the SIP MESSAGE request would be sent via RTP if
2706 T.140 text is negotiated for a call.
2710 * parkedmusicclass can now be set for non-default parking lots.
2712 Asterisk Manager Interface
2713 --------------------------
2714 * PeerStatus now includes Address and Port.
2715 * Added Hold events for when the remote party puts the call on and off hold
2716 for chan_dahdi ISDN channels.
2717 * Added new action MeetmeListRooms to list active conferences (shows same
2718 data as "meetme list" at the CLI).
2719 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
2720 Description field that is set by 'description' in the channel configuration
2722 * Added Uniqueid header to UserEvent.
2723 * Added new action FilterAdd to control event filters for the current session.
2724 This requires the system permission and uses the same filter syntax as
2725 filters that can be defined in manager.conf
2726 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
2727 versions had some instances of the event converted, but others were left
2728 as-is. All Unlink events should now be converted to Bridge events. The AMI
2729 protocol version number was incremented to 1.2 as a result of this change.
2731 Asterisk HTTP Server
2732 --------------------------
2733 * The HTTP Server can bind to IPv6 addresses.
2736 --------------------------
2737 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
2738 with busydetect. usage example: busypattern=200,200,200,600
2741 --------------------------
2742 * New 'gtalk show settings' command showing the current settings loaded from
2744 * The 'logger reload' command now supports an optional argument, specifying an
2745 alternate configuration file to use.
2746 * 'dialplan add extension' command will now automatically create a context if
2747 the specified context does not exist with a message indicated it did so.
2748 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
2749 Description field which can be populated with 'description' in the channel
2750 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
2753 --------------------------
2754 * The filter option in cdr_adaptive_odbc now supports negating the argument,
2755 thus allowing records which do NOT match the specified filter.
2756 * Added ability to log CONGESTION calls to CDR
2759 --------------------------
2760 * Ability to define custom SILK formats in codecs.conf.
2761 * Addition of speex32 audio format with translation.
2762 * CELT codec pass-through support and ability to define
2763 custom CELT formats in codecs.conf.
2764 * Ability to read raw signed linear files with sample rates
2765 ranging from 8khz - 192khz. The new file extensions introduced
2766 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
2767 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
2768 Skinny, H.323, etc) can still only support the following codecs:
2769 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
2770 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
2771 Video: h261, h263, h263p, h264, mpeg4
2776 --------------------------
2777 * New highly optimized and customizable ConfBridge application capable of
2778 mixing audio at sample rates ranging from 8khz-96khz.
2779 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
2780 and bridge profiles on a channel.
2781 * CONFBRIDGE_INFO dialplan function capable of retrieving information
2782 about a conference such as locked status and number of parties, admins,
2784 * Addition of video_mode option in confbridge.conf for adding video support
2785 into a bridge profile.
2786 * Addition of the follow_talker video_mode in confbridge.conf. This video
2787 mode dynamically switches the video feed to always display the loudest talker
2788 supplying video in the conference.
2792 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
2793 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
2794 variables from asterisk.conf.
2798 * Addition of the JITTERBUFFER dialplan function. This function allows
2799 for jitterbuffering to occur on the read side of a channel. By using
2800 this function conference applications such as ConfBridge and MeetMe can
2801 have the rx streams jitterbuffered before conference mixing occurs.
2802 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
2804 * Added STRREPLACE function. This function let's the user search a variable
2805 for a given string to replace with another string as many times as the
2806 user specifies or just throughout the whole string.
2807 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
2808 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
2809 * Added extensions to chan_ooh323 in function CHANNEL()
2811 libpri channel driver (chan_dahdi) DAHDI changes
2812 --------------------------
2813 * Added moh_signaling option to specify what to do when the channel's bridged
2814 peer puts the ISDN channel on hold.
2815 * Added display_send and display_receive options to control how the display ie
2816 is handled. To send display text from the dialplan use the SendText()
2817 application when the option is enabled.
2818 * Added mcid_send option to allow sending a MCID request on a span.
2821 --------------------------
2822 * Added setvar option to calendar.conf to allow setting channel variables on
2823 notification channels.
2824 * Added "calendar show types" CLI command to list registered calendar
2828 --------------------------
2829 * Added two new options, r and t with file name arguments to record
2830 single direction (unmixed) audio recording separate from the bidirectional
2831 (mixed) recording. The mixed file name argument is optional now as long
2832 as at least one recording option is used.
2835 --------------------------
2836 * Added a new option, l, which will disable local call optimization for
2837 channels involved with the FollowMe thread. Use this option to improve
2838 compatability for a FollowMe call with certain dialplan apps, options, and
2842 --------------------------
2843 * Added option "k" that will automatically close the conference when there's
2844 only one person left when a user exits the conference.
2847 --------------------------
2848 * cel_pgsql now supports the 'extra' column for data added using the
2849 CELGenUserEvent() application.
2852 --------------------------
2853 * Support for defining hints has been added to pbx_lua. See the 'hints' table
2854 in the sample extensions.lua file for syntax details.
2855 * Applications that perform jumps in the dialplan such as Goto will now
2856 execute properly. When pbx_lua detects that the context, extension, or
2857 priority we are executing on has changed it will immediately return control
2858 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
2859 the priority after the currently executing priority.
2860 * An autoservice is now started by default for pbx_lua channels. It can be
2861 stopped and restarted using the autoservice_stop() and autoservice_start()
2865 --------------------------
2866 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
2867 into a FAXStatus event with an 'Operation' header that will be either
2868 'send', 'receive', and 'gateway'.
2869 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
2870 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
2871 feature will handle converting a fax call between an audio T.30 fax terminal
2872 and an IFP T.38 fax terminal.
2876 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
2877 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
2878 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
2882 * Added general option negative_penalty_invalid default off. when set
2883 members are seen as invalid/logged out when there penalty is negative.
2884 for realtime members when set remove from queue will set penalty to -1.
2885 * Added queue option autopausedelay when autopause is enabled it will be
2886 delayed for this number of seconds since last successful call if there
2887 was no prior call the agent will be autopaused immediately.
2888 * Added member option ignorebusy this when set and ringinuse is not
2889 will allow per member control of multiple calls as ringinuse does for
2894 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
2896 * Added 'k' option to MeetMe to automatically kill the conference when there's only
2897 one participant left (much like a normal call bridge)
2898 * Added extra argument to Originate to set timeout.
2902 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
2903 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
2904 utility in the UTILS section of menuselect. If an existing astdb is found and no
2905 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
2906 convert an existing astdb to the SQLite3 version automatically at runtime.
2910 * Modules marked as deprecated are no longer marked as building by default. Enabling
2911 these modules is still available via menuselect.
