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 12 to Asterisk 13 --------------------
13 ------------------------------------------------------------------------------
15 ------------------------------------------------------------------------------
16 --- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------
17 ------------------------------------------------------------------------------
21 * A new Playback URI 'tone' has been added. Tones are specified either as
22 an indication name (e.g. 'tone:busy') from indications.conf or as a tone
23 pattern (e.g. 'tone:240/250,0/250'). Tones differ from normal playback
24 URIs in that they must be stopped manually and will continue to occupy
25 a channel's ARI control queue until they are stopped. They also can not
26 be rewound or fastforwarded.
28 ------------------------------------------------------------------------------
29 --- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
30 ------------------------------------------------------------------------------
33 --------------------------
34 * Record application now has an option 'o' which allows 0 to act as an exit
35 key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'
36 * Monitor() - A new option, B(), has been added that will turn on a periodic
37 beep while the call is being recorded.
40 --------------------------
41 * A new function was added: PERIODIC_HOOK. This allows running a periodic
42 dialplan hook on a channel. Any audio generated by this hook will be
43 injected into the call.
46 --------------------------
47 * ChanSpy now accepts a channel uniqueid or a fully specified channel name
48 as the chanprefix parameter if the 'u' option is specified.
51 --------------------------
52 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
53 conference user menus.
55 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
56 menus, bridge settings, and user settings that have been applied by the
57 CONFBRIDGE dialplan function.
59 * The ConfBridge dialplan application now sets a channel variable,
60 CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
61 how a channel exited the conference.
63 * Added conference user option 'announce_join_leave_review'. This option
64 implies 'announce_join_leave' with the added effect that the user will
65 be asked if they want to confirm or re-record the recording of their
66 name when entering the conference
69 --------------------------
70 * At exit, the Directory application now sets a channel variable
71 DIRECTORY_RESULT to one of the following based on the reason for exiting:
72 OPERATOR user requested operator by pressing '0' for operator
73 ASSISTANT user requested assistant by pressing '*' for assistant
74 TIMEOUT user pressed nothing and Directory stopped waiting
75 HANGUP user's channel hung up
76 SELECTED user selected a user from the directory and is routed
77 USEREXIT user pressed '#' from the selection prompt to exit
78 FAILED directory failed in a way that wasn't accounted for. Dang.
81 --------------------------
82 * MusicOnHold streams (all modes other than "files") now support wide band
86 --------------------------
87 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
88 and for the channel executing Page respectively.
91 --------------------------
92 * PickupChan now accepts channel uniqueids of channels to pickup.
95 --------------------------
96 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
97 to 'true' (case insensitive), then any Say application (SayNumber,
98 SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
99 anticipate DTMF. If DTMF is received, these applications will behave like
100 the background application and jump to the received extension once a match
101 is established or after a short period of inactivity.
104 -------------------------
105 * A new function, MIXMONITOR, has been added to allow access to individual
106 instances of MixMonitor on a channel.
107 * A new option, B(), has been added that will turn on a periodic beep while the
108 call is being recorded.
112 -------------------------
115 -------------------------
116 * TEL URI support for inbound INVITE requests has been added. chan_sip will
117 now handle TEL schemes in the Request and From URIs. The phone-context in
118 the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on
122 -------------------------
123 * Core Show Locks output now includes Thread/LWP ID if the platform
124 supports this feature.
125 * New "logger add channel" and "logger remove channel" CLI commands have
126 been added to allow creation and deletion of dynamic logger channels
127 without configuration changes. These dynamic logger channels will only
128 exist until the next restart of asterisk.
132 * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
133 the new AST_SORCERY diaplan function.
137 * The live recording object on recording events now contains a target_uri
138 field which contains the URI of what is being recorded.
140 * The bridge type used when creating a bridge is now a comma separated list of
141 bridge properties. Valid options are: mixing, holding, dtmf_events, and
144 * A channelId can now be provided when creating a channel, either in the
145 uri (POST channels/my-channel-id) or as query parameter. A local channel
146 will suffix the second channel id with ';2' unless provided as query
147 parameter otherChannelId.
149 * A bridgeId can now be provided when creating a bridge, either in the uri
150 (POST bridges/my-bridge-id) or as a query parameter.
152 * A playbackId can be provided when starting a playback, either in the uri
153 (POST channels/my-channel-id/play/my-playback-id /
154 POST bridges/my-bridge-id/play/my-playback-id) or as a query parameter.
156 * A snoop channel can be started with a snoopId, in the uri or query.
160 * Originate now takes optional parameters ChannelId and OtherChannelId,
161 used to set the UniqueId on creation. The other id is assigned to the
162 second channel when dialing LOCAL, or defaults to appending ;2 if only
163 the single Id is given.
165 * The Mixmonitor action now has a "Command" header that can be used to
166 indicate a post-process command to run once recording finishes.
170 * A new set of Alembic scripts has been added for CDR tables. This will create
171 a 'cdr' table with the default schema that Asterisk expects.
175 * A new module, res_hep, has been added, that acts as a generic packet
176 capture agent for the Homer Encapsulation Protocol (HEP) version 3.
177 It can be configured via hep.conf. Other modules can use res_hep to send
178 message traffic to a HEP capture server.
182 * A new module, res_hep_pjsip, has been added that will forward PJSIP
183 message traffic to a HEP capture server. See res_hep for more
188 * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
189 be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
191 * Added the following new CLI commands:
192 - "pjsip show contacts" - list all current PJSIP contacts.
193 - "pjsip show contact" - show specific information about a current PJSIP
195 - "pjsip show channel" - show detailed information about a PJSIP channel.
199 * A new module, res_pjsip_multihomed handles situations where the system
200 Asterisk is running out has multiple interfaces. res_pjsip_multihomed
201 determines which interface should be used during message sending.
203 res_pjsip_pidf_digium_body_supplement
205 * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
206 request body formatting for presence support in Digium phones.
208 res_pjsip_send_to_voicemail
210 * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
211 particular headers to transfer a PJSIP channel directly to a particular
212 extension that has VoiceMail. This is intended to be used with Digium
213 phones that support this feature.
215 res_pjsip_outbound_registration
217 * A new CLI command has been added: "pjsip show registrations", which lists
218 all configured PJSIP registrations
221 ------------------------------------------------------------------------------
222 --- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
223 ------------------------------------------------------------------------------
227 * Added a new module that provides AMI control over MWI within Asterisk,
228 res_mwi_external_ami. Note that this module depends on res_mwi_external;
229 for more information on enabling this module, see res_mwi_external.
230 This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
231 the MWIGet/MWIGetComplete events.
233 * The DialStatus field in the DialEnd event can now contain additional
234 statuses that convey how the dial operation terminated. This includes
235 ABORT, CONTINUE, and GOTO.
237 * AMI will now emit security events. A new class authorization has been
238 added in manager.conf for the security events, 'security'. The new events
240 - FailedACL - raised when a request violates an ACL check
241 - InvalidAccountID - raised when a request fails an authentication
242 check due to an invalid account ID
243 - SessionLimit - raised when a request fails due to exceeding the
244 number of allowed concurrent sessions for a service
245 - MemoryLimit - raised when a request fails due to an internal memory
247 - LoadAverageLimit - raised when a request fails because a configured
248 load average limit has been reached
249 - RequestNotAllowed - raised when a request is not allowed by
251 - AuthMethodNotAllowed - raised when a request used an authentication
252 method not allowed by the service
253 - RequestBadFormat - raised when a request is received with bad formatting
254 - SuccessfulAuth - raised when a request successfully authenticates
255 - UnexpectedAddress - raised when a request has a different source address
256 then what is expected for a session already in progress with a service
257 - ChallengeResponseFailed - raised when a request's attempt to authenticate
258 has been challenged, and the request failed the authentication challenge
259 - InvalidPassword - raised when a request provides an invalid password
260 during an authentication attempt
261 - ChallengeSent - raised when an Asterisk service send an authentication
262 challenge to a request
263 - InvalidTransport - raised when a request attempts to use a transport not
264 allowed by the Asterisk service
266 * Bridge related events now have two additional fields: BridgeName and
267 BridgeCreator. BridgeName is a descriptive name for the bridge;
268 BridgeCreator is the name of the entity that created the bridge. This
269 affects the following events: ConfbridgeStart, ConfbridgeEnd,
270 ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
271 ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
272 AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
276 * The Bridge data model now contains the additional fields 'name' and
277 'creator'. The 'name' field conveys a descriptive name for the bridge;
278 the 'creator' field conveys the name of the entity that created the bridge.
279 This affects all responses to HTTP requests that return a Bridge data model
280 as well as all event derived data models that contain a Bridge data model.
281 The POST /bridges operation may now optionally specify a name to give to
282 the bridge being created.
284 * Added a new ARI resource 'mailboxes' which allows the creation and
285 modification of mailboxes managed by external MWI. Modules res_mwi_external
286 and res_stasis_mailbox must be enabled to use this resource. For more
287 information on external MWI control, see res_mwi_external.
289 * Added new events for externally initiated transfers. The event
290 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
291 of a bridge in the ARI controlled application to the dialplan; the
292 BridgeAttendedTransfer event is raised when a channel initiates an
293 attended transfer of a bridge in the ARI controlled application to the
296 * Channel variables may now be specified as a body parameter to the
297 POST /channels operation. The 'variables' key in the JSON is interpreted
298 as a sequence of key/value pairs that will be added to the created channel
299 as channel variables. Other parameters in the JSON body are treated as
300 query parameters of the same name.
304 * Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be
305 automatically handled by the HTTP server if a request is received with a
306 Transfer-Encoding type of "chunked".
310 * Path support has been added with the 'support_path' option in registration
313 * A 'debug' option has been added to the globals section that will allow
314 sip messages to be logged.
316 * A 'set_var' option has been added to endpoints that will automatically
317 set the desired variable(s) on a channel created for that endpoint.
319 * Several new tables and columns have been added to the realtime schema for
320 the res_pjsip related modules. See the UPGRADE.txt notes for updating
325 * A new module, res_mwi_external, has been added to Asterisk. This module
326 acts as a base framework that other modules can build on top of to allow
327 an external system to control MWI within Asterisk. For implementations
328 that make use of res_mwi_external, see res_mwi_external_ami and
329 res_ari_mailboxes. Note that res_mwi_external canflicts with other modules
330 that may produce MWI themselves, such as app_voicemail. res_mwi_external
331 and other modules that depend on it cannot be built or loaded with
332 app_voicemail present.
336 * DNS functionality will now automatically be enabled if the system configured
337 nameservers can be retrieved. If the system configured nameservers can not be
338 retrieved the functionality will resort to using system resolution. Functionalty
339 such as SRV records and failover will not be available if system resolution
342 ------------------------------------------------------------------------------
343 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
344 ------------------------------------------------------------------------------
349 Asterisk 12 is a standard release of the Asterisk project. As such, the
350 focus of development for this release was on core architectural changes and
351 major new features. This includes:
352 * A more flexible bridging core based on the Bridging API
353 * A new internal message bus, Stasis
354 * Major standardization and consistency improvements to AMI
355 * Addition of the Asterisk RESTful Interface (ARI)
356 * A new SIP channel driver, chan_pjsip
357 In addition, as the vast majority of bridging in Asterisk was migrated to the
358 Bridging API used by ConfBridge, major changes were made to most of the
359 interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
361 Specifications have been written for the affected interfaces. These
362 specifications are available on the Asterisk wiki:
363 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
364 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
365 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
367 It is *highly* recommended that anyone migrating to Asterisk 12 read the
368 information regarding its release both in this file and in the accompanying
369 UPGRADE.txt file. More detailed information on the major changes can be found
370 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
375 * Added build option DISABLE_INLINE. This option can be used to work around a
376 bug in gcc. For more information, see
377 http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
379 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
380 the CHANNEL_TRACE build option were incompatible with the new bridging
383 * Asterisk now optionally uses libxslt to improve XML documentation generation
384 and maintainability. If libxslt is not available on the system, some XML
385 documentation will be incomplete.
387 * Asterisk now depends on libjansson. If a package of libjansson is not
388 available on your distro, please see http://www.digip.org/jansson/.
390 * Asterisk now depends on libuuid and, optionally, uriparser. It is
391 recommended that you install uriparser, even if it is optional.
393 * The new SIP stack and channel driver uses a particular version of PJSIP.
394 Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
395 configuring and installing PJSIP for usage with Asterisk.
397 * Optional API was re-implemented to be more portable, and no longer requires
398 weak reference support from the compiler. The build option OPTIONAL_API may
399 be disabled to disable Optional API support.
406 * Along with AgentRequest, this application has been modified to be a
407 replacement for chan_agent. The act of a channel calling the AgentLogin
408 application places the channel into a pool of agents that can be
409 requested by the AgentRequest application. Note that this application, as
410 well as all other agent related functionality, is now provided by the
411 app_agent_pool module. See chan_agent and AgentRequest for more information.
413 * This application no longer performs agent authentication. If authentication
414 is desired, the dialplan needs to perform this function using the
415 Authenticate or VMAuthenticate application or through an AGI script before
418 * If this application is called and the agent is already logged in, the
419 dialplan will continue exection with the AGENT_STATUS channel variable set
420 to ALREADY_LOGGED_IN.
422 * The agents.conf schema has changed. Rather than specifying agents on a
423 single line in comma delineated fashion, each agent is defined in a separate
424 context. This allows agents to use the power of context templates in their
427 * A number of parameters from agents.conf have been removed. This includes
428 maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
429 urlprefix, and savecallsin. These options were obsoleted by the move from
430 a channel driver model to the bridging/application model provided by
435 * A new application, this will request a logged in agent from the pool and
436 bridge the requested channel with the channel calling this application.
437 Logged in agents are those channels that called the AgentLogin application.
438 If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
439 application will be set with an appropriate error value.
443 * This application has been removed. It was a holdover from when
444 AgentCallbackLogin was removed.
448 * Added support for additional Ademco DTMF signalling formats, including
449 Express 4+1, Express 4+2, High Speed and Super Fast.
451 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
452 call time, in milliseconds, to run the application.
454 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
455 maximum number of times to retry the call.
457 * Added a new configuration option answait. If set, the AlarmReceiver
458 application will wait the number of milliseconds specified by answait
459 after the channel has answered. Valid values range between 500
460 milliseconds and 10000 milliseconds.
462 * Added configuration option no_group_meta. If enabled, grouping of metadata
463 information in the AlarmReceiver log file will be skipped.
467 * It is now no longer possible to bypass updating the CDR on the channel
468 when answering. CDRs reflect the state of the channel and will always
469 reflect the time they were Answered.
473 * A new application in Asterisk, this will place the calling channel
474 into a holding bridge, optionally entertaining them with some form of
475 media. Channels participating in a holding bridge do not interact with
476 other channels in the same holding bridge. Optionally, however, a channel
477 may join as an announcer. Any media passed from an announcer channel is
478 played to all channels in the holding bridge. Channels leave a holding
479 bridge either when an optional timer expires, or via the ChannelRedirect
480 application or AMI Redirect action.
484 * All participants in a bridge can now be kicked out of a conference room
485 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
486 command, i.e., 'confbridge kick <conference> all'
488 * CLI output for the 'confbridge list' command has been improved. When
489 displaying information about a particular bridge, flags will now be shown
490 for the participating users indicating properties of that user.
492 * The ConfbridgeList event now contains the following fields: WaitMarked,
493 EndMarked, and Waiting. This displays additional properties about the
494 user's profile, as well as whether or not the user is waiting for a
495 Marked user to enter the conference.
497 * Added a new option for conference recording, record_file_append. If enabled,
498 when the recording is stopped and then re-started, the existing recording
499 will be used and appended to.
501 * ConfBridge now has the ability to set the language of announcements to the
502 conference. The language can be set on a bridge profile in confbridge.conf
503 or by the dialplan function CONFBRIDGE(bridge,language)=en.
507 * The channel variable CPLAYBACKSTATUS may now return the value
508 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
509 such as AMI. See the AMI action ControlPlayback for more information.
513 * Added the 'a' option, which allows the caller to enter in an additional
514 alias for the user in the directory. This option must be used in conjunction
515 with the 'f', 'l', or 'b' options. Note that the alias for a user can be
516 specified in voicemail.conf.
520 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
521 fields. Instead, if a channel is in a bridge, it includes a BridgeID field
522 containing the unique ID of the bridge that the channel happens to be in.
526 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
527 for more information.
529 * Variables are no longer purged from the original CDR. See the 'v' option for
532 * The 'A' option has been removed. The Answer time on a CDR is never updated
535 * The 'd' option has been removed. The disposition on a CDR is a function of
536 the state of the channel and cannot be altered.
538 * The 'D' option has been removed. Who the Party B is on a CDR is a function
539 of the state of the respective channels involved in the CDR and cannot be
542 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
543 such that the start time and, if applicable, the answer time was updated.
544 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
545 'r' option now triggers the Reset, setting the start time (and answer time
546 if applicable) to the current time. Note that the 'a' option still sets
547 the answer time to the current time if the channel was already answered.
549 * The 's' option has been removed. A variable can be set on the original CDR
550 if desired using the CDR function, and removed from a forked CDR using the
553 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
554 longer applies in the CDR engine.
556 * The 'v' option now prevents the copy of the variables from the original CDR
557 to the forked CDR. Previously the variables were always copied but were
558 removed from the original. This was changed as removing variables from a CDR
559 can have unintended side effects - this option allows the user to prevent
560 propagation of variables from the original to the forked without modifying
565 * Added the 'n' option to MeetMe to prevent application of the DENOISE
566 function to a channel joining a conference. Some channel drivers that vary
567 the number of audio samples in a voice frame will experience significant
568 quality problems if a denoiser is attached to the channel; this option gives
569 them the ability to remove the denoiser without having to unload func_speex.
573 * The 'b' option now includes conferences as well as sounds played to the
576 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
577 running during a transfer. If a MixMonitor is started on a channel,
578 the MixMonitor will continue to record the audio passing through the
579 channel even in the presence of transfers.
583 * The NoCDR application is deprecated. Please use the CDR_PROP function to
586 * While the NoCDR application will prevent CDRs for a channel from being
587 propagated to registered CDR backends, it will not prevent that data from
588 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
589 function that enables CDRs on a channel will restore those records that have
590 not yet been finalized.
594 * The app_parkandannounce module has been removed. The application
595 ParkAndAnnounce is now provided by the res_parking module. See the
596 res_parking changes for more information.
600 * Added queue available hint. The hint can be added to the dialplan using the
601 following syntax: exten,hint,Queue:{queue_name}_avail
602 For example, if the name of the queue is 'markq':
603 exten => 8501,hint,Queue:markq_avail
604 This will report 'InUse' if there are no logged in agents or no free agents.
605 It will report 'Idle' when an agent is free.
607 * Queues now support a hint for member paused state. The hint uses the form
608 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
609 are the name of the queue and the name of the member to subscribe to,
610 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
611 Members will show as In Use when paused.
613 * The configuration options eventwhencalled and eventmemberstatus have been
614 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
615 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
616 sent. The "Variable" fields will also no longer exist on the Agent* events.
617 These events can be filtered out from a connected AMI client using the
618 eventfilter setting in manager.conf.
620 * The queue log now differentiates between blind and attended transfers. A
621 blind transfer will result in a BLINDTRANSFER message with the destination
622 context and extension. An attended transfer will result in an
623 ATTENDEDTRANSFER message. This message will indicate the method by which
624 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
625 for running an application on a bridge or channel, or "LINK" for linking
626 two bridges together with local channels. The queue log will also now detect
627 externally initiated blind and attended transfers and record the transfer
630 * When performing queue pause/unpause on an interface without specifying an
631 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
632 least one member of any queue exists for that interface.
634 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
635 for realtime queue log entries.
639 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
640 CDRs when they were previously disabled on a channel.
642 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
643 backends occurs on an as-needed basis in order to preserve linkedid
644 propagation and other needed behavior.
648 * A new application, this is similar to SayAlpha except that it supports
649 case sensitive playback of the specified characters. For example,
650 SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
654 * This application is deprecated in favor of CHANNEL(amaflags).
658 * The SendDTMF application will now accept 'W' as valid input. This will cause
659 the application to delay one second while streaming DTMF.
663 * A new application in Asterisk 12, this hands control of the channel calling
664 the application over to an external system. Currently, external systems
665 manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
669 * UserEvent will now handle duplicate keys by overwriting the previous value
672 * In addition to AMI, UserEvent invocations will now be distributed to any
673 interested Stasis applications.
677 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
678 system as mailbox@context. The rest of the system cannot add @default
679 to mailbox identifiers for app_voicemail that do not specify a context
680 any longer. It is a mailbox identifier format that should only be
681 interpreted by app_voicemail.
683 * The voicemail.conf configuration file now has an 'alias' configuration
684 parameter for use with the Directory application. The voicemail realtime
685 database table schema has also been updated with an 'alias' column.
690 * Pass through support has been added for both VP8 and Opus.
692 * Added format attribute negotiation for the Opus codec. Format attribute
693 negotiation is provided by the res_format_attr_opus module.
698 * Masquerades as an operation inside Asterisk have been effectively hidden
699 by the migration to the Bridging API. As such, many 'quirks' of Asterisk
700 no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
701 dropping of frame/audio hooks, and other internal implementation details
702 that users had to deal with. This fundamental change has large implications
703 throughout the changes documented for this version. For more information
704 about the new core architecture of Asterisk, please see the Asterisk wiki.
706 * Multiple parties in a bridge may now be transferred. If a participant in a
707 multi-party bridge initiates a blind transfer, a Local channel will be used
708 to execute the dialplan location that the transferer sent the parties to. If
709 a participant in a multi-party bridge initiates an attended transfer,
710 several options are possible. If the attended transfer results in a transfer
711 to an application, a Local channel is used. If the attended transfer results
712 in a transfer to another channel, the resulting channels will be merged into
715 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
716 driver specific. If the channel variable is set on the transferrer channel,
717 the sound will be played to the target of an attended transfer.
719 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
720 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
721 listed. Any more peers in the bridge will not be included in the list.
722 BRIDGEPEER is not valid in holding bridges like parking since those channels
723 do not talk to each other even though they are in a bridge.
725 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
726 and will contain a value if the BRIDGEPEER's channel driver supports it.
728 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
729 was responsible for an attended transfer in a similar fashion to
732 * Modules using the Configuration Framework or Sorcery must have XML
733 configuration documentation. This configuration documentation is included
734 with the rest of Asterisk's XML documentation, and is accessible via CLI
735 commands. See the CLI changes for more information.
737 AMI (Asterisk Manager Interface)
739 * Major changes were made to both the syntax as well as the semantics of the
740 AMI protocol. In particular, AMI events have been substantially improved
741 in this version of Asterisk. For more information, please see the AMI
742 specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
744 * AMI events that reference a particular channel or bridge will now always
745 contain a standard set of fields. When multiple channels or bridges are
746 referenced in an event, fields for at least some subset of the channels
747 and bridges in the event will be prefixed with a descriptive name to avoid
748 name collisions. See the AMI event documentation on the Asterisk wiki for
751 * The CLI command 'manager show commands' no longer truncates command names
752 longer than 15 characters and no longer shows authorization requirement
753 for commands. 'manager show command' now displays the privileges needed
754 for using a given manager command instead.
756 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
757 peer in its response if the peer has a subscribe context set.
759 * The SIPqualifypeer action now acknowledges the request once it has
760 established that the request is against a known peer. It also issues a new
761 event, 'SIPQualifyPeerDone', once the qualify action has been completed.
763 * The PlayDTMF action now supports an optional 'Duration' parameter. This
764 specifies the duration of the digit to be played, in milliseconds.
766 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
767 updates when changes occur instead of requiring the use of pollmailboxes.
769 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
770 AMI client to manipulate audio currently being played back on a channel. The
771 supported operations depend on the application being used to send audio to
772 the channel. When the audio playback was initiated using the ControlPlayback
773 application or CONTROL STREAM FILE AGI command, the audio can be paused,
774 stopped, restarted, reversed, or skipped forward. When initiated by other
775 mechanisms (such as the Playback application), the audio can be stopped,
776 reversed, or skipped forward.
778 * Channel related events now contain a snapshot of channel state, adding new
779 fields to many of these events.
781 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
782 in a future release. Please use the common 'Exten' field instead.
784 * The AMI event 'UserEvent' from app_userevent now contains the channel state
785 fields. The channel state fields will come before the body fields.
787 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
788 'UnParkedCall' have changed significantly in the new res_parking module.
790 The 'Channel' and 'From' headers are gone. For the channel that was parked
791 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
792 has a number of fields associated with it. The old 'Channel' header relayed
793 the same data as the new 'ParkeeChannel' header.
795 The 'From' field was ambiguous and changed meaning depending on the event.
796 for most of these, it was the name of the channel that parked the call
797 (the 'Parker'). There is no longer a header that provides this channel name,
798 however the 'ParkerDialString' will contain a dialstring to redial the
799 device that parked the call.
801 On UnParkedCall events, the 'From' header would instead represent the
802 channel responsible for retrieving the parkee. It receives a channel
803 snapshot labeled 'Retriever'. The 'from' field is is replaced with
806 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
808 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
809 fashion has changed the field names 'StartExten' and 'StopExten' to
810 'StartSpace' and 'StopSpace' respectively.
812 * The deprecated use of | (pipe) as a separator in the channelvars setting in
813 manager.conf has been removed.
815 * Channel Variables conveyed with a channel no longer contain the name of the
816 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
817 ChanVariable: bar=baz. When multiple channels are present in a single AMI
818 event, the various ChanVariable fields will contain a suffix that specifies
819 which channel they correspond to.
821 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
822 event always conveys the AMI event for a particular channel.
824 * All 'Reload' events have been consolidated into a single event type. This
825 event will always contain a Module field specifying the name of the module
826 and a Status field denoting the result of the reload. All modules now issue
827 this event when being reloaded.
829 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
830 fail to receive this event due to being connected after modules have loaded.
831 AMI connections that want to know when Asterisk is ready should listen for
832 the 'FullyBooted' event.
834 * app_fax now sends the same send fax/receive fax events as res_fax. The
835 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
836 now the 'ReceiveFAX' event.
838 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
839 'MusicOnHoldStop'. The sub type field has been removed.
841 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
842 carrier for another protocol.
844 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
845 options. 'Channel1' and 'Channel2' may be specified in order to play a tone
846 to the specific channel. 'Both' may be specified to play a tone to both
847 channels. The old 'yes' option is still accepted as a way of playing the
848 tone to Channel2 only.
850 * The AMI 'Status' response event to the AMI Status action replaces the
851 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
852 indicate what bridge the channel is currently in.
854 * The AMI 'Hold' event has been moved out of individual channel drivers, into
855 core, and is now two events: 'Hold' and 'Unhold'. The status field has been
858 * The AMI events in app_queue have been made more consistent with each other.