2915 * authdebug is now disabled by default. To enable this functionaility again
2916 set authdebug = yes in iax.conf.
2920 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
2921 releases it was disabled.
2925 * The PBX core previously made a call with a non-existing extension test for
2926 extension s@default and jump there if the extension existed.
2927 This was a bad default behaviour and violated the principle of least surprise.
2928 It has therefore been changed in this release. It may affect some
2929 applications and configurations that rely on this behaviour. Most channel
2930 drivers have avoided this for many releases by testing whether the extension
2931 called exists before starting the PBX and generating a local error.
2932 This behaviour still exists and works as before.
2934 Extension "s" is used when no extension is given in a channel driver,
2935 like immediate answer in DAHDI or calling to a domain with no user part
2938 ------------------------------------------------------------------------------
2939 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
2940 ------------------------------------------------------------------------------
2944 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
2945 now defaults to force_rport. It is very important that phones requiring nat=no be
2946 specifically set as such instead of relying on the default setting. If at all
2947 possible, all devices should have nat settings configured in the general section as
2948 opposed to configuring nat per-device.
2949 * Added preferred_codec_only option in sip.conf. This feature limits the joint
2950 codecs sent in response to an INVITE to the single most preferred codec.
2951 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
2952 to be used for the outgoing call. It must be one of the codecs configured
2954 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
2955 to be used for holding a private key. If tlsprivatekey is not specified,
2956 tlscertfile is searched for both public and private key.
2957 * Added tlsclientmethod option to sip.conf. This allows the protocol for
2958 outbound client connections to be specified.
2959 * The sendrpid parameter has been expanded to include the options
2960 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
2961 header to be sent (equivalent to setting sendrpid=yes) and setting
2962 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
2963 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
2964 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
2965 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
2966 will accept the SDP even if the SDP version number is not properly incremented,
2967 but will generate a warning in the log indicating that the SIP peer that sent
2968 the SDP should have the 'ignoresdpversion' option set.
2969 * The 'nat' option has now been been changed to have yes, no, force_rport, and
2970 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
2971 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
2972 remote side requests it and disables symmetric RTP support. Setting it to
2973 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
2974 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
2975 and enables symmetric RTP support.
2976 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
2977 response. This permits the master channel to know how each channel dialled
2978 in a multi-channel setup resolved in an individual way. This carries a
2979 performance penalty and can be disabled in sip.conf using the
2980 'storesipcause' option.
2981 * Added 'externtcpport' and 'externtlsport' options to allow custom port
2982 configuration for the externip and externhost options when tcp or tls is used.
2983 * Added support for message body (stored in content variable) to SIP NOTIFY message
2984 accessible via AMI and CLI.
2985 * Added 'media_address' configuration option which can be used to explicitly specify
2986 the IP address to use in the SDP for media (audio, video, and text) streams.
2987 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
2988 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
2990 * Added 'use_q850_reason' configuration option for generating and parsing
2991 if available Reason: Q.850;cause=<cause code> header. It is implemented
2992 in some gateways for better passing PRI/SS7 cause codes via SIP.
2993 * When dialing SIP peers, a new component may be added to the end of the dialstring
2994 to indicate that a specific remote IP address or host should be used when dialing
2995 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
2996 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
2997 ability to selectively force bridged channels to also be encrypted is also
2998 implemented. Branching in the dialplan can be done based on whether or not
2999 a channel has secure media and/or signaling.
3000 * Added directmediapermit/directmediadeny to limit which peers can send direct media
3002 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
3003 Charge messages to snom phones.
3004 * Added support for G.719 media streams.
3005 * Added support for 16khz signed linear media streams.
3006 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
3007 RTP has been outfitted with the same abilities.
3008 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
3009 available in device configurations as well as in the dial plan.
3010 * Addition of the 'subscribe_network_change' option for turning on and off
3011 res_stun_monitor module support in chan_sip.
3012 * Addition of the 'auth_options_requests' option for turning on and off
3013 authentication for OPTIONS requests in chan_sip.
3017 * Add #tryinclude statement for config files. This provides the same
3018 functionality as the #include statement however an asterisk module will
3019 still load if the filename does not exist. Using the #include statement
3020 Asterisk will not allow the module to load.
3024 * Added rtsavesysname option into iax.conf to allow the systname to be saved
3025 on realtime updates.
3026 * Added the ability for chan_iax2 to inform the dialplan whether or not
3027 encryption is being used. This interoperates with the SIP SRTP implementation
3028 so that a secure SIP call can be bridged to a secure IAX call when the
3029 dialplan requires bridged channels to be "secure".
3030 * Addition of the 'subscribe_network_change' option for turning on and off
3031 res_stun_monitor module support in chan_iax.
3036 * Added ability to preset channel variables on indicated lines with the setvar
3037 configuration option. Also, clearvars=all resets the list of variables back
3039 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
3040 See configs/res_pktccops.conf for more information.
3042 XMPP Google Talk/Jingle changes
3043 -------------------------------
3044 * Added the externip option to gtalk.conf.
3045 * Added the stunaddr option to gtalk.conf which allows for the automatic
3046 retrieval of the external ip from a stun server.
3050 * Added 'p' option to PickupChan() to allow for picking up channel by the first
3051 match to a partial channel name.
3052 * Added .m3u support for Mp3Player application.
3053 * Added progress option to the app_dial D() option. When progress DTMF is
3054 present, those values are sent immediately upon receiving a PROGRESS message
3055 regardless if the call has been answered or not.
3056 * Added functionality to the app_dial F() option to continue with execution
3057 at the current location when no parameters are provided.
3058 * Added the 'a' option to app_dial to answer the calling channel before any
3059 announcements or macros are executed.
3060 * Modified app_dial to set answertime when the called channel answers even if
3061 the called channel hangs up during playback of an announcement.
3062 * Modified app_dial 'r' option to support an additional parameter to play an
3063 indication tone from indications.conf
3064 * Added c() option to app_chanspy. This option allows custom DTMF to be set
3065 to cycle through the next available channel. By default this is still '*'.
3066 * Added x() option to app_chanspy. This option allows DTMF to be set to
3067 exit the application.
3068 * The Voicemail application has been improved to automatically ignore messages
3069 that only contain silence.
3070 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
3071 associated mailbox(es) to be greetings-only.
3072 * The ChanSpy application now has the 'S' option, which makes the application
3073 automatically exit once it hits a point where no more channels are available
3075 * The ChanSpy application also now has the 'E' option, which spies on a single
3076 channel and exits when that channel hangs up.
3077 * The MeetMe application now turns on the DENOISE() function by default, for
3078 each participant. In our tests, this has significantly decreased background
3079 noise (especially noisy data centers).