859 Events that reference channels (QueueCaller* and Agent*) will show
860 information about each channel. The (infamous) 'Join' and 'Leave' AMI
861 events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
863 * The 'MCID' AMI event now publishes a channel snapshot when available and
864 its non-channel-snapshot parameters now use either the "MCallerID" or
865 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
866 of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
867 parameters in the channel snapshot.
869 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
870 'AgentLogin' and 'AgentLogoff' respectively.
872 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
873 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
875 * 'ChannelUpdate' events have been removed.
877 * All AMI events now contain a 'SystemName' field, if available.
879 * Local channel optimization is now conveyed in two events:
880 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
881 when the Local channel driver begins attempting to optimize itself out of
882 the media path; the End event is sent after the channel halves have
883 successfully optimized themselves out of the media path.
885 * Local channel information in events is now prefixed with 'LocalOne' and
886 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
887 the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
888 and 'LocalOptimizationEnd' events.
890 * The option 'allowmultiplelogin' can now be set or overriden in a particular
891 account. When set in the general context, it will act as the default
892 setting for defined accounts.
894 * The 'BridgeAction' event was removed. It technically added no value, as the
895 Bridge Action already receives confirmation of the bridge through a
896 successful completion Event.
898 * The 'BridgeExec' events were removed. These events duplicated the events that
899 occur in the Briding API, and are conveyed now through BridgeCreate,
900 BridgeEnter, and BridgeLeave events.
902 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
903 previous versions. They now report all SR/RR packets sent/received, and
904 have been restructured to better reflect the data sent in a SR/RR. In
905 particular, the event structure now supports multiple report blocks.
907 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
908 raised when a blind transfer/attended transfer completes successfully.
909 They contain information about the transfer that just completed, including
910 the location of the transfered channel.
912 * Added a 'security' class to AMI which outputs the required fields for
913 security messages similar to the log messages from res_security_log
915 * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
916 that describes the status value in a human readable string.
918 CDR (Call Detail Records)
920 * Significant changes have been made to the behavior of CDRs. The CDR engine
921 was effectively rewritten and built on the Stasis message bus. For a full
922 definition of CDR behavior in Asterisk 12, please read the specification
923 on the Asterisk wiki (wiki.asterisk.org).
925 * CDRs will now be created between all participants in a bridge. For each
926 pair of channels in a bridge, a CDR is created to represent the path of
927 communication between those two endpoints. This lets an end user choose who
928 to bill for what during bridge operations with multiple parties.
930 * The duration, billsec, start, answer, and end times now reflect the times
931 associated with the current CDR for the channel, as opposed to a cumulative
932 measurement of all CDRs for that channel.
934 * When a CDR is dispatched, user defined CDR variables from both parties are
935 included in the resulting CDR. If both parties have the same variable, only
936 the Party A value is provided.
938 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
939 information regarding the CDR engine is logged as verbose messages. This
940 option should only be used if the behavior of the CDR engine needs to be
943 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
944 normally configured in cdr.conf.
946 * Added CLI command 'cdr show active {channel}'. When {channel} is not
947 specified, this command provides a summary of the channels with CDR
948 information and their statistics. When {channel} is specified, it shows
949 detailed information about all records associated with {channel}.
951 CEL (Channel Event Logging)
953 * CEL has undergone significant rework in Asterisk 12, and is now built on the
954 Stasis message bus. Please see the specification for CEL on the Asterisk
955 wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
958 * The 'extra' field of all CEL events that use it now consists of a JSON blob
959 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
961 * BLINDTRANSFER events now report the transferee bridge unique
962 identifier, extension, and context in a JSON blob as the extra string
963 instead of the transferee channel name as the peer.
965 * ATTENDEDTRANSFER events now report the peer as NULL and additional
966 information in the 'extra' string as a JSON blob. For transfers that occur
967 between two bridged channels, the 'extra' JSON blob contains the primary
968 bridge unique identifier, the secondary channel name, and the secondary
969 bridge unique identifier. For transfers that occur between a bridged channel
970 and a channel running an app, the 'extra' JSON blob contains the primary
971 bridge unique identifier, the secondary channel name, and the app name.
973 * LOCAL_OPTIMIZE events have been added to convey local channel
974 optimizations with the record occurring for the semi-one channel and
975 the semi-two channel name in the peer field.
977 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
978 CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
979 events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
980 and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
981 regardless of whether or not that bridge happens to contain multiple
986 * When compiled with '--enable-dev-mode', the astobj2 library will now add
987 several CLI commands that allow for inspection of ao2 containers that
988 register themselves with astobj2. The CLI commands are 'astobj2 container
989 dump', 'astobj2 container stats', and 'astobj2 container check'.
991 * Added specific CLI commands for bridge inspection. This includes 'bridge
992 show all', which lists all bridges in the system, and 'bridge show {id}',
993 which provides specific information about a bridge.
995 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
996 ejecting the channels currently in the bridge. If the channels cannot
997 continue in the dialplan or application that put them in the bridge, they
1000 * Added command 'bridge kick'. This will eject a single channel from a bridge.
1002 * Added commands to inspect and manipulate the registered bridge technologies.
1003 This include 'bridge technology show', which lists the registered bridge
1004 technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
1005 which controls whether or not a registered bridge technology can be used
1006 during smart bridge operations. If a technology is suspended, it will not
1007 be used when a bridge technology is picked for channels; when unsuspended,
1008 it can be used again.
1010 * The command 'config show help {module} {type} {option}' will show
1011 configuration documentation for modules with XML configuration
1012 documentation. When {module}, {type}, and {option} are omitted, a listing
1013 of all modules with registered documentation is displayed. When {module}
1014 is specified, a listing of all configuration types for that module is
1015 displayed, along with their synopsis. When {module} and {type} are
1016 specified, a listing of all configuration options for that type are
1017 displayed along with their synopsis. When {module}, {type}, and {option}
1018 are specified, detailed information for that configuration option is
1021 * Added 'core show sounds' and 'core show sound' CLI commands. These display
1022 a listing of all installed media sounds available on the system and
1023 detailed information about a sound, respectively.
1025 * 'xmldoc dump' has been added. This CLI command will dump the XML
1026 documentation DOM as a string to the specified file. The Asterisk core
1027 will populate certain XML elements pulled from the source files with
1028 additional run-time information; this command lets a user produce the
1029 XML documentation with all information.
1033 * Parking has been pulled from core and placed into a separate module called
1034 res_parking. See Parking changes below for more details. Configuration for
1035 parking should now be performed in res_parking.conf. Configuration for
1036 parking in features.conf is now unsupported.
1038 * Core attended transfers now have several new options. While performing an
1039 attended transfer, the transferer now has the following options:
1040 - *1 - cancel the attended transfer (configurable via atxferabort)
1041 - *2 - complete the attended transfer, dropping out of the call
1042 (configurable via atxfercomplete)
1043 - *3 - complete the attended transfer, but stay in the call. This will turn
1044 the call into a multi-party bridge (configurable via atxferthreeway)
1045 - *4 - swap to the other party. Once an attended transfer has begun, this
1046 options may be used multiple times (configurable via atxferswap)
1048 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
1049 must be on the channel initiating the transfer to have any effect.
1051 * The BRIDGE_FEATURES channel variable would previously only set features for
1052 the calling party and would set this feature regardless of whether the
1053 feature was in caps or in lowercase. Use of a caps feature for a letter
1054 will now apply the feature to the calling party while use of a lowercase
1055 letter will apply that feature to the called party.
1057 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
1059 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
1060 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
1061 activated the dynamic feature.
1063 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
1064 only on the channel executing the dynamic feature. Executing a dynamic
1065 feature on the bridge peer in a multi-party bridge will execute it on all
1066 peers of the activating channel.
1068 * You can now have the settings for a channel updated using the FEATURE()
1069 and FEATUREMAP() functions inherited to child channels by setting
1070 FEATURE(inherit)=yes.
1072 * automixmon now supports additional channel variables from automon including:
1073 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
1074 and TOUCH_MIXMONITOR_MESSAGE_STOP
1076 * A new general features.conf option 'recordingfailsound' has been added which
1077 allowssetting a failure sound for a user tries to invoke a recording feature
1078 such as automon or automixmon and it fails.
1080 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
1081 features.c for atxferdropcall=no to work properly. This option now just
1086 * Added log rotation strategy 'none'. If set, no log rotation strategy will
1087 be used. Given that this can cause the Asterisk log files to grow quickly,
1088 this option should only be used if an external mechanism for log management
1093 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
1094 will store the path information for that peer when it registers. Realtime
1095 tables can also use the 'supportpath' field to enable Path header support.
1097 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
1098 objectIdentifier. This maps to the supportpath option in sip.conf.
1102 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
1103 provides modules a useful abstraction on top of the many storage mechanisms
1104 in Asterisk, including the Asterisk Database, static configuration files,
1105 static Realtime, and dynamic Realtime. It also provides a caching service.
1106 Users can configure a hierarchy of data storage layers for specific modules
1109 * All future modules which utilize Sorcery for object persistence must have a
1110 column named "id" within their schema when using the Sorcery realtime module.
1111 This column must be able to contain a string of up to 128 characters in length.
1113 Security Events Framework
1115 * Security Event timestamps now use ISO 8601 formatted date/time instead of
1116 the "seconds-microseconds" format that it was using previously.
1120 * The Stasis message bus is a publish/subscribe message bus internal to
1121 Asterisk. Many services in Asterisk are built on the Stasis message bus,
1122 including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
1123 Stasis can be configured in stasis.conf. Note that these parameters operate
1124 at a very low level in Asterisk, and generally will not require changes.
1128 * When a channel driver is configured to enable jiterbuffers, they are now
1129 applied unconditionally when a channel joins a bridge. If a jitterbuffer
1130 is already set for that channel when it enters, such as by the JITTERBUFFER
1131 function, then the existing jitterbuffer will be used and the one set by
1132 the channel driver will not be applied.
1136 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
1137 dialplan applications provided by the app_agent_pool module. Agents are
1138 connected with callers using the new AgentRequest dialplan application.
1139 The Agents:<agent-id> device state is available to monitor the status of an
1140 agent. See agents.conf.sample for valid configuration options.
1142 * The updatecdr option has been removed. Altering the names of channels on a
1143 CDR is not supported - the name of the channel is the name of the channel,
1144 and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
1145 has also been removed, for the same reason.
1147 * The endcall and enddtmf configuration options are removed. Use the
1148 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
1149 channel before calling AgentLogin.
1153 * chan_bridge has been removed. Its functionality has been incorporated
1154 directly into the ConfBridge application itself.
1158 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
1159 of the specified span and its B-channels. Note that this command should
1160 only be used if you understand the risks it entails.
1162 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
1163 A range of channels can be specified to be destroyed. Note that this command
1164 should only be used if you understand the risks it entails.
1166 * Added the CLI command 'dahdi create channels'. A range of channels can be
1167 specified to be created, or the keyword 'new' can be used to add channels
1170 * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
1171 the exact configured mailbox name. For app_voicemail mailboxes this is
1174 * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
1178 * IPv6 support has been added. We are now able to bind to and
1179 communicate using IPv6 addresses.
1183 * The /b option has been removed.
1185 * chan_local moved into the system core and is no longer a loadable module.
1189 * Added general support for busy detection.
1191 * Added ECAM command support for Sony Ericsson phones.
1195 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
1196 SIP stack. A collection of resource modules provides the bulk of the SIP
1197 functionality. For more information on the new SIP channel driver, see
1198 https://wiki.asterisk.org/wiki/x/JYGLAQ
1202 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
1203 using the 'supportpath' setting, either on a global basis or on a peer basis.
1204 This setting enables Asterisk to route outgoing out-of-dialog requests via a
1205 set of proxies by using a pre-loaded route-set defined by the Path headers in
1206 the REGISTER request. See Realtime updates for more configuration information.
1208 * The SIP_CODEC family of variables may now specify more than one codec. Each
1209 codec must be separated by a comma. The first codec specified is the
1210 preferred codec for the offer. This allows a dialplan writer to specify both
1211 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
1213 * The 'callevents' parameter has been removed. Hold AMI events are now raised
1214 in the core, and can be filtered out using the 'eventfilter' parameter
1217 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
1218 codecs configured for a peer instead of the requested codec.
1220 * The option "register_retry_403" has been added to chan_sip to work around
1221 servers that are known to erroneously send 403 in response to valid
1222 REGISTER requests and allows Asterisk to continue attepmting to connect.
1226 * Added the 'immeddialkey' parameter. If set, when the user presses the
1227 configured key the already entered number will be immediately dialed. This
1228 is useful when the dialplan allows for variable length pattern matching.
1229 Valid options are '*' and '#'.
1231 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
1232 milliseconds) before a call forward is considered to not be answered.
1234 * The 'serviceurl' parameter allows Service URLs to be attached to line
1243 * The password option has been disabled, as the AgentLogin application no
1244 longer provides authentication.
1248 * Due to changes in the Asterisk core, this function is no longer needed to
1249 preserve a MixMonitor on a channel during transfer operations and dialplan
1250 execution. It is effectively obsolete.
1254 * The 'amaflags' and 'accountcode' attributes for the CDR function are
1255 deprecated. Use the CHANNEL function instead to access these attributes.
1257 * The 'l' option has been removed. When reading a CDR attribute, the most
1258 recent record is always used. When writing a CDR attribute, all non-finalized
1261 * The 'r' option has been removed, for the same reason as the 'l' option.