3080 * Voicemail now permits storage of secrets in a separate file, located in the
3081 spool directory of each individual user. The control for this is located in
3082 the "passwordlocation" option in voicemail.conf. Please see the sample
3083 configuration for more information.
3084 * The ChanIsAvail application now exposes the returned cause code using a separate
3085 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
3086 * Added 'd' option to app_followme. This option disables the "Please hold"
3088 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
3089 received will terminate recording.
3090 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
3091 Previously the folder could only be set per context, but has now been extended
3092 using the imapfolder option.
3093 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
3094 * Voicemail now allows the pager date format to be specified separately from the
3096 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
3097 to allow joining, leaving, and sending text to group chats.
3098 * MeetMe has a new option 'G' to play an announcement before joining a conference.
3099 * Page has a new option 'A(x)' which will playback an announcement simultaneously
3100 to all paged phones (and optionally excluding the caller's one using the new
3101 option 'n') before the call is bridged.
3102 * The 'f' option to Dial has been augmented to take an optional argument. If no
3103 argument is provided, the 'f' option works as it always has. If an argument is
3104 provided, then the connected party information of all outgoing channels created
3105 during the Dial will be set to the argument passed to the 'f' option.
3106 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
3108 * The OSP lookup application adds in/outbound network ID, optional security,
3109 number portability, QoS reporting, destination IP port, custom info and service
3111 * Added new application VMSayName that will play the recorded name of the voicemail
3112 user if it exists, otherwise will play the mailbox number.
3113 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
3114 retrieve state for a particular bridge, where <name> is the conference name
3115 * app_directory now allows exiting at any time using the operator or pound key.
3116 * Voicemail now supports setting a locale per-mailbox.
3117 * Two new applications are provided for declining counting phrases in multiple
3118 languages. See the application notes for SayCountedNoun and SayCountedAdj for
3120 * Voicemail now runs the externnotify script when pollmailboxes is activated and
3122 * Voicemail now includes rdnis within msgXXXX.txt file.
3123 * ExternalIVR now supports IPv6 addresses.
3124 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
3125 at https://wiki.asterisk.org/wiki/x/oQBB
3126 * ParkedCall and Park can now specify the parking lot to use.
3130 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
3131 over SRV records associated with a specific service. From the CLI, type
3132 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
3133 details on how these may be used.
3134 * PITCH_SHIFT dialplan function added. This function can be used to modify the
3135 pitch of a channel's tx and rx audio streams.
3136 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
3137 setting various connected line and redirecting party information.
3138 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
3139 support ISDN subaddressing.
3140 * The CHANNEL() function now supports the "name" and "checkhangup" options.
3141 * For DAHDI channels, the CHANNEL() dialplan function now allows
3142 the dialplan to request changes in the configuration of the active
3143 echo canceller on the channel (if any), for the current call only.
3146 exten => s,n,Set(CHANNEL(echocan_mode)=off)
3148 The possible values are:
3150 on - normal mode (the echo canceller is actually reinitialized)
3152 fax - FAX/data mode (NLP disabled if possible, otherwise completely
3154 voice - voice mode (returns from FAX mode, reverting the changes that
3155 were made when FAX mode was requested)
3156 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
3157 and setting variables on the channel which created the current channel.
3158 Administrators should take care to avoid naming conflicts, when multiple
3159 channels are dialled at once, especially when used with the Local channel
3160 construct (which all could set variables on the master channel). Usage
3161 of the HASH() dialplan function, with the key set to the name of the slave
3162 channel, is one approach that will avoid conflicts.
3163 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
3165 * func_odbc now allows multiple row results to be retrieved without using
3166 mode=multirow. If rowlimit is set, then additional rows may be retrieved
3167 from the same query by using the name of the function which retrieved the
3168 first row as an argument to ODBC_FETCH().
3169 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
3170 dialplan. This function returns the content of the received message.
3171 * Added REPLACE, which searches a given variable name for a set of characters,
3172 then either replaces them with a single character or deletes them.
3173 * Added PASSTHRU, which literally passes the same argument back as its return
3174 value. The intent is to be able to use a literal string argument to
3175 functions that currently require a variable name as an argument.
3176 * HASH-associated variables now can be inherited across channel creation, by
3177 prefixing the name of the hash at assignment with the appropriate number of
3178 underscores, just like variables.
3179 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
3180 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
3181 whether or not channels that are bridged to the current channel will be
3182 required to have secure signaling and/or media.
3183 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
3184 the current channel has secure signaling and/or media.
3185 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
3186 "no_media_path" option.
3187 Returns "0" if there is a B channel associated with the call.
3188 Returns "1" if no B channel is associated with the call. The call is either
3189 on hold or is a call waiting call.
3190 * Added option to dialplan function CDR(), the 'f' option
3191 allows for high resolution times for billsec and duration fields.
3192 * FILE() now supports line-mode and writing.
3193 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
3194 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
3198 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
3199 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
3200 and is set when a dynamic feature is triggered.
3201 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
3202 to dynamically create a new parking lot matching the value this varible is
3204 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
3205 features.conf that should be the base for dynamic parkinglots.
3206 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
3207 parkinglot should have.
3208 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
3209 parkinglot should have.
3210 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
3215 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
3216 timeout has expired.
3217 * Added 'R' option to app_queue. This option stops moh and indicates ringing
3218 to the caller when an Agent's phone is ringing. This can be used to indicate
3219 to the caller that their call is about to be picked up, which is nice when
3220 one has been on hold for an extened period of time.
3221 * A new config option, penaltymemberslimit, has been added to queues.conf.
3222 When set this option will disregard penalty settings when a queue has too
3224 * A new option, 'I' has been added to both app_queue and app_dial.
3225 By setting this option, Asterisk will not update the caller with
3226 connected line changes or redirecting party changes when they occur.
3227 * A 'relative-periodic-announce' option has been added to queues.conf. When
3228 enabled, this option will cause periodic announce times to be calculated
3229 from the end of announcements rather than from the beginning.
3230 * The autopause option in queues.conf can be passed a new value, "all." The
3231 result is that if a member becomes auto-paused, he will be paused in all
3232 queues for which he is a member, not just the queue that failed to reach
3234 * Added dialplan function QUEUE_EXISTS to check if a queue exists
3235 * The queue logger now allows events to optionally propagate to a file,
3236 even when realtime logging is turned on. Additionally, realtime logging
3237 supports sending the event arguments to 5 individual fields, although it
3238 will fallback to the previous data definition, if the new table layout is
3241 mISDN channel driver (chan_misdn) changes
3242 ----------------------------------------
3243 * Added display_connected parameter to misdn.conf to put a display string
3244 in the CONNECT message containing the connected name and/or number if
3245 the presentation setting permits it.
3246 * Added display_setup parameter to misdn.conf to put a display string
3247 in the SETUP message containing the caller name and/or number if the
3248 presentation setting permits it.
3249 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
3250 indicate the dialplan settings are to be obtained from the asterisk
3252 * Made misdn.conf parameter callerid accept the "name" <number> format
3253 used by the rest of the system.
3254 * Made use the nationalprefix and internationalprefix misdn.conf
3255 parameters to prefix any received number from the ISDN link if that
3256 number has the corresponding Type-Of-Number. NOTE: This includes
3257 comparing the incoming call's dialed number against the MSN list.
3258 * Added the following new parameters: unknownprefix, netspecificprefix,
3259 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
3260 received number from the ISDN link if that number has the corresponding
3262 * Added new dialplan application misdn_command which permits controlling
3263 the CCBS/CCNR functionality.
3264 * Added new dialplan function mISDN_CC which permits retrieval of various
3265 values from an active call completion record.
3266 * For PTP, you should manually send the COLR of the redirected-to party
3267 for an incomming redirected call if the incoming call could experience
3268 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
3269 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
3270 if the REDIRECTING(from-num) is not empty.
3271 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
3272 option on all of the REDIRECTING statements before dialing the
3273 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
3274 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
3275 redirecting-to presentation (COLR) when it becomes available.
3276 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
3279 thirdparty mISDN enhancements
3280 -----------------------------
3281 mISDN has been modified by Digium, Inc. to greatly expand facility message
3283 * Enhanced COLP support for call diversion and transfer.
3284 * CCBS/CCNR support.
3286 The latest modified mISDN v1.1.x based version is available at:
3287 http://svn.digium.com/svn/thirdparty/mISDN/trunk
3288 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
3290 Tagged versions of the modified mISDN code are available under:
3291 http://svn.digium.com/svn/thirdparty/mISDN/tags
3292 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
3294 libpri channel driver (chan_dahdi) DAHDI changes
3295 -------------------------------------------
3296 * The channel variable PRIREDIRECTREASON is now just a status variable
3297 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
3298 to read and alter the reason.
3299 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
3300 redirected-to party for an incomming redirected call if the incoming call
3301 could experience further redirects. Just set the
3302 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
3303 to the COLR. A call has been redirected if the REDIRECTING(count) is not
3305 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
3306 use the inhibit(i) option on all of the REDIRECTING statements before
3307 dialing the redirected-to party. You still have to set the
3308 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
3309 will update the redirecting-to presentation (COLR) when it becomes available.
3310 * Added the ability to ignore calls that are not in a Multiple Subscriber
3311 Number (MSN) list for PTMP CPE interfaces.
3312 * Added dynamic range compression support for dahdi channels. It is
3313 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
3314 * Added support for ISDN calling and called subaddress with partial support
3315 for connected line subaddress.
3316 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
3317 * Added handling of received HOLD/RETRIEVE messages and the optional ability
3318 to transfer a held call on disconnect similar to an analog phone.
3319 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
3320 Will reroute/deflect an outgoing call when receive the message.
3321 Can use the DAHDISendCallreroutingFacility to send the message for the
3323 * Added standard location to add options to chan_dahdi dialing:
3324 Dial(DAHDI/g1[/extension[/options]])
3327 R Reverse charging indication
3328 * Added Reverse Charging Indication (Collect calls) send/receive option.
3329 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
3330 Dial(DAHDI/g1/extension/R)
3331 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
3332 (requires latest LibPRI)
3333 * Added ability to send/receive keypad digits in the SETUP message.
3334 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
3335 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
3336 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
3337 (requires latest LibPRI)
3338 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
3339 to eliminate tromboned calls. A tromboned call goes out an interface and comes
3340 back into the same interface. Tromboned calls happen because of call routing,
3341 call deflection, call forwarding, and call transfer.
3342 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
3343 * Added the ability to support call waiting calls. (The SETUP has no B channel
3345 * Added Malicious Call ID (MCID) event to the AMI call event class.
3346 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
3348 Asterisk Manager Interface
3349 --------------------------
3350 * The Hangup action now accepts a Cause header which may be used to
3351 set the channel's hangup cause.
3352 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
3353 to specify a separate .pem file to hold a private key. By default sslcert
3354 is used to hold both the public and private key.
3355 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
3356 for options containing the 'tls' prefix. For example, 'sslenable' is now
3357 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
3358 across all .conf files. All affected sample.conf files have been modified to
3359 reflect this change. Previous options such as 'sslenable' still work,
3360 but options with the 'tls' prefix are preferred.
3361 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
3362 in a channel. (res_mutestream.so)
3363 * The configuration file manager.conf now supports a channelvars option, which
3364 specifies a list of channel variables to include in each channel-oriented
3366 * The redirect command now has new parameters ExtraContext, ExtraExtension,
3367 and ExtraPriority to allow redirecting the second channel to a different
3368 location than the first.
3369 * Added new event "JabberStatus" in the Jabber module to monitor buddies
3371 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
3372 in a MixMonitor recording.
3373 * The 'iax2 show peers' output is now similar to the expected output of
3375 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
3377 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
3378 AOC-E messages on a channel.
3379 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
3380 conform more closely to similar events.
3381 * Added a new eventfilter option per user to allow whitelisting and blacklisting
3383 * Added optional parkinglot variable for park command.
3384 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
3385 if CallerIDNum and CallerIDName headers are also present.
3387 Channel Event Logging
3388 ---------------------
3389 * A new interface, CEL, is introduced here. CEL logs single events, much like
3390 the AMI, but it differs from the AMI in that it logs to db backends much
3391 like CDR does; is based on the event subsystem introduced by Russell, and
3392 can share in all its benefits; allows multiple backends to operate like CDR;
3393 is specialized to event data that would be of concern to billing sytems,
3394 like CDR. Backends for logging and accounting calls have been produced,
3395 but a new CDR backend is still in development.
3399 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
3400 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
3401 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
3402 * Multiple files and formats can now be specified in cdr_custom.conf.
3403 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
3404 See configs/cdr_syslog.conf.sample for more information.
3405 * A 'sequence' field has been added to CDRs which can be combined with
3406 linkedid or uniqueid to uniquely identify a CDR.
3407 * Handling of billsec and duration field has changed. If your table definition
3408 specifies those fields as float,double or similar they will now be logged with
3409 microsecond accuracy instead of a whole integer.
3411 Calendaring for Asterisk
3412 ------------------------
3413 * A new set of modules were added supporing calendar integration with Asterisk.
3414 Dialplan functions for reading from and writing to calendars are included,
3415 as well as the ability to execute dialplan logic upon calendar event notifications.
3416 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
3417 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
3418 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
3419 2003 support does not support forms-based authentication).