1263 * The 's' option has been removed, as LOCKED semantics no longer exist in the
1268 * A new function CDR_PROP has been added. This function lets you set properties
1269 on a channel's active CDRs. This function is write-only. Properties accept
1270 boolean values to set/clear them on the channel's CDRs. Valid properties
1272 - 'party_a' - make this channel the preferred Party A in any CDR between two
1273 channels. If two channels have this property set, the creation time of the
1274 channel is used to determine who is Party A. Note that dialed channels are
1275 never Party A in a CDR.
1276 - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
1277 application when set to True, and analogous to the 'e' option in ResetCDR
1282 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
1283 enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
1284 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
1287 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
1288 string, i.e., [[context],extension],priority. If set on a channel, if a
1289 channel leaves a bridge but is not hung up it will resume dialplan execution
1294 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
1295 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
1296 The value of this setting is ignored when disabled is used for the argument.
1300 * A new function provided by chan_pjsip, this function can be used in
1301 conjunction with the Dial application to construct a dial string that will
1302 dial all contacts on an Address of Record associated with a chan_pjsip
1307 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
1308 outbound channel prior to dialing.
1312 * Redirecting reasons can now be set to arbitrary strings. This means
1313 that the REDIRECTING dialplan function can be used to set the redirecting
1314 reason to any string. It also allows for custom strings to be read as the
1315 redirecting reason from SIP Diversion headers.
1319 * The SPEECH_ENGINE function now supports read operations. When read from, it
1320 will return the current value of the requested attribute.
1324 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
1325 system as mailbox@context. The rest of the system cannot add @default
1326 to mailbox identifiers for app_voicemail that do not specify a context
1327 any longer. It is a mailbox identifier format that should only be
1328 interpreted by app_voicemail.
1334 res_agi (Asterisk Gateway Interface)
1336 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
1338 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
1341 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
1342 will start the playback of the audio at the position specified. It will
1343 also return the final position of the file in 'endpos'.
1345 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
1346 channel variable if the user stopped the file playback or if a remote
1347 entity stopped the playback. If neither stopped the playback, it will
1348 indicate the overall success/failure of the playback. If stopped early,
1349 the final offset of the file will be set in the CPLAYBACKOFFSET channel
1352 * The SAY ALPHA command now accepts an additional parameter to control
1353 whether it specifies the case of uppercase, lowercase, or all letters to
1354 provide functionality similar to SayAlphaCase.
1356 res_ari (Asterisk RESTful Interface) (and others)
1358 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
1359 control telephony primitives in Asterisk by remote client. This includes
1360 channels, bridges, endpoints, media, and other fundamental concepts. Users
1361 of ARI can develop their own communications applications, controlling
1362 multiple channels using an HTTP RESTful interface and receiving JSON events
1363 about the objects via a WebSocket connection. ARI can be configured in
1364 Asterisk via ari.conf. For more information on ARI, see
1365 https://wiki.asterisk.org/wiki/x/0YCLAQ
1369 * Parking has been extracted from the Asterisk core as a loadable module,
1370 res_parking. Configuration for parking is now provided by res_parking.conf.
1371 Configuration through features.conf is no longer supported.
1373 * res_parking uses the configuration framework. If an invalid configuration is
1374 supplied, res_parking will fail to load or fail to reload. Previously,
1375 invalid configurations would generally be accepted, with certain errors
1376 resulting in individually disabled parking lots.
1378 * Parked calls are now placed in bridges. While this is largely an
1379 architectural change, it does have implications on how channels in a parking
1380 lot are viewed. For example, commands that display channels in bridges will
1381 now also display the channels in a parking lot.
1383 * The order of arguments for the new parking applications have been modified.
1384 Timeout and return context/exten/priority are now implemented as options,
1385 while the name of the parking lot is now the first parameter. See the
1386 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
1387 in-depth information as well as syntax.
1389 * Extensions are by default no longer automatically created in the dialplan to
1390 park calls or pickup parked calls. Generation of dialplan extensions can be
1391 enabled using the 'parkext' configuration option.
1393 * ADSI functionality for parking is no longer supported. The 'adsipark'
1394 configuration option has been removed as a result.
1396 * The PARKINGSLOT channel variable has been deprecated in favor of
1397 PARKING_SPACE to match the naming scheme of the new system.
1399 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
1400 channel even when the configuration option 'comebactoorigin' is enabled.
1402 * A new CLI command 'parking show' has been added. This allows a user to
1403 inspect the parking lots that are currently in use.
1404 'parking show <parkinglot>' will also show the parked calls in a specific
1407 * The CLI command 'parkedcalls' is now deprecated in favor of
1408 'parking show <parkinglot>'.
1410 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
1411 can be used to get a list of parked calls for a specific parking lot.
1413 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
1414 with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
1415 specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
1416 longer a required argument.
1418 * The ParkAndAnnounce application is now provided through res_parking instead
1419 of through the separate app_parkandannounce module.
1421 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
1422 by default. Instead, it will follow the timeout rules of the parking lot. The
1423 old behavior can be reproduced by using the 'c' option.
1425 * Dynamic parking lots will now fail to be created under the following
1427 - if the parking lot specified by PARKINGDYNAMIC does not exist
1428 - if they require exclusive park and parkedcall extensions which overlap
1429 with existing parking lots.
1431 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
1432 currently contain no calls. Dynamic parking lots containing parked calls
1433 will persist through the reloads without alteration.
1435 * If 'parkext_exclusive' is set for a parking lot and that extension is
1436 already in use when that parking lot tries to register it, this is now
1437 considered a parking system configuration error. Configurations which do
1438 this will be rejected.
1440 * Added channel variable PARKER_FLAT. This contains the name of the extension
1441 that would be used if 'comebacktoorigin' is enabled. This can be useful when
1442 comebacktoorigin is disabled, but the dialplan or an external control
1443 mechanism wants to use the extension in the park-dial context that was
1444 generated to re-dial the parker on timeout.
1446 res_pjsip (and many others)
1448 * A large number of resource modules make up the SIP stack based on pjsip.
1449 The chan_pjsip channel driver users these resource modules to provide
1450 various SIP functionality in Asterisk. The majority of configuration for
1451 these modules is performed in pjsip.conf. Other modules may use their
1452 own configuration files.
1454 * Added 'set_var' option for an endpoint. For each variable specified that
1455 variable gets set upon creation of a channel involving the endpoint.
1459 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
1460 them, an Asterisk-specific version of PJSIP needs to be installed.
1461 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
1463 res_statsd/res_chan_stats
1465 * A new resource module, res_statsd, has been added, which acts as a statsd
1466 client. This module allows Asterisk to publish statistics to a statsd
1467 server. In conjunction with res_chan_stats, it will publish statistics about
1468 channels to the statsd server. It can be configured via res_statsd.conf.
1472 * Device state for XMPP buddies is now available using the following format:
1473 XMPP/<client name>/<buddy address>
1474 If any resource is available the device state is considered to be not in use.
1475 If no resources exist or all are unavailable the device state is considered
1482 Realtime/Database Scripts
1484 * Asterisk previously included example db schemas in the contrib/realtime/
1485 directory of the source tree. This has been replaced by a set of database
1486 migrations using the Alembic framework. This allows you to use alembic to
1487 initialize the database for you. It will also serve as a database migration
1488 tool when upgrading Asterisk in the future.
1490 See contrib/ast-db-manage/README.md for more details.
1494 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
1495 This python script will convert an existing sip.conf file to a
1496 pjsip.conf file, for use with the chan_pjsip channel driver. This script
1497 is meant to be an aid in converting an existing chan_sip configuration to
1498 a chan_pjsip configuration, but it is expected that configuration beyond
1499 what the script provides will be needed.
1502 ------------------------------------------------------------------------------
1503 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
1504 ------------------------------------------------------------------------------
1508 * The Asterisk build system will now build and install a shared library
1509 (libasteriskssl.so) used to wrap various initialization and shutdown functions
1510 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
1511 that Asterisk can ensure that these functions do *not* get called by any
1512 modules that are loaded into Asterisk, since they should only be called once
1513 in any single process. If desired, this feature can be disabled by supplying
1514 the "--disable-asteriskssl" option to the configure script.
1516 * A new make target, 'full', has been added to the Makefile. This performs
1517 the same compilation actions as make all, but will also scan the entirety of
1518 each source file for documentation. This option is needed to generate AMI
1519 event documentation. Note that your system must have Python in order for
1520 this make target to succeed.
1522 * The optimization portion of the build system has been reworked to avoid
1523 broken builds on certain architectures. All architecture-specific
1524 optimization has been removed in favor of using -march=native to allow gcc
1525 to detect the environment in which it is running when possible. This can
1526 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
1528 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
1529 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
1531 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
1532 previously parsed the header file to obtain the version of Asterisk, you
1533 will now have to go through Asterisk to get the version information.
1541 * Added 'F()' option. Similar to the dial option, this can be supplied with
1542 arguments indicating where the callee should go after the caller is hung up,
1543 or without options specified, the priority after the Queue will be used.
1548 * Added menu action admin_toggle_mute_participants. This will mute / unmute
1549 all non-admin participants on a conference. The confbridge configuration
1550 file also allows for the default sounds played to all conference users when
1551 this occurs to be overriden using sound_participants_unmuted and
1552 sound_participants_muted.
1554 * Added menu action participant_count. This will playback the number of
1555 current participants in a conference.
1557 * Added announcement configuration option to user profile. If set the sound
1558 file will be played to the user, and only the user, upon joining the
1561 * Added record_file_append option that defaults to "yes", but if set to no
1562 will create a new file between each start/stop recording.
1567 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
1568 channels respectively before the callee channels are called.
1573 * Added support for IPv6.
1575 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
1576 external process will cause the current playlist to be cleared, including
1577 stopping any audio file that is currently playing. This is useful when you
1578 want to interrupt audio playback only when specific DTMF is entered by the
1584 * A new option, 'I' has been added to app_followme. By setting this option,
1585 Asterisk will not update the caller with connected line changes when they
1586 occur. This is similar to app_dial and app_queue.
1588 * The 'N' option is now ignored if the call is already answered.
1590 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
1591 and caller channels respectively before the callee channels are called.
1593 * The winning FollowMe outgoing call is now put on hold if the caller put it on
1599 * MixMonitor hooks now have IDs associated with them which can be used to
1600 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
1601 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
1602 now accepts that ID as an argument.
1604 * Added 'm' option, which stores a copy of the recording as a voicemail in the
1605 indicated mailboxes.
1610 * The connect action in app_mysql now allows you to specify a port number to
1611 connect to. This is useful if you run a MySQL server on a non-standard
1617 * Increased the default number of allowed destinations from 5 to 12.
1622 * The app_page application now no longer depends on DAHDI or app_meetme. It
1623 has been re-architected to use app_confbridge internally.
1628 * Added queue options autopausebusy and autopauseunavail for automatically
1629 pausing a queue member when their device reports busy or congestion.
1631 * The 'ignorebusy' option for queue members has been deprecated in favor of
1632 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
1633 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
1634 per interface basis. Individual ringinuse values can now be set in
1635 queues.conf via an argument to member definitions. Lastly, the queue
1636 'ringinuse' setting now only determines defaults for the per member
1637 'ringinuse' setting and does not override per member settings like it does
1638 in earlier versions.
1640 * Added 'F()' option. Similar to the dial option, this can be supplied with
1641 arguments indicating where the callee should go after the caller is hung up,
1642 or without options specified, the priority after the Queue will be used.
1644 * Added new option log_member_name_as_agent, which will cause the membername to
1645 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
1646 state_interface has been set.
1648 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
1650 * App_queue will now play periodic announcements for the caller that
1651 holds the first position in the queue while waiting for answer.
1655 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
1656 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
1657 changed arguments to SayUnixTime so that every option is truly optional even
1658 when using multiple options (so that j option could be used without having to
1659 manually specify timezone and format) There are other benefits, e.g., format
1660 can now be used without specifying time zone as well.
1665 * Addition of the VM_INFO function - see Function changes.
1667 * The imapserver, imapport, and imapflags configuration options can now be
1668 overriden on a user by user basis.
1670 * When voicemail plays a message's envelope with saycid set to yes, when
1671 reaching the caller id field it will play a recording of a file with the same
1672 base name as the sender's callerid if there is a similarly named file in
1673 <astspooldir>/recordings/callerids/
1675 * Voicemails now contains a unique message identifier "msg_id", which is stored
1676 in the message envelope with the sound files. IMAP backends will now store
1677 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
1678 backends will store the message identifier in a "msg_id" column. See
1679 UPGRADE.txt for more information.
1681 * Added VoiceMailPlayMsg application. This application will play a single
1682 voicemail message from a mailbox. The result of the application, SUCCESS or
1683 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
1688 * Hangup handlers can be attached to channels using the CHANNEL() function.
1689 Hangup handlers will run when the channel is hung up similar to the h
1690 extension. The hangup_handler_push option will push a GoSub compatible
1691 location in the dialplan onto the channel's hangup handler stack. The
1692 hangup_handler_pop option will remove the last added location, and optionally
1693 replace it with a new GoSub compatible location. The hangup_handler_wipe
1694 option will remove all locations on the stack, and optionally add a new
1697 * The expression parser now recognizes the ABS() absolute value function,
1698 which will convert negative floating point values to positive values.
1700 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
1701 control of faxdetect.
1703 * Addition of the VM_INFO function that can be used to retrieve voicemail
1704 user information, such as the email address and full name.
1705 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
1708 * The REDIRECTING function now supports the redirecting original party id
1711 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
1712 lets you set some of the configuration options from the [general] section
1713 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
1714 the key sequence used to activate built-in features, such as blindxfer,
1715 and automon. See the built-in documentation for details.