3421 Call Completion Supplementary Services for Asterisk
3422 ---------------------------------------------------
3423 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
3424 DAHDI/ISDN supports call completion for the following switch types:
3425 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
3426 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
3428 Multicast RTP Support
3429 ---------------------
3430 * A new RTP engine and channel driver have been added which supports Multicast RTP.
3431 The channel driver can be used with the Page application to perform multicast RTP
3432 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
3433 Type can be either basic or linksys.
3434 Destination is the IP address and port for the RTP packets.
3435 Control address is specific to the linksys type and is used for sending the control
3436 packets unique to them.
3438 Security Events Framework
3439 -------------------------
3440 * Asterisk has a new C API for reporting security events. The module res_security_log
3441 sends these events to the "security" logger level. Currently, AMI is the only
3442 Asterisk component that reports security events. However, SIP support will be
3443 coming soon. For more information on the security events framework, see the
3444 "Asterisk Security Framework" section of the Asterisk wiki at
3445 https://wiki.asterisk.org/wiki/x/wgBQ
3446 * SIP support was added in Asterisk 10
3447 * This API now supports IPv6 addresses
3451 * A technology independent fax frontend (res_fax) has been added to Asterisk.
3452 * A spandsp based fax backend (res_fax_spandsp) has been added.
3453 * The app_fax module has been deprecated in favor of the res_fax module and
3454 the new res_fax_spandsp backend.
3455 * The SendFAX and ReceiveFAX applications now send their log messages to a
3456 'fax' logger level, instead of to the generic logger levels. To see these
3457 messages, the system's logger.conf file will need to direct the 'fax' logger
3458 level to one or more destinations; the logger.conf.sample file includes an
3459 example of how to do this. Note that if the 'fax' logger level is *not*
3460 directed to at least one destination, log messages generated by these
3461 applications will be lost, and that if the 'fax' logger level is directed to
3462 the console, the 'core set verbose' and 'core set debug' CLI commands will
3463 have no effect on whether the messages appear on the console or not.
3467 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
3468 Now, in order to enable transmitting silence during record the transmit_silence
3469 option should be used. transmit_silence_during_record remains a valid option, but
3470 defaults to the behavior of the transmit_silence option.
3471 * Addition of the Unit Test Framework API for managing registration and execution
3472 of unit tests with the purpose of verifying the operation of C functions.
3473 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
3474 XMPP text messages to the remote JID.
3475 * Modules.conf has a new option - "require" - that marks a module as critical for
3476 the execution of Asterisk.
3477 If one of the required modules fail to load, Asterisk will exit with a return
3479 * An 'X' option has been added to the asterisk application which enables #exec support.
3480 This allows #exec to be used in asterisk.conf.
3481 * jabber.conf supports a new option auth_policy that toggles auto user registration.
3482 * A new lockconfdir option has been added to asterisk.conf to protect the
3483 configuration directory (/etc/asterisk by default) during reloads.
3484 * The parkeddynamic option has been added to features.conf to enable the creation
3485 of dynamic parkinglots.
3486 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
3487 the reportalarms config option.
3488 * chan_dahdi supports dialing configuring and dialing by device file name.
3489 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
3490 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
3491 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
3492 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
3493 Handy for the above name-based syntax as it does not depend on
3494 initialization order.
3495 * The Realtime dialplan switch now caches entries for 1 second. This provides a
3496 significant increase in performance (about 3X) for installations using this switchtype.
3497 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
3498 AIS. For more information, please see the Distributed Device State section of the
3499 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
3500 * The addition of G.719 pass-through support.
3501 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
3502 during device configuration.
3503 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
3504 have less than 3 lines on the LCD.
3505 * Realtime now supports database failover. See the sample extconfig.conf for details.
3506 * The addition of improved translation path building for wideband codecs. Sample
3507 rate changes during translation are now avoided unless absolutely necessary.
3508 * The addition of the res_stun_monitor module for monitoring and reacting to network
3509 changes while behind a NAT.
3510 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
3511 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
3512 These allow support for any Administration. Default is AT&T values.
3516 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
3517 optionally accept a filename, to apply the setting only to the code generated from
3518 that source file when Asterisk was built. However, there are some modules in Asterisk
3519 that are composed of multiple source files, so this did not result in the behavior
3520 that users expected. In this version, 'core set debug' and 'core set verbose'
3521 can optionally accept *module* names instead (with or without the .so extension),
3522 which applies the setting to the entire module specified, regardless of which source
3523 files it was built from.
3524 * New 'manager show settings' command showing the current settings loaded from
3526 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
3527 the channel hangup request to all channels.
3528 * Added a "core reload" CLI command that executes a global reload of Asterisk.
3530 ------------------------------------------------------------------------------
3531 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
3532 ------------------------------------------------------------------------------
3536 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
3537 Snom phones use this for call pickup of extensions that the phone is
3539 * Added support for setting the domain in the URI for caller of an
3540 outbound call by using the SIPFROMDOMAIN channel variable.
3541 * Added a new configuration option "remotesecret" for authentication to
3542 remote services. For backwards compatibility, "secret" still has the
3543 same function as before, but now you can configure both a remote secret and a
3544 local secret for mutual authentication.
3545 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
3546 the sound will be played to the target of an attended transfer
3547 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
3548 finer control over how many peers Asterisk will qualify and the gap between them
3549 when all peers need to be qualified at the same time.
3550 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
3551 (either globally or for a specific peer), chan_sip will treat any SDP data
3552 it receives as new data and update the media stream accordingly. By
3553 default, Asterisk will only modify the media stream if the SDP session
3554 version received is different from the current SDP session version. This
3555 option is required to interoperate with devices that have non-standard SDP
3556 session version implementations (observed with Microsoft OCS). This option
3557 is disabled by default.
3558 * The parsing of register => lines in sip.conf has been modified to allow a port
3559 to be present in the "user" portion. Please see the sip.conf.sample file for more
3561 * Added support for subscribing to MWI on a remote server and making the status available
3562 as a mailbox. Please see the sip.conf.sample file for more information.
3563 * Added a function to remove SIP headers added in the dialplan before the
3564 first INVITE is generated - SIPRemoveHeader()
3565 * Channel variables set with setvar= in a device configuration is now
3566 set both for inbound and outbound calls.
3567 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
3571 * Added immediate option to iax.conf
3572 * Added forceencryption option to iax.conf
3573 * Added Encryption and Trunk status to manager command "iaxpeers"
3577 * The configuration file now holds separate sections for devices and lines.
3578 Please have a look at configs/skinny.conf.sample and change your skinny.conf
3583 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
3584 support for LibOpenR2. http://www.libopenr2.org/
3585 * The UK option waitfordialtone has been added for use with BT analog
3587 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
3588 is used in conjunction with the 'faxdetect' configuration option. When
3589 'faxbuffers' is used and fax tones are detected, the channel will dynamically
3590 switch to the configured faxbuffers policy. For example, to use 6 buffers
3591 and a 'full' buffer policy for a fax transmission, add:
3593 The faxbuffers configuration will be in affect until the call is torn down.
3594 * Added service message support for 4ESS/5ESS switches.
3598 * For DAHDI channels, the CHANNEL() dialplan function now
3599 supports changing the channel's buffer policy (for the current
3600 call only), using this syntax:
3602 exten => s,n,Set(CHANNEL(buffers)=6,full)
3604 This would change the channel to the 'full' buffer policy and
3605 6 (six) buffers. Possible options for this setting are the same
3606 as those in chan_dahdi.conf.
3607 * Added a new dialplan function, CURLOPT, which permits setting various
3608 options that may be useful with the CURL dialplan function, such as
3609 cookies, proxies, connection timeouts, passwords, etc.
3610 * Permit the syntax and synopsis fields of the corresponding dialplan
3611 functions to be individually set from func_odbc.conf.
3612 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
3613 * func_odbc now may specify an insert query to execute, when the write query
3614 affects 0 rows (usually indicating that no such row exists).
3615 * Added a new dialplan function, LISTFILTER, which permits removing elements
3616 from a set list, by name. Uses the same general syntax as the existing CUT
3617 and FIELDQTY dialplan functions, which also manage lists.
3618 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
3619 obtaining realtime data from the dialplan.
3620 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
3621 a subroutine when using the GoSub() and Return() applications.
3622 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
3623 of "core show function AUDIOHOOK_INHERIT" from the CLI
3624 * Added AES_ENCRYPT. For information on its use, please see the output
3625 of "core show function AES_ENCRYPT" from the CLI
3626 * Added AES_DECRYPT. For information on its use, please see the output
3627 of "core show function AES_DECRYPT" from the CLI
3628 * func_odbc now supports database transactions across multiple queries.
3632 * Scheduled meetme conferences may now have their end times extended by
3634 * app_authenticate now gives the ability to select a prompt other than
3636 * app_directory now pays attention to the searchcontexts setting in
3637 voicemail.conf and will look through all contexts, if no context is
3638 specified in the initial argument.
3639 * A new application, Originate, has been introduced, that allows asynchronous
3640 call origination from the dialplan.
3641 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
3642 in addition to the setting in the "general" context.
3643 * Added ConfBridge dialplan application which does conference bridges without
3644 DAHDI. For information on its use, please see the output of
3645 "core show application ConfBridge" from the CLI.
3649 * The Asterisk CLI has a new command, "channel redirect", which is similar in
3650 operation to the AMI Redirect action.
3651 * extensions.conf now allows you to use keyword "same" to define an extension
3652 without actually specifying an extension. It uses exactly the same pattern
3653 as previously used on the last "exten" line. For example:
3654 exten => 123,1,NoOp(something)
3655 same => n,SomethingElse()
3656 * musiconhold.conf classes of type 'files' can now use relative directory paths,
3657 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
3658 * All deprecated CLI commands are removed from the sourcecode. They are now handled
3659 by the new clialiases module. See cli_aliases.conf.sample file.
3660 * Times within timespecs are now accurate down to the minute. This is a change
3661 from historical Asterisk, which only provided timespecs rounded to the nearest
3662 even (read: evenly divisible by 2) minute mark.
3663 * The realtime switch now supports an option flag, 'p', which disables searches for
3665 * In addition to a time range and date range, timespecs now accept a 5th optional
3666 argument, timezone. This allows you to perform time checks on alternate
3667 timezones, especially if those daylight savings time ranges vary from your
3668 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
3670 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
3671 give you the correct output for an asterisk box behind nat. It will give you the
3672 externhost and localnet settings.
3673 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
3674 can connect calls in passthrough mode, as well as record and play back files.
3675 * Successful and unsuccessful call pickup can now be alerted through sounds, by
3676 using pickupsound and pickupfailsound in features.conf.
3677 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
3678 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
3679 instead of the /var/run/asterisk.pid where it used to be. This will make
3680 installs as non-root easier to manage.
3685 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
3686 be written; they will no longer be explicitly written.
3688 Asterisk Manager Interface
3689 --------------------------
3690 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
3691 a non-empty value) in your request. If you do this, any pending AMI events will
3692 *not* be included in the response to your request as they would normally, but
3693 will be left in the event queue for the next request you make to retrieve. For
3694 some applications, this will allow you to guarantee that you will only see
3695 events in responses to 'WaitEvent' actions, and can better know when to expect them.
3696 To know whether the Asterisk server supports this header or not, your client can
3697 inspect the first response back from the server to see if it includes this header:
3699 Pragma: SuppressEvents
3701 If this is included, the server supports event suppression.
3703 * Added 4 new Actions to list skinny device(s) and line(s)
3709 LDAP Schema File Additions
3710 --------------------------
3711 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
3712 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
3714 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
3715 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
3716 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
3717 * Removed redundant IPaddr (there's already IPAddress)
3718 - Gives more configuration Flags for SIP-Users available (tested)
3719 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
3720 without extensibleObject (which really should be the last resort); gives
3721 also additional possibilities for LDAP-filter
3723 ------------------------------------------------------------------------------
3724 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
3725 ------------------------------------------------------------------------------
3727 Device State Handling
3728 ---------------------
3729 * The event infrastructure in Asterisk got another big update to help support
3730 distributed events. It currently supports distributed device state and
3731 distributed Voicemail MWI (Message Waiting Indication). A new module has
3732 been merged, res_ais, which facilitates communicating events between servers.
3733 It uses the SAForum AIS (Service Availability Forum Application Interface
3734 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
3735 a cluster of Asterisk servers, and to share events between them. For more
3736 information on setting this up, refer to the Distributed Device State section
3737 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
3741 * Added a new dialplan function, AST_CONFIG(), which allows you to access
3742 variables from an Asterisk configuration file.
3743 * The JACK_HOOK function now has a c() option to supply a custom client name.
3744 * Added two new dialplan functions from libspeex for audio gain control and
3745 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
3746 rx directions of a channel from the dialplan.
3747 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
3748 based on other parameters. The default is still to search based on the
3749 forwarding station ID. However, there are new options that allow you to search
3750 based on the message desk terminal ID, or the message desk number.
3751 * TIMEOUT() has been modified to be accurate down to the millisecond.
3752 * ENUM*() functions now include the following new options:
3753 - 'u' returns the full URI and does not strip off the URI-scheme.