1717 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
1718 instead of simply the uri. This is the format that MessageSend() can use
1719 in the from parameter for outgoing SIP messages.
1721 * Added the PRESENCE_STATE function. This allows retrieving presence state
1722 information from any presence state provider. It also allows setting
1723 presence state information from a CustomPresence presence state provider.
1724 See AMI/CLI changes for related commands.
1726 * Added the AMI_CLIENT function to make manager account attributes available
1727 to the dialplan. It currently supports returning the current number of
1728 active sessions for a given account.
1730 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
1731 and the REDIRECTING functions.
1739 * Added a manager event "LocalBridge" for local channel call bridges between
1740 the two pseudo-channels created.
1745 * Added dialtone_detect option for analog ports to disconnect incoming
1746 calls when dialtone is detected.
1748 * Added option colp_send to send ISDN connected line information. Allowed
1749 settings are block, to not send any connected line information; connect, to
1750 send connected line information on initial connect; and update, to send
1751 information on any update during a call. Default is update.
1753 * Add options namedcallgroup and namedpickupgroup to support installations
1754 where a higher number of groups (>64) is required.
1756 * Added support to use private party ID information with PRI calls.
1761 * A new channel driver named chan_motif has been added which provides support for
1762 Google Talk and Jingle in a single channel driver. This new channel driver includes
1763 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
1764 hold, unhold, and ringing notification. It is also compliant with the current Jingle
1765 specification, current Google Jingle specification, and the original Google Talk
1771 * Added NAT support for RTP. Setting in config is 'nat', which can be set
1772 globally and overriden on a peer by peer basis.
1774 * Direct media functionality has been added. Options in config are:
1775 directmedia (directrtp) and directrtpsetup (earlydirect)
1777 * ChannelUpdate events now contain a CallRef header.
1782 * Asterisk will no longer substitute CID number for CID name in the display
1783 name field if CID number exists without a CID name. This change improves
1784 compatibility with certain device features such as Avaya IP500's directory
1787 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
1788 created using that setting to not be removed during SIP reload.
1790 * Added settings recordonfeature and recordofffeature. When receiving an INFO
1791 request with a "Record:" header, this will turn the requested feature on/off.
1792 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
1793 dynamic features must be enabled and configured properly on the requesting
1794 channel for this to function properly.
1796 * Add support to realtime for the 'callbackextension' option.
1798 * When multiple peers exist with the same address, but differing
1799 callbackextension options, incoming requests that are matched by address
1800 will be matched to the peer with the matching callbackextension if it is
1803 * Two new NAT options, auto_force_rport and auto_comedia, have been added
1804 which set the force_rport and comedia options automatically if Asterisk
1805 detects that an incoming SIP request crossed a NAT after being sent by
1806 the remote endpoint.
1808 * The default global nat setting in sip.conf has been changed from force_rport
1809 to auto_force_rport.
1811 * NAT settings are now a combinable list of options. The equivalent of the
1812 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
1814 * Adds an option send_diversion which can be disabled to prevent
1815 diversion headers from automatically being added to INVITE requests.
1817 * Add support for lightweight NAT keepalive. If enabled a blank packet will
1818 be sent to the remote host at a given interval to keep the NAT mapping open.
1819 This can be enabled using the keepalive configuration option.
1821 * Add option 'tonezone' to specify country code for indications. This option
1822 can be set both globally and overridden for specific peers.
1824 * The SIP Security Events Framework now supports IPv6.
1826 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
1827 between multiple user agents. When set, for directmedia reinvites,
1828 Asterisk will not send an immediate reinvite on an incoming call leg. This
1829 option is useful when peered with another SIP user agent that is known to
1830 send immediate direct media reinvites upon call establishment.
1832 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
1835 * Add options subminexpiry and submaxexpiry to set limits of subscription
1836 timer independently from registration timer settings. The setting of the
1837 registration timer limits still is done by options minexpiry, maxexpiry
1838 and defaultexpiry. For backwards compatibility the setting of minexpiry
1839 and maxexpiry also is used to configure the subscription timer limits if
1840 subminexpiry and submaxexpiry are not set in sip.conf.
1842 * Set registration timer limits to default values when reloading sip
1843 configuration and values are not set by configuration.
1845 * Add options namedcallgroup and namedpickupgroup to support installations
1846 where a higher number of groups (>64) is required.
1848 * When a MESSAGE request is received, the address the request was received from
1849 is now saved in the SIP_RECVADDR variable.
1851 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
1852 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
1853 the ANI2/OLI information is set on the channel, which can be retrieved using
1854 the CALLERID function.
1856 * Peers can now be configured to support negotiation of ICE candidates using
1857 the setting icesupport. See res_rtp_asterisk changes for more information.
1859 * Added support for format attribute negotiation. See the Codecs changes for
1862 * Extra headers specified with SIPAddHeader are sent with the REFER message
1863 when using Transfer application. See refer_addheaders in sip.conf.sample.
1865 * Added support to use private party ID information with calls.
1867 * Adds an option discard_remote_hold_retrieval that when set stops telling
1868 the peer to start music on hold.
1873 * Added skinny version 17 protocol support.
1877 --------------------
1878 * Added ability to use multiple lines for a single phone. This allows multiple
1879 calls to occur on a single phone, using callwaiting and switching between calls.
1881 * Added option 'sharpdial' allowing end dialing by pressing # key
1883 * Added option 'interdigit_timer' to control phone dial timeout
1885 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
1887 * Added global 'debug' option, that enables debug in channel driver
1889 * Added ability to translate on-screen menu in multiple languages. Tested on
1890 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
1891 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
1894 * In addition to English added French and Russian languages for on-screen menus
1896 * Reworked dialing number input: added dialing by timeout, immediate dial on
1897 on dialplan compare, phone number length now not limited by screen size
1899 * Added ability to pickup a call using features.conf defined value and
1905 * Add options namedcallgroup and namedpickupgroup to support installations
1906 where a higher number of groups (>64) is required.
1908 * Added support to use private party ID information with calls.
1913 * The minimum DTMF duration can now be configured in asterisk.conf
1914 as "mindtmfduration". The default value is (as before) set to 80 ms.
1915 (previously it was only available in source code)
1917 * Named ACLs can now be specified in acl.conf and used in configurations that
1918 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
1919 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
1920 working ACL. In addition, some CLI commands have been added to provide
1921 show information and allow for module reloading - see CLI Changes.
1923 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
1924 items (separated by commas), and items in the rule can be negated by prefixing
1925 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
1926 longer necessray to control the order that the 'permit' and 'deny' columns are
1927 returned from queries.
1929 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
1930 be used within the dynamic weight attribute when specifying a mapping.
1932 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
1933 header, instead of putting the user defined event name there. When enabled
1934 the UserDefType header is added for user defined events. This feature is
1935 enabled with the setting show_user_defined.
1937 * Macro has been deprecated in favor of GoSub. For redirecting and connected
1938 line purposes use the following variables instead of their macro equivalents:
1939 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
1940 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
1941 cc_callback_macro in channel configurations.
1943 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
1946 * Call files now support the "early_media" option to connect with an outgoing
1947 extension when early media is received.
1949 * Added support to use private party ID information with calls.
1954 * A new channel variable, AGIEXITONHANGUP, has been added which allows
1955 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
1956 AGI application would exit immediately after a channel hangup is detected.
1958 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
1959 are resolved and each address is attempted in turn until one succeeds or
1963 AMI (Asterisk Manager Interface)
1965 * The originate action now has an option "EarlyMedia" that enables the
1966 call to bridge when we get early media in the call. Previously,
1967 early media was disregarded always when originating calls using AMI.
1969 * Added setvar= option to manager accounts (much like sip.conf)
1971 * Originate now generates an error response if the extension given is not found
1974 * MixMonitor will now show IDs associated with the mixmonitor upon creating
1975 them if the i(variable) option is used. StopMixMonitor will accept
1976 MixMonitorID as an option to close specific MixMonitors.
1978 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
1979 updated to include information about peers configured with
1980 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
1981 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
1982 returned if auto_force_rport is not enabled.
1984 * Added SIPpeerstatus manager command which will generate PeerStatus events
1985 similar to the existing PeerStatus events found in chan_sip on demand.
1987 * Hangup now can take a regular expression as the Channel option. If you want
1988 to hangup multiple channels, use /regex/ as the Channel option. Existing
1989 behavior to hanging up a single channel is unchanged, but if you pass a regex,
1990 the manager will send you a list of channels back that were hung up.
1992 * Support for IPv6 addresses has been added.
1994 * AMI Events can now be documented in the Asterisk source. Note that AMI event
1995 documentation is only generated when Asterisk is compiled using 'make full'.
1996 See the CLI section for commands to display AMI event information.
1998 * The AMI Hangup event now includes the AccountCode header so you can easily
1999 correlate with AMI Newchannel events.
2001 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
2002 the StateInterface of the queue member.
2004 * Added AMI event SessionTimeout in the Call category that is issued when a
2005 call is terminated due to either RTP stream inactivity or SIP session timer
2008 * CEL events can now contain a user defined header UserDefType. See core
2009 changes for more information.
2011 * OOH323 ChannelUpdate events now contain a CallRef header.
2013 * Added PresenceState command. This command will report the presence state for
2014 the given presence provider.
2016 * Added Parkinglots command. This will list all parking lots as a series of
2017 AMI Parkinglot events.
2019 * Added MessageSend command. This behaves in the same manner as the
2020 MessageSend application, and is a technolgoy agnostic mechanism to send out
2021 of call text messages.
2023 * Added "message" class authorization. This grants an account permission to
2024 send out of call messages. Write-only.
2029 * The "dialplan add include" command has been modified to create context a context
2030 if one does not already exist. For instance, "dialplan add include foo into bar"
2031 will create context "bar" if it does not already exist.
2033 * A "dialplan remove context" command has been added to remove a context from
2036 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
2037 filenames of all running mixmonitors on a channel.
2039 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
2040 numeric instead of 0, 1, or 2.
2042 * "stun show status" will show a table describing how the STUN client is
2045 * "acl show [named acl]" will show information regarding a Named ACL. The
2046 acl module can be reloaded with "reload acl".
2048 * Added CLI command to display AMI event information - "manager show events",
2049 which shows a list of all known and documented AMI events, and "manager show
2050 event [event name]", which shows detail information about a specific AMI
2053 * The result of the CLI command "queue show" now includes the state interface
2054 information of the queue member.
2056 * The command "core set verbose" will now set a separate level of logging for
2057 each remote console without affecting any other console.
2059 * Added command "cdr show pgsql status" to check connection status
2061 * "sip show channel" will now display the complete route set.
2063 * Added "presencestate list" command. This command will list all custom
2064 presence states that have been set by using the PRESENCE_STATE dialplan
2067 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
2068 command. This changes a custom presence to a new state.
2073 * Codec lists may now be modified by the '!' character, to allow succinct
2074 specification of a list of codecs allowed and disallowed, without the
2075 requirement to use two different keywords. For example, to specify all
2076 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
2078 * Add support for parsing SDP attributes, generating SDP attributes, and
2079 passing it through. This support includes codecs such as H.263, H.264, SILK,
2080 and CELT. You are able to set up a call and have attribute information pass.
2081 This should help considerably with video calls.
2083 * The iLBC codec can now use a system-provided iLBC library if one is installed,
2084 just like the GSM codec.
2088 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
2089 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
2093 * Asterisk version and build information is now logged at the beginning of a
2096 * Threads belonging to a particular call are now linked with callids which get
2097 added to any log messages produced by those threads. Log messages can now be
2098 easily identified as involved with a certain call by looking at their call id.
2099 Call ids may also be attached to log messages for just about any case where
2100 it can be determined to be related to a particular call.
2102 * Each logging destination and console now have an independent notion of the
2103 current verbosity level. Logger.conf now allows an optional argument to
2104 the 'verbose' specifier, indicating the level of verbosity sent to that
2105 particular logging destination. Additionally, remote consoles now each
2106 have their own verbosity level. The command 'core set verbose' will now set
2107 a separate level for each remote console without affecting any other
2113 * Added 'announcement' option which will play at the start of MOH and between
2114 songs in modes of MOH that can detect transitions between songs (eg.
2120 * New per parking lot options: comebackcontext and comebackdialtime. See
2121 configs/features.conf.sample for more details.
2123 * Channel variable PARKER is now set when comebacktoorigin is disabled in
2126 * Channel variable PARKEDCALL is now set with the name of the parking lot
2127 when a timeout occurs.
2133 CDR Postgresql Driver
2135 * Added command "cdr show pgsql status" to check connection status
2138 CDR Adaptive ODBC Driver
2140 * Added schema option for databases that support specifying a schema.
2148 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
2149 CALENDAR_WRITE has completed successfully.
2154 * A new option, 'probation' has been added to rtp.conf
2155 RTP in strictrtp mode can now require more than 1 packet to exit learning
2156 mode with a new source (and by default requires 4). The probation option
2157 allows the user to change the required number of packets in sequence to any
2158 desired value. Use a value of 1 to essentially restore the old behavior.
2159 Also, with strictrtp on, Asterisk will now drop all packets until learning
2160 mode has successfully exited. These changes are based on how pjmedia handles
2161 media sources and source changes.
2163 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
2164 enabled or disabled using the icesupport setting. A variety of other
2165 settings have been introduced to configure STUN/TURN connections.