3754 - 's' triggers ISN specific rewriting
3755 - 'i' looks for branches into an Infrastructure ENUM tree
3756 - 'd' for a direct DNS lookup without any flipping of digits.
3757 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
3758 * CHANNEL() now has options for the maximum, minimum, and standard or normal
3759 deviation of jitter, rtt, and loss for a call using chan_sip.
3761 DAHDI channel driver (chan_dahdi) Changes
3762 ----------------------------------------
3763 * Channels can now be configured using named sections in chan_dahdi.conf, just
3764 like other channel drivers, including the use of templates.
3765 * The default for pridialplan has changed from 'national' to 'unknown'.
3769 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
3770 to something that matches the pattern a hint will be created using the contents
3771 and variables evaluated.
3772 * Dialplan matching has been extended to allow an extension to return to the
3773 PBX core to wait for more digits. This is done by using the new dialplan
3774 application called "Incomplete". This will permit a whole new level of
3775 extension control, by giving the administrator more control over early
3776 matches employing one of the short-circuit pattern match operators. Note
3777 that custom applications can trigger this same behavior by returning the
3778 special value AST_PBX_INCOMPLETE.
3782 * Directory now permits both first and last names to be matched at the same
3783 time. In addition, the number of digits to enter of the name can be set in
3784 the arguments to Directory; previously, you could enter only 3, regardless
3785 of how many names are in your company. For large companies, this should be
3787 * Voicemail now permits a mailbox setting to wrap around from first to last
3788 messages, if the "messagewrap" option is set to a true value.
3789 * Voicemail now permits an external script to be run, for password validation.
3790 The script should output "VALID" or "INVALID" on stdout, depending upon the
3791 wish to validate or invalidate the password given. Arguments are:
3792 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
3794 * Dial has a new option: F(context^extension^pri), which permits a callee to
3795 continue in the dialplan, at the specified label, if the caller hangs up.
3796 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
3797 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
3798 * The Jack application now has a c() option to supply a custom client name.
3799 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
3800 like the pre-existing whisper mode, except that the spy can also talk to the
3801 participant on the bridged channel as well.
3802 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
3803 to be spoken instead of the channel name or number. For more information on the
3804 use of this option, issue the command "core show application ChanSpy" from the
3806 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
3807 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
3808 words, if using the 'd' option, it is not possible to enter a number to append to
3809 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
3810 change to whisper mode, and pressing 6 will change to barge mode.
3811 * ExternalIVR now takes several options that affect the way it performs, as
3812 well as having several new commands. Please see the External IVR page on the Asterisk
3813 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
3814 * Added ability to communicate over a TCP socket instead of forking a child process for the
3815 ExternalIVR application.
3816 * ChanIsAvail has a new option, 'a', which will return all available channels instead
3817 of just the first one if you give the function more then one channel to check.
3818 * PrivacyManager now takes an option where you can specify a context where the
3819 given number will be matched. This way you have more control over who is allowed
3820 and it stops the people who blindly enter 10 digits.
3821 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
3822 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
3823 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
3824 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
3825 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
3826 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
3827 * The Dial() application no longer copies the language used by the caller to the callee's
3828 channel. If you desire for the caller's channel's language to be used for file playback
3829 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
3830 * SendImage() no longer hangs up the channel on error; instead, it sets the
3831 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
3832 'UNSUPPORTED'. This change makes SendImage() more consistent with other
3834 * Park has a new option, 's', which silences the announcement of the parking space number.
3835 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
3836 invalid input and will be assumed to mean that no timeout is desired.
3840 * Added DNS manager support to registrations for peers referencing peer entries.
3841 DNS manager runs in the background which allows DNS lookups to be run asynchronously
3842 as well as periodically updating the IP address. These properties allow for
3843 better performance as well as recovery in the event of an IP change.
3844 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
3845 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
3846 These changes also provide performance improvements for call setup and tear down.
3847 * Added ability to specify registration expiry time on a per registration basis in
3849 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
3851 * Added t38pt_usertpsource option. See sip.conf.sample for details.
3852 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
3853 * 'sip show peers' and 'sip show users' display their entries sorted in
3854 alphabetical order, as opposed to the order they were in, in the config
3856 * Videosupport now supports an additional option, "always", which always sets
3857 up video RTP ports, even on clients that don't support it. This helps with
3858 callfiles and certain transfers to ensure that if two video phones are
3859 connected, they will always share video feeds.
3863 * Existing DNS manager lookups extended to check for SRV records.
3864 * IAX2 encryption support has been improved to support periodic key rotation
3865 within a call for enhanced security. The option "keyrotate" has been
3866 provided to disable this functionality to preserve backwards compatibility
3867 with older versions of IAX2 that do not support key rotation.
3871 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
3872 data tree based on the given <path>.
3873 * New CLI command "data show providers" that will display all the registered
3875 * New CLI command, "config reload <file.conf>" which reloads any module that
3876 references that particular configuration file. Also added "config list"
3877 which shows which configuration files are in use.
3878 * New CLI commands, "pri show version" and "ss7 show version" that will
3879 display which version of libpri and libss7 are being used, respectively.
3880 A new API call was added so trunk will now have to be compiled against
3881 a versions of libpri and libss7 that have them or it will not know that
3882 these libraries exist.
3883 * The commands "core show globals", "core set global" and "core set chanvar" has
3884 been deprecated in favor of the more semanticly correct "dialplan show globals",
3885 "dialplan set chanvar" and "dialplan set global".
3886 * New CLI command "dialplan show chanvar" to list all variables associated
3887 with a given channel.
3891 * Addresses managed by DNS manager now can check to see if there is a DNS
3892 SRV record for a given domain and will use that hostname/port if present.
3894 AMI - The manager (TCP/TLS/HTTP)
3895 --------------------------------
3896 * The Status command now takes an optional list of variables to display
3897 along with channel status.
3898 * The QueueEntry event now also includes the channel's uniqueid
3902 * res_odbc no longer has a limit of 1023 total possible unshared connections,
3903 as some people were running into this limit. This limit has been increased
3908 * The TRANSFER queue log entry now includes the the caller's original
3909 position in the transferred-from queue.
3910 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
3911 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
3912 as well as an explanation about timeout options in general
3913 * Added a new option - C - for forcing the "answered elsewhere" flag on
3914 cancellation of calls in to members of the queue. This is to avoid the
3915 call to a member of a queue having the call listed as a "missed call".
3919 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
3920 adaptive capabilities. What this means in practical terms is that if your
3921 realtime table lacks critical fields, Asterisk will now emit warnings to
3922 that effect. Also, some of the realtime drivers have the ability (if
3923 configured) to automatically add those columns to the table with the
3924 correct type and length.
3928 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
3929 the 'setvar' option to cause a given audio file to be played upon completion
3930 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
3931 Skinny channels only.
3932 * You can now compile Asterisk against the Hoard Memory Allocator, see the
3933 Hoard page on the Asterisk wiki for more information:
3934 https://wiki.asterisk.org/wiki/x/pQBB
3935 * Config file variables may now be appended to, by using the '+=' append
3936 operator. This is most helpful when working with long SQL queries in
3937 func_odbc.conf, as the queries no longer need to be specified on a single
3939 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
3940 which will add a second to the billsec when the ending
3941 time is set, if the number in the microseconds field of the end time is
3942 greater than the number of microseconds in the answer time. This allows
3943 users to count the 'initiated' seconds in their billing records.
3945 ------------------------------------------------------------------------------
3946 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
3947 ------------------------------------------------------------------------------
3949 AMI - The manager (TCP/TLS/HTTP)
3950 --------------------------------
3951 * Manager has undergone a lot of changes, all of them documented
3952 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
3953 * Manager version has changed to 1.1
3954 * Added a new action 'CoreShowChannels' to list currently defined channels
3955 and some information about them.
3956 * Added a new action 'SIPshowregistry' to list SIP registrations.
3957 * Added TLS support for the manager interface and HTTP server
3958 * Added the URI redirect option for the built-in HTTP server
3959 * The output of CallerID in Manager events is now more consistent.
3960 CallerIDNum is used for number and CallerIDName for name.
3961 * Enable https support for builtin web server.
3962 See configs/http.conf.sample for details.
3963 * Added a new action, GetConfigJSON, which can return the contents of an
3964 Asterisk configuration file in JSON format. This is intended to help
3965 improve the performance of AJAX applications using the manager interface
3967 * SIP and IAX manager events now use "ChannelType" in all cases where we
3968 indicate channel driver. Previously, we used a mixture of "Channel"
3969 and "ChannelDriver" headers.
3970 * Added a "Bridge" action which allows you to bridge any two channels that
3971 are currently active on the system.
3972 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
3973 the voicemail users setup.
3974 * Added 'DBDel' and 'DBDelTree' manager commands.
3975 * cdr_manager now reports events via the "cdr" level, separating it from
3976 the very verbose "call" level.
3977 * Manager users are now stored in memory. If you change the manager account
3978 list (delete or add accounts) you need to reload manager.
3979 * Added Masquerade manager event for when a masquerade happens between
3981 * Added "manager reload" command for the CLI
3982 * Lots of commands that only provided information are now allowed under the
3983 Reporting privilege, instead of only under Call or System.
3984 * The IAX* commands now require either System or Reporting privilege, to
3985 mirror the privileges of the SIP* commands.
3986 * Added ability to retrieve list of categories in a config file.
3987 * Added ability to retrieve the content of a particular category.
3988 * Added ability to empty a context.
3989 * Created new action to create a new file.
3990 * Updated delete action to allow deletion by line number with respect to category.
3991 * Added new action insert to add new variable to category at specified line.
3992 * Updated action newcat to allow new category to be inserted in file above another
3994 * Added new event "JitterBufStats" in the IAX2 channel
3995 * Originate now requires the Originate privilege and, if you want to call out
3996 to a subshell, it requires the System privilege, as well. This was done to
3997 enhance manager security.
3998 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
3999 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
4000 or manager show command Atxfer from the CLI
4001 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
4002 details or manager show command IAXregistry from the CLI
4006 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
4007 state in the dialplan, as well as creating custom device states that are
4008 controllable from the dialplan.
4009 * Extend CALLERID() function with "pres" and "ton" parameters to
4010 fetch string representation of calling number presentation indicator
4011 and numeric representation of type of calling number value.
4012 * MailboxExists converted to dialplan function
4013 * A new option to Dial() for telling IP phones not to count the call
4014 as "missed" when dial times out and cancels.
4015 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
4016 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
4017 held for any given channel. Also, locks are automatically freed when a
4019 * Added HINT() dialplan function that allows retrieving hint information.
4020 Hints are mappings between extensions and devices for the sake of
4021 determining the state of an extension. This function can retrieve the list
4022 of devices or the name associated with a hint.
4023 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
4025 * Added SYSINFO() dialplan function which allows retrieval of system information
4026 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
4027 the existence of a dialplan target.
4028 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
4029 upper and lower case, respectively.
4030 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
4031 ID for the call (not the Asterisk call ID or unique ID), provided that the
4032 channel driver supports this. For SIP, you get the SIP call-ID for the
4033 bridged channel which you can store in the CDR with a custom field.
4037 * Added CLI permissions, config file: cli_permissions.conf
4038 default is to allow all commands for every local user/group.
4039 Also this new feature added three new CLI commands:
4040 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
4041 - cli reload permissions
4042 - cli show permissions
4043 * New CLI command "core show hint" (usage: core show hint <exten>)
4044 * New CLI command "core show settings"
4045 * Added 'core show channels count' CLI command.
4046 * Added the ability to set the core debug and verbose values on a per-file basis.
4047 * Added 'queue pause member' and 'queue unpause member' CLI commands
4048 * Ability to set process limits ("ulimit") without restarting Asterisk
4049 * Enhanced "agi debug" to print the channel name as a prefix to the debug
4050 output to make debugging on busy systems much easier.
4051 * New CLI commands "dialplan set extenpatternmatching true/false"
4052 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
4053 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
4054 listed in the startup_commands section of cli.conf will get executed.
4055 * Added a CLI command, "devstate change", which allows you to set custom device
4056 states from the func_devstate module that provides the DEVICE_STATE() function
4057 and handling of the "Custom:" devices.
4058 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
4059 sorted into the different possible callbacks, with the number of entries
4060 currently scheduled for each. Gives you a feel for how busy the sip channel
4062 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
4063 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
4064 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
4068 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
4069 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
4070 for a received call. If it is detected, the channel will jump to the
4071 'fax' extension in the dialplan.
4072 * The default SIP useragent= identifier now includes the Asterisk version
4073 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
4074 If set, and the incoming request carries authentication info,
4075 the username to match in the users list is taken from the Digest header
4076 rather than from the From: field. This feature is considered experimental.
4077 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
4078 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
4079 * The "localmask" setting was removed in version 1.2 and the reminder about it
4080 being removed is now also removed.
4081 * A new option "busylevel" for setting a level of calls where asterisk reports
4082 a device as busy, to separate it from call-limit. This value is also added
4083 to the SIP_PEER dialplan function.
4084 * A new realtime family called "sipregs" is now supported to store SIP registration
4085 data. If this family is defined, "sippeers" will be used for configuration and
4086 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
4087 registration data, as before.
4088 * The SIPPEER function have new options for port address,&nb
projects / asterisk/asterisk.git / blob