2170 * A new module, res_corosync, has been introduced. This module uses the
2171 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
2172 of Asterisk servers to both Message Waiting Indication (MWI) and/or
2173 Device State (presence) information. This module is very similar to, and
2174 is a replacement for the res_ais module that was in previous releases of
2180 * This module adds a cleaned up, drop-in replacement for res_jabber called
2181 res_xmpp. This provides the same externally facing functionality but is
2182 implemented differently internally. res_jabber has been deprecated in favor
2183 of res_xmpp; please see the UPGRADE.txt file for more information.
2188 * The safe_asterisk script has been updated to allow several of its parameters
2189 to be set from environment variables. This also enables a custom run
2190 directory of Asterisk to be specified, instead of defaulting to /tmp.
2192 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
2193 its value to determine the directory to assume is the top-level directory of
2194 the source tree. If the variable is not set, it defaults to the current
2195 behavior and uses the current working directory.
2197 ------------------------------------------------------------------------------
2198 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
2199 ------------------------------------------------------------------------------
2203 * Asterisk now has protocol independent support for processing text messages
2204 outside of a call. Messages are routed through the Asterisk dialplan.
2205 SIP MESSAGE and XMPP are currently supported. There are options in
2206 jabber.conf and sip.conf to allow enabling these features.
2207 -> jabber.conf: see the "sendtodialplan" and "context" options.
2208 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
2209 and "outofcall_message_context" options.
2210 The MESSAGE() dialplan function and MessageSend() application have been
2211 added to go along with this functionality. More detailed usage information
2212 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
2213 * If real-time text support (T.140) is negotiated, it will be preferred for
2214 sending text via the SendText application. For example, via SIP, messages
2215 that were once sent via the SIP MESSAGE request would be sent via RTP if
2216 T.140 text is negotiated for a call.
2220 * parkedmusicclass can now be set for non-default parking lots.
2222 Asterisk Manager Interface
2223 --------------------------
2224 * PeerStatus now includes Address and Port.
2225 * Added Hold events for when the remote party puts the call on and off hold
2226 for chan_dahdi ISDN channels.
2227 * Added new action MeetmeListRooms to list active conferences (shows same
2228 data as "meetme list" at the CLI).
2229 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
2230 Description field that is set by 'description' in the channel configuration
2232 * Added Uniqueid header to UserEvent.
2233 * Added new action FilterAdd to control event filters for the current session.
2234 This requires the system permission and uses the same filter syntax as
2235 filters that can be defined in manager.conf
2236 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
2237 versions had some instances of the event converted, but others were left
2238 as-is. All Unlink events should now be converted to Bridge events. The AMI
2239 protocol version number was incremented to 1.2 as a result of this change.
2241 Asterisk HTTP Server
2242 --------------------------
2243 * The HTTP Server can bind to IPv6 addresses.
2246 --------------------------
2247 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
2248 with busydetect. usage example: busypattern=200,200,200,600
2251 --------------------------
2252 * New 'gtalk show settings' command showing the current settings loaded from
2254 * The 'logger reload' command now supports an optional argument, specifying an
2255 alternate configuration file to use.
2256 * 'dialplan add extension' command will now automatically create a context if
2257 the specified context does not exist with a message indicated it did so.
2258 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
2259 Description field which can be populated with 'description' in the channel
2260 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
2263 --------------------------
2264 * The filter option in cdr_adaptive_odbc now supports negating the argument,
2265 thus allowing records which do NOT match the specified filter.
2266 * Added ability to log CONGESTION calls to CDR
2269 --------------------------
2270 * Ability to define custom SILK formats in codecs.conf.
2271 * Addition of speex32 audio format with translation.
2272 * CELT codec pass-through support and ability to define
2273 custom CELT formats in codecs.conf.
2274 * Ability to read raw signed linear files with sample rates
2275 ranging from 8khz - 192khz. The new file extensions introduced
2276 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
2277 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
2278 Skinny, H.323, etc) can still only support the following codecs:
2279 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
2280 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
2281 Video: h261, h263, h263p, h264, mpeg4
2286 --------------------------
2287 * New highly optimized and customizable ConfBridge application capable of
2288 mixing audio at sample rates ranging from 8khz-96khz.
2289 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
2290 and bridge profiles on a channel.
2291 * CONFBRIDGE_INFO dialplan function capable of retrieving information
2292 about a conference such as locked status and number of parties, admins,
2294 * Addition of video_mode option in confbridge.conf for adding video support
2295 into a bridge profile.
2296 * Addition of the follow_talker video_mode in confbridge.conf. This video
2297 mode dynamically switches the video feed to always display the loudest talker
2298 supplying video in the conference.
2302 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
2303 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
2304 variables from asterisk.conf.
2308 * Addition of the JITTERBUFFER dialplan function. This function allows
2309 for jitterbuffering to occur on the read side of a channel. By using
2310 this function conference applications such as ConfBridge and MeetMe can
2311 have the rx streams jitterbuffered before conference mixing occurs.
2312 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
2314 * Added STRREPLACE function. This function let's the user search a variable
2315 for a given string to replace with another string as many times as the
2316 user specifies or just throughout the whole string.
2317 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
2318 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
2319 * Added extensions to chan_ooh323 in function CHANNEL()
2321 libpri channel driver (chan_dahdi) DAHDI changes
2322 --------------------------
2323 * Added moh_signaling option to specify what to do when the channel's bridged
2324 peer puts the ISDN channel on hold.
2325 * Added display_send and display_receive options to control how the display ie
2326 is handled. To send display text from the dialplan use the SendText()
2327 application when the option is enabled.
2328 * Added mcid_send option to allow sending a MCID request on a span.
2331 --------------------------
2332 * Added setvar option to calendar.conf to allow setting channel variables on
2333 notification channels.
2334 * Added "calendar show types" CLI command to list registered calendar
2338 --------------------------
2339 * Added two new options, r and t with file name arguments to record
2340 single direction (unmixed) audio recording separate from the bidirectional
2341 (mixed) recording. The mixed file name argument is optional now as long
2342 as at least one recording option is used.
2345 --------------------------
2346 * Added a new option, l, which will disable local call optimization for
2347 channels involved with the FollowMe thread. Use this option to improve
2348 compatability for a FollowMe call with certain dialplan apps, options, and
2352 --------------------------
2353 * Added option "k" that will automatically close the conference when there's
2354 only one person left when a user exits the conference.
2357 --------------------------
2358 * cel_pgsql now supports the 'extra' column for data added using the
2359 CELGenUserEvent() application.
2362 --------------------------
2363 * Support for defining hints has been added to pbx_lua. See the 'hints' table
2364 in the sample extensions.lua file for syntax details.
2365 * Applications that perform jumps in the dialplan such as Goto will now
2366 execute properly. When pbx_lua detects that the context, extension, or
2367 priority we are executing on has changed it will immediately return control
2368 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
2369 the priority after the currently executing priority.
2370 * An autoservice is now started by default for pbx_lua channels. It can be
2371 stopped and restarted using the autoservice_stop() and autoservice_start()
2375 --------------------------
2376 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
2377 into a FAXStatus event with an 'Operation' header that will be either
2378 'send', 'receive', and 'gateway'.
2379 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
2380 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
2381 feature will handle converting a fax call between an audio T.30 fax terminal
2382 and an IFP T.38 fax terminal.
2386 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
2387 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
2388 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
2392 * Added general option negative_penalty_invalid default off. when set
2393 members are seen as invalid/logged out when there penalty is negative.
2394 for realtime members when set remove from queue will set penalty to -1.
2395 * Added queue option autopausedelay when autopause is enabled it will be
2396 delayed for this number of seconds since last successful call if there
2397 was no prior call the agent will be autopaused immediately.
2398 * Added member option ignorebusy this when set and ringinuse is not
2399 will allow per member control of multiple calls as ringinuse does for
2404 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
2406 * Added 'k' option to MeetMe to automatically kill the conference when there's only
2407 one participant left (much like a normal call bridge)
2408 * Added extra argument to Originate to set timeout.
2412 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
2413 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
2414 utility in the UTILS section of menuselect. If an existing astdb is found and no
2415 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
2416 convert an existing astdb to the SQLite3 version automatically at runtime.
2420 * Modules marked as deprecated are no longer marked as building by default. Enabling
2421 these modules is still available via menuselect.
2425 * authdebug is now disabled by default. To enable this functionaility again
2426 set authdebug = yes in iax.conf.
2430 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
2431 releases it was disabled.
2435 * The PBX core previously made a call with a non-existing extension test for
2436 extension s@default and jump there if the extension existed.
2437 This was a bad default behaviour and violated the principle of least surprise.
2438 It has therefore been changed in this release. It may affect some
2439 applications and configurations that rely on this behaviour. Most channel
2440 drivers have avoided this for many releases by testing whether the extension
2441 called exists before starting the PBX and generating a local error.
2442 This behaviour still exists and works as before.
2444 Extension "s" is used when no extension is given in a channel driver,
2445 like immediate answer in DAHDI or calling to a domain with no user part
2448 ------------------------------------------------------------------------------
2449 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
2450 ------------------------------------------------------------------------------
2454 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
2455 now defaults to force_rport. It is very important that phones requiring nat=no be
2456 specifically set as such instead of relying on the default setting. If at all
2457 possible, all devices should have nat settings configured in the general section as
2458 opposed to configuring nat per-device.
2459 * Added preferred_codec_only option in sip.conf. This feature limits the joint
2460 codecs sent in response to an INVITE to the single most preferred codec.
2461 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
2462 to be used for the outgoing call. It must be one of the codecs configured
2464 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
2465 to be used for holding a private key. If tlsprivatekey is not specified,
2466 tlscertfile is searched for both public and private key.
2467 * Added tlsclientmethod option to sip.conf. This allows the protocol for
2468 outbound client connections to be specified.
2469 * The sendrpid parameter has been expanded to include the options
2470 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
2471 header to be sent (equivalent to setting sendrpid=yes) and setting
2472 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
2473 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
2474 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
2475 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
2476 will accept the SDP even if the SDP version number is not properly incremented,
2477 but will generate a warning in the log indicating that the SIP peer that sent
2478 the SDP should have the 'ignoresdpversion' option set.
2479 * The 'nat' option has now been been changed to have yes, no, force_rport, and
2480 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
2481 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
2482 remote side requests it and disables symmetric RTP support. Setting it to
2483 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
2484 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
2485 and enables symmetric RTP support.
2486 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
2487 response. This permits the master channel to know how each channel dialled
2488 in a multi-channel setup resolved in an individual way. This carries a
2489 performance penalty and can be disabled in sip.conf using the
2490 'storesipcause' option.
2491 * Added 'externtcpport' and 'externtlsport' options to allow custom port
2492 configuration for the externip and externhost options when tcp or tls is used.
2493 * Added support for message body (stored in content variable) to SIP NOTIFY message
2494 accessible via AMI and CLI.
2495 * Added 'media_address' configuration option which can be used to explicitly specify
2496 the IP address to use in the SDP for media (audio, video, and text) streams.
2497 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
2498 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
2500 * Added 'use_q850_reason' configuration option for generating and parsing
2501 if available Reason: Q.850;cause=<cause code> header. It is implemented
2502 in some gateways for better passing PRI/SS7 cause codes via SIP.
2503 * When dialing SIP peers, a new component may be added to the end of the dialstring
2504 to indicate that a specific remote IP address or host should be used when dialing
2505 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
2506 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
2507 ability to selectively force bridged channels to also be encrypted is also
2508 implemented. Branching in the dialplan can be done based on whether or not
2509 a channel has secure media and/or signaling.
2510 * Added directmediapermit/directmediadeny to limit which peers can send direct media
2512 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
2513 Charge messages to snom phones.
2514 * Added support for G.719 media streams.
2515 * Added support for 16khz signed linear media streams.
2516 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
2517 RTP has been outfitted with the same abilities.
2518 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
2519 available in device configurations as well as in the dial plan.
2520 * Addition of the 'subscribe_network_change' option for turning on and off
2521 res_stun_monitor module support in chan_sip.
2522 * Addition of the 'auth_options_requests' option for turning on and off
2523 authentication for OPTIONS requests in chan_sip.
2527 * Add #tryinclude statement for config files. This provides the same
2528 functionality as the #include statement however an asterisk module will
2529 still load if the filename does not exist. Using the #include statement
2530 Asterisk will not allow the module to load.
2534 * Added rtsavesysname option into iax.conf to allow the systname to be saved
2535 on realtime updates.
2536 * Added the ability for chan_iax2 to inform the dialplan whether or not
2537 encryption is being used. This interoperates with the SIP SRTP implementation
2538 so that a secure SIP call can be bridged to a secure IAX call when the
2539 dialplan requires bridged channels to be "secure".
2540 * Addition of the 'subscribe_network_change' option for turning on and off
2541 res_stun_monitor module support in chan_iax.
2546 * Added ability to preset channel variables on indicated lines with the setvar
2547 configuration option. Also, clearvars=all resets the list of variables back
2549 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
2550 See configs/res_pktccops.conf for more information.
2552 XMPP Google Talk/Jingle changes
2553 -------------------------------
2554 * Added the externip option to gtalk.conf.
2555 * Added the stunaddr option to gtalk.conf which allows for the automatic
2556 retrieval of the external ip from a stun server.
2560 * Added 'p' option to PickupChan() to allow for picking up channel by the first
2561 match to a partial channel name.
2562 * Added .m3u support for Mp3Player application.
2563 * Added progress option to the app_dial D() option. When progress DTMF is
2564 present, those values are sent immediately upon receiving a PROGRESS message
2565 regardless if the call has been answered or not.
2566 * Added functionality to the app_dial F() option to continue with execution
2567 at the current location when no parameters are provided.
2568 * Added the 'a' option to app_dial to answer the calling channel before any
2569 announcements or macros are executed.
2570 * Modified app_dial to set answertime when the called channel answers even if
2571 the called channel hangs up during playback of an announcement.
2572 * Modified app_dial 'r' option to support an additional parameter to play an
2573 indication tone from indications.conf
2574 * Added c() option to app_chanspy. This option allows custom DTMF to be set
2575 to cycle through the next available channel. By default this is still '*'.
2576 * Added x() option to app_chanspy. This option allows DTMF to be set to
2577 exit the application.
2578 * The Voicemail application has been improved to automatically ignore messages
2579 that only contain silence.
2580 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
2581 associated mailbox(es) to be greetings-only.
2582 * The ChanSpy application now has the 'S' option, which makes the application
2583 automatically exit once it hits a point where no more channels are available
2585 * The ChanSpy application also now has the 'E' option, which spies on a single
2586 channel and exits when that channel hangs up.
2587 * The MeetMe application now turns on the DENOISE() function by default, for
2588 each participant. In our tests, this has significantly decreased background
2589 noise (especially noisy data centers).
2590 * Voicemail now permits storage of secrets in a separate file, located in the
2591 spool directory of each individual user. The control for this is located in
2592 the "passwordlocation" option in voicemail.conf. Please see the sample
2593 configuration for more information.
2594 * The ChanIsAvail application now exposes the returned cause code using a separate
2595 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
2596 * Added 'd' option to app_followme. This option disables the "Please hold"
2598 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
2599 received will terminate recording.
2600 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
2601 Previously the folder could only be set per context, but has now been extended
2602 using the imapfolder option.
2603 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
2604 * Voicemail now allows the pager date format to be specified separately from the
2606 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
2607 to allow joining, leaving, and sending text to group chats.
2608 * MeetMe has a new option 'G' to play an announcement before joining a conference.
2609 * Page has a new option 'A(x)' which will playback an announcement simultaneously
2610 to all paged phones (and optionally excluding the caller's one using the new
2611 option 'n') before the call is bridged.
2612 * The 'f' option to Dial has been augmented to take an optional argument. If no
2613 argument is provided, the 'f' option works as it always has. If an argument is
2614 provided, then the connected party information of all outgoing channels created
2615 during the Dial will be set to the argument passed to the 'f' option.
2616 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
2618 * The OSP lookup application adds in/outbound network ID, optional security,
2619 number portability, QoS reporting, destination IP port, custom info and service
2621 * Added new application VMSayName that will play the recorded name of the voicemail
2622 user if it exists, otherwise will play the mailbox number.
2623 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
2624 retrieve state for a particular bridge, where <name> is the conference name
2625 * app_directory now allows exiting at any time using the operator or pound key.
2626 * Voicemail now supports setting a locale per-mailbox.
2627 * Two new applications are provided for declining counting phrases in multiple
2628 languages. See the application notes for SayCountedNoun and SayCountedAdj for
2630 * Voicemail now runs the externnotify script when pollmailboxes is activated and
2632 * Voicemail now includes rdnis within msgXXXX.txt file.
2633 * ExternalIVR now supports IPv6 addresses.
2634 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
2635 at https://wiki.asterisk.org/wiki/x/oQBB
2636 * ParkedCall and Park can now specify the parking lot to use.
2640 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
2641 over SRV records associated with a specific service. From the CLI, type
2642 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
2643 details on how these may be used.
2644 * PITCH_SHIFT dialplan function added. This function can be used to modify the
2645 pitch of a channel's tx and rx audio streams.
2646 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
2647 setting various connected line and redirecting party information.
2648 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
2649 support ISDN subaddressing.
2650 * The CHANNEL() function now supports the "name" and "checkhangup" options.
2651 * For DAHDI channels, the CHANNEL() dialplan function now allows
2652 the dialplan to request changes in the configuration of the active
2653 echo canceller on the channel (if any), for the current call only.
2656 exten => s,n,Set(CHANNEL(echocan_mode)=off)
2658 The possible values are:
2660 on - normal mode (the echo canceller is actually reinitialized)
2662 fax - FAX/data mode (NLP disabled if possible, otherwise completely
2664 voice - voice mode (returns from FAX mode, reverting the changes that
2665 were made when FAX mode was requested)
2666 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
2667 and setting variables on the channel which created the current channel.
2668 Administrators should take care to avoid naming conflicts, when multiple
2669 channels are dialled at once, especially when used with the Local channel
2670 construct (which all could set variables on the master channel). Usage
2671 of the HASH() dialplan function, with the key set to the name of the slave
2672 channel, is one approach that will avoid conflicts.
2673 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
2675 * func_odbc now allows multiple row results to be retrieved without using
2676 mode=multirow. If rowlimit is set, then additional rows may be retrieved
2677 from the same query by using the name of the function which retrieved the
2678 first row as an argument to ODBC_FETCH().
2679 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
2680 dialplan. This function returns the content of the received message.
2681 * Added REPLACE, which searches a given variable name for a set of characters,
2682 then either replaces them with a single character or deletes them.
2683 * Added PASSTHRU, which literally passes the same argument back as its return
2684 value. The intent is to be able to use a literal string argument to
2685 functions that currently require a variable name as an argument.
2686 * HASH-associated variables now can be inherited across channel creation, by
2687 prefixing the name of the hash at assignment with the appropriate number of
2688 underscores, just like variables.
2689 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
2690 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
2691 whether or not channels that are bridged to the current channel will be
2692 required to have secure signaling and/or media.
2693 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
2694 the current channel has secure signaling and/or media.
2695 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
2696 "no_media_path" option.
2697 Returns "0" if there is a B channel associated with the call.
2698 Returns "1" if no B channel is associated with the call. The call is either
2699 on hold or is a call waiting call.
2700 * Added option to dialplan function CDR(), the 'f' option
2701 allows for high resolution times for billsec and duration fields.
2702 * FILE() now supports line-mode and writing.
2703 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
2704 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
2708 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
2709 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
2710 and is set when a dynamic feature is triggered.
2711 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
2712 to dynamically create a new parking lot matching the value this varible is
2714 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
2715 features.conf that should be the base for dynamic parkinglots.
2716 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
2717 parkinglot should have.
2718 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
2719 parkinglot should have.
2720 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
2725 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
2726 timeout has expired.
2727 * Added 'R' option to app_queue. This option stops moh and indicates ringing
2728 to the caller when an Agent's phone is ringing. This can be used to indicate
2729 to the caller that their call is about to be picked up, which is nice when
2730 one has been on hold for an extened period of time.
2731 * A new config option, penaltymemberslimit, has been added to queues.conf.
2732 When set this option will disregard penalty settings when a queue has too
2734 * A new option, 'I' has been added to both app_queue and app_dial.
2735 By setting this option, Asterisk will not update the caller with
2736 connected line changes or redirecting party changes when they occur.
2737 * A 'relative-periodic-announce' option has been added to queues.conf. When
2738 enabled, this option will cause periodic announce times to be calculated
2739 from the end of announcements rather than from the beginning.
2740 * The autopause option in queues.conf can be passed a new value, "all." The
2741 result is that if a member becomes auto-paused, he will be paused in all
2742 queues for which he is a member, not just the queue that failed to reach
2744 * Added dialplan function QUEUE_EXISTS to check if a queue exists
2745 * The queue logger now allows events to optionally propagate to a file,
2746 even when realtime logging is turned on. Additionally, realtime logging
2747 supports sending the event arguments to 5 individual fields, although it
2748 will fallback to the previous data definition, if the new table layout is
2751 mISDN channel driver (chan_misdn) changes
2752 ----------------------------------------
2753 * Added display_connected parameter to misdn.conf to put a display string
2754 in the CONNECT message containing the connected name and/or number if
2755 the presentation setting permits it.
2756 * Added display_setup parameter to misdn.conf to put a display string
2757 in the SETUP message containing the caller name and/or number if the
2758 presentation setting permits it.
2759 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
2760 indicate the dialplan settings are to be obtained from the asterisk
2762 * Made misdn.conf parameter callerid accept the "name" <number> format
2763 used by the rest of the system.
2764 * Made use the nationalprefix and internationalprefix misdn.conf
2765 parameters to prefix any received number from the ISDN link if that
2766 number has the corresponding Type-Of-Number. NOTE: This includes
2767 comparing the incoming call's dialed number against the MSN list.
2768 * Added the following new parameters: unknownprefix, netspecificprefix,
2769 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
2770 received number from the ISDN link if that number has the corresponding
2772 * Added new dialplan application misdn_command which permits controlling
2773 the CCBS/CCNR functionality.
2774 * Added new dialplan function mISDN_CC which permits retrieval of various
2775 values from an active call completion record.
2776 * For PTP, you should manually send the COLR of the redirected-to party
2777 for an incomming redirected call if the incoming call could experience
2778 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
2779 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
2780 if the REDIRECTING(from-num) is not empty.
2781 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
2782 option on all of the REDIRECTING statements before dialing the
2783 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
2784 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
2785 redirecting-to presentation (COLR) when it becomes available.
2786 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
2789 thirdparty mISDN enhancements
2790 -----------------------------
2791 mISDN has been modified by Digium, Inc. to greatly expand facility message
2793 * Enhanced COLP support for call diversion and transfer.
2794 * CCBS/CCNR support.
2796 The latest modified mISDN v1.1.x based version is available at:
2797 http://svn.digium.com/svn/thirdparty/mISDN/trunk
2798 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
2800 Tagged versions of the modified mISDN code are available under:
2801 http://svn.digium.com/svn/thirdparty/mISDN/tags
2802 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
2804 libpri channel driver (chan_dahdi) DAHDI changes
2805 -------------------------------------------
2806 * The channel variable PRIREDIRECTREASON is now just a status variable
2807 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
2808 to read and alter the reason.
2809 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
2810 redirected-to party for an incomming redirected call if the incoming call
2811 could experience further redirects. Just set the
2812 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
2813 to the COLR. A call has been redirected if the REDIRECTING(count) is not
2815 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
2816 use the inhibit(i) option on all of the REDIRECTING statements before
2817 dialing the redirected-to party. You still have to set the
2818 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
2819 will update the redirecting-to presentation (COLR) when it becomes available.
2820 * Added the ability to ignore calls that are not in a Multiple Subscriber
2821 Number (MSN) list for PTMP CPE interfaces.
2822 * Added dynamic range compression support for dahdi channels. It is
2823 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
2824 * Added support for ISDN calling and called subaddress with partial support
2825 for connected line subaddress.
2826 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
2827 * Added handling of received HOLD/RETRIEVE messages and the optional ability
2828 to transfer a held call on disconnect similar to an analog phone.
2829 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
2830 Will reroute/deflect an outgoing call when receive the message.
2831 Can use the DAHDISendCallreroutingFacility to send the message for the
2833 * Added standard location to add options to chan_dahdi dialing:
2834 Dial(DAHDI/g1[/extension[/options]])
2837 R Reverse charging indication
2838 * Added Reverse Charging Indication (Collect calls) send/receive option.
2839 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
2840 Dial(DAHDI/g1/extension/R)
2841 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
2842 (requires latest LibPRI)
2843 * Added ability to send/receive keypad digits in the SETUP message.
2844 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
2845 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
2846 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
2847 (requires latest LibPRI)
2848 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
2849 to eliminate tromboned calls. A tromboned call goes out an interface and comes
2850 back into the same interface. Tromboned calls happen because of call routing,
2851 call deflection, call forwarding, and call transfer.
2852 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
2853 * Added the ability to support call waiting calls. (The SETUP has no B channel
2855 * Added Malicious Call ID (MCID) event to the AMI call event class.
2856 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
2858 Asterisk Manager Interface
2859 --------------------------
2860 * The Hangup action now accepts a Cause header which may be used to
2861 set the channel's hangup cause.
2862 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
2863 to specify a separate .pem file to hold a private key. By default sslcert
2864 is used to hold both the public and private key.
2865 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
2866 for options containing the 'tls' prefix. For example, 'sslenable' is now
2867 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
2868 across all .conf files. All affected sample.conf files have been modified to
2869 reflect this change. Previous options such as 'sslenable' still work,
2870 but options with the 'tls' prefix are preferred.
2871 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
2872 in a channel. (res_mutestream.so)
2873 * The configuration file manager.conf now supports a channelvars option, which
2874 specifies a list of channel variables to include in each channel-oriented
2876 * The redirect command now has new parameters ExtraContext, ExtraExtension,
2877 and ExtraPriority to allow redirecting the second channel to a different
2878 location than the first.
2879 * Added new event "JabberStatus" in the Jabber module to monitor buddies
2881 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
2882 in a MixMonitor recording.
2883 * The 'iax2 show peers' output is now similar to the expected output of
2885 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
2887 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
2888 AOC-E messages on a channel.
2889 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
2890 conform more closely to similar events.
2891 * Added a new eventfilter option per user to allow whitelisting and blacklisting
2893 * Added optional parkinglot variable for park command.
2894 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
2895 if CallerIDNum and CallerIDName headers are also present.
2897 Channel Event Logging
2898 ---------------------
2899 * A new interface, CEL, is introduced here. CEL logs single events, much like
2900 the AMI, but it differs from the AMI in that it logs to db backends much
2901 like CDR does; is based on the event subsystem introduced by Russell, and
2902 can share in all its benefits; allows multiple backends to operate like CDR;
2903 is specialized to event data that would be of concern to billing sytems,
2904 like CDR. Backends for logging and accounting calls have been produced,
2905 but a new CDR backend is still in development.
2909 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
2910 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
2911 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
2912 * Multiple files and formats can now be specified in cdr_custom.conf.
2913 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
2914 See configs/cdr_syslog.conf.sample for more information.
2915 * A 'sequence' field has been added to CDRs which can be combined with
2916 linkedid or uniqueid to uniquely identify a CDR.
2917 * Handling of billsec and duration field has changed. If your table definition
2918 specifies those fields as float,double or similar they will now be logged with
2919 microsecond accuracy instead of a whole integer.
2921 Calendaring for Asterisk
2922 ------------------------
2923 * A new set of modules were added supporing calendar integration with Asterisk.
2924 Dialplan functions for reading from and writing to calendars are included,
2925 as well as the ability to execute dialplan logic upon calendar event notifications.
2926 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
2927 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
2928 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
2929 2003 support does not support forms-based authentication).
2931 Call Completion Supplementary Services for Asterisk
2932 ---------------------------------------------------
2933 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
2934 DAHDI/ISDN supports call completion for the following switch types:
2935 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
2936 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
2938 Multicast RTP Support
2939 ---------------------
2940 * A new RTP engine and channel driver have been added which supports Multicast RTP.
2941 The channel driver can be used with the Page application to perform multicast RTP
2942 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
2943 Type can be either basic or linksys.
2944 Destination is the IP address and port for the RTP packets.
2945 Control address is specific to the linksys type and is used for sending the control
2946 packets unique to them.
2948 Security Events Framework
2949 -------------------------
2950 * Asterisk has a new C API for reporting security events. The module res_security_log
2951 sends these events to the "security" logger level. Currently, AMI is the only
2952 Asterisk component that reports security events. However, SIP support will be
2953 coming soon. For more information on the security events framework, see the
2954 "Asterisk Security Framework" section of the Asterisk wiki at
2955 https://wiki.asterisk.org/wiki/x/wgBQ
2956 * SIP support was added in Asterisk 10
2957 * This API now supports IPv6 addresses
2961 * A technology independent fax frontend (res_fax) has been added to Asterisk.
2962 * A spandsp based fax backend (res_fax_spandsp) has been added.
2963 * The app_fax module has been deprecated in favor of the res_fax module and
2964 the new res_fax_spandsp backend.
2965 * The SendFAX and ReceiveFAX applications now send their log messages to a
2966 'fax' logger level, instead of to the generic logger levels. To see these
2967 messages, the system's logger.conf file will need to direct the 'fax' logger
2968 level to one or more destinations; the logger.conf.sample file includes an
2969 example of how to do this. Note that if the 'fax' logger level is *not*
2970 directed to at least one destination, log messages generated by these
2971 applications will be lost, and that if the 'fax' logger level is directed to
2972 the console, the 'core set verbose' and 'core set debug' CLI commands will
2973 have no effect on whether the messages appear on the console or not.
2977 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
2978 Now, in order to enable transmitting silence during record the transmit_silence
2979 option should be used. transmit_silence_during_record remains a valid option, but
2980 defaults to the behavior of the transmit_silence option.
2981 * Addition of the Unit Test Framework API for managing registration and execution
2982 of unit tests with the purpose of verifying the operation of C functions.
2983 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
2984 XMPP text messages to the remote JID.
2985 * Modules.conf has a new option - "require" - that marks a module as critical for
2986 the execution of Asterisk.
2987 If one of the required modules fail to load, Asterisk will exit with a return
2989 * An 'X' option has been added to the asterisk application which enables #exec support.
2990 This allows #exec to be used in asterisk.conf.
2991 * jabber.conf supports a new option auth_policy that toggles auto user registration.
2992 * A new lockconfdir option has been added to asterisk.conf to protect the
2993 configuration directory (/etc/asterisk by default) during reloads.
2994 * The parkeddynamic option has been added to features.conf to enable the creation
2995 of dynamic parkinglots.
2996 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
2997 the reportalarms config option.
2998 * chan_dahdi supports dialing configuring and dialing by device file name.
2999 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
3000 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
3001 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
3002 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
3003 Handy for the above name-based syntax as it does not depend on
3004 initialization order.
3005 * The Realtime dialplan switch now caches entries for 1 second. This provides a
3006 significant increase in performance (about 3X) for installations using this switchtype.
3007 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
3008 AIS. For more information, please see the Distributed Device State section of the
3009 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
3010 * The addition of G.719 pass-through support.
3011 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
3012 during device configuration.
3013 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
3014 have less than 3 lines on the LCD.
3015 * Realtime now supports database failover. See the sample extconfig.conf for details.
3016 * The addition of improved translation path building for wideband codecs. Sample
3017 rate changes during translation are now avoided unless absolutely necessary.
3018 * The addition of the res_stun_monitor module for monitoring and reacting to network
3019 changes while behind a NAT.
3020 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
3021 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
3022 These allow support for any Administration. Default is AT&T values.
3026 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
3027 optionally accept a filename, to apply the setting only to the code generated from
3028 that source file when Asterisk was built. However, there are some modules in Asterisk
3029 that are composed of multiple source files, so this did not result in the behavior
3030 that users expected. In this version, 'core set debug' and 'core set verbose'
3031 can optionally accept *module* names instead (with or without the .so extension),
3032 which applies the setting to the entire module specified, regardless of which source
3033 files it was built from.
3034 * New 'manager show settings' command showing the current settings loaded from
3036 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
3037 the channel hangup request to all channels.
3038 * Added a "core reload" CLI command that executes a global reload of Asterisk.
3040 ------------------------------------------------------------------------------
3041 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
3042 ------------------------------------------------------------------------------
3046 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
3047 Snom phones use this for call pickup of extensions that the phone is
3049 * Added support for setting the domain in the URI for caller of an
3050 outbound call by using the SIPFROMDOMAIN channel variable.
3051 * Added a new configuration option "remotesecret" for authentication to
3052 remote services. For backwards compatibility, "secret" still has the
3053 same function as before, but now you can configure both a remote secret and a
3054 local secret for mutual authentication.
3055 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
3056 the sound will be played to the target of an attended transfer
3057 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
3058 finer control over how many peers Asterisk will qualify and the gap between them
3059 when all peers need to be qualified at the same time.
3060 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
3061 (either globally or for a specific peer), chan_sip will treat any SDP data
3062 it receives as new data and update the media stream accordingly. By
3063 default, Asterisk will only modify the media stream if the SDP session
3064 version received is different from the current SDP session version. This
3065 option is required to interoperate with devices that have non-standard SDP
3066 session version implementations (observed with Microsoft OCS). This option
3067 is disabled by default.
3068 * The parsing of register => lines in sip.conf has been modified to allow a port
3069 to be present in the "user" portion. Please see the sip.conf.sample file for more
3071 * Added support for subscribing to MWI on a remote server and making the status available
3072 as a mailbox. Please see the sip.conf.sample file for more information.
3073 * Added a function to remove SIP headers added in the dialplan before the
3074 first INVITE is generated - SIPRemoveHeader()
3075 * Channel variables set with setvar= in a device configuration is now
3076 set both for inbound and outbound calls.
3077 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
3081 * Added immediate option to iax.conf
3082 * Added forceencryption option to iax.conf
3083 * Added Encryption and Trunk status to manager command "iaxpeers"
3087 * The configuration file now holds separate sections for devices and lines.
3088 Please have a look at configs/skinny.conf.sample and change your skinny.conf
3093 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
3094 support for LibOpenR2. http://www.libopenr2.org/
3095 * The UK option waitfordialtone has been added for use with BT analog
3097 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
3098 is used in conjunction with the 'faxdetect' configuration option. When
3099 'faxbuffers' is used and fax tones are detected, the channel will dynamically
3100 switch to the configured faxbuffers policy. For example, to use 6 buffers
3101 and a 'full' buffer policy for a fax transmission, add:
3103 The faxbuffers configuration will be in affect until the call is torn down.
3104 * Added service message support for 4ESS/5ESS switches.
3108 * For DAHDI channels, the CHANNEL() dialplan function now
3109 supports changing the channel's buffer policy (for the current
3110 call only), using this syntax:
3112 exten => s,n,Set(CHANNEL(buffers)=6,full)
3114 This would change the channel to the 'full' buffer policy and
3115 6 (six) buffers. Possible options for this setting are the same
3116 as those in chan_dahdi.conf.
3117 * Added a new dialplan function, CURLOPT, which permits setting various
3118 options that may be useful with the CURL dialplan function, such as
3119 cookies, proxies, connection timeouts, passwords, etc.
3120 * Permit the syntax and synopsis fields of the corresponding dialplan
3121 functions to be individually set from func_odbc.conf.
3122 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
3123 * func_odbc now may specify an insert query to execute, when the write query
3124 affects 0 rows (usually indicating that no such row exists).
3125 * Added a new dialplan function, LISTFILTER, which permits removing elements
3126 from a set list, by name. Uses the same general syntax as the existing CUT
3127 and FIELDQTY dialplan functions, which also manage lists.
3128 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
3129 obtaining realtime data from the dialplan.
3130 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
3131 a subroutine when using the GoSub() and Return() applications.
3132 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
3133 of "core show function AUDIOHOOK_INHERIT" from the CLI
3134 * Added AES_ENCRYPT. For information on its use, please see the output
3135 of "core show function AES_ENCRYPT" from the CLI
3136 * Added AES_DECRYPT. For information on its use, please see the output
3137 of "core show function AES_DECRYPT" from the CLI
3138 * func_odbc now supports database transactions across multiple queries.
3142 * Scheduled meetme conferences may now have their end times extended by
3144 * app_authenticate now gives the ability to select a prompt other than
3146 * app_directory now pays attention to the searchcontexts setting in
3147 voicemail.conf and will look through all contexts, if no context is
3148 specified in the initial argument.
3149 * A new application, Originate, has been introduced, that allows asynchronous
3150 call origination from the dialplan.
3151 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
3152 in addition to the setting in the "general" context.
3153 * Added ConfBridge dialplan application which does conference bridges without
3154 DAHDI. For information on its use, please see the output of
3155 "core show application ConfBridge" from the CLI.
3159 * The Asterisk CLI has a new command, "channel redirect", which is similar in
3160 operation to the AMI Redirect action.
3161 * extensions.conf now allows you to use keyword "same" to define an extension
3162 without actually specifying an extension. It uses exactly the same pattern
3163 as previously used on the last "exten" line. For example:
3164 exten => 123,1,NoOp(something)
3165 same => n,SomethingElse()
3166 * musiconhold.conf classes of type 'files' can now use relative directory paths,
3167 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
3168 * All deprecated CLI commands are removed from the sourcecode. They are now handled
3169 by the new clialiases module. See cli_aliases.conf.sample file.
3170 * Times within timespecs are now accurate down to the minute. This is a change
3171 from historical Asterisk, which only provided timespecs rounded to the nearest
3172 even (read: evenly divisible by 2) minute mark.
3173 * The realtime switch now supports an option flag, 'p', which disables searches for
3175 * In addition to a time range and date range, timespecs now accept a 5th optional
3176 argument, timezone. This allows you to perform time checks on alternate
3177 timezones, especially if those daylight savings time ranges vary from your
3178 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
3180 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
3181 give you the correct output for an asterisk box behind nat. It will give you the
3182 externhost and localnet settings.
3183 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
3184 can connect calls in passthrough mode, as well as record and play back files.
3185 * Successful and unsuccessful call pickup can now be alerted through sounds, by
3186 using pickupsound and pickupfailsound in features.conf.
3187 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
3188 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
3189 instead of the /var/run/asterisk.pid where it used to be. This will make
3190 installs as non-root easier to manage.
3195 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
3196 be written; they will no longer be explicitly written.
3198 Asterisk Manager Interface
3199 --------------------------
3200 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
3201 a non-empty value) in your request. If you do this, any pending AMI events will
3202 *not* be included in the response to your request as they would normally, but
3203 will be left in the event queue for the next request you make to retrieve. For
3204 some applications, this will allow you to guarantee that you will only see
3205 events in responses to 'WaitEvent' actions, and can better know when to expect them.
3206 To know whether the Asterisk server supports this header or not, your client can
3207 inspect the first response back from the server to see if it includes this header:
3209 Pragma: SuppressEvents
3211 If this is included, the server supports event suppression.
3213 * Added 4 new Actions to list skinny device(s) and line(s)
3219 LDAP Schema File Additions
3220 --------------------------
3221 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
3222 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
3224 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
3225 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
3226 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
3227 * Removed redundant IPaddr (there's already IPAddress)
3228 - Gives more configuration Flags for SIP-Users available (tested)
3229 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
3230 without extensibleObject (which really should be the last resort); gives
3231 also additional possibilities for LDAP-filter
3233 ------------------------------------------------------------------------------
3234 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
3235 ------------------------------------------------------------------------------
3237 Device State Handling
3238 ---------------------
3239 * The event infrastructure in Asterisk got another big update to help support
3240 distributed events. It currently supports distributed device state and
3241 distributed Voicemail MWI (Message Waiting Indication). A new module has
3242 been merged, res_ais, which facilitates communicating events between servers.
3243 It uses the SAForum AIS (Service Availability Forum Application Interface
3244 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
3245 a cluster of Asterisk servers, and to share events between them. For more
3246 information on setting this up, refer to the Distributed Device State section
3247 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
3251 * Added a new dialplan function, AST_CONFIG(), which allows you to access
3252 variables from an Asterisk configuration file.
3253 * The JACK_HOOK function now has a c() option to supply a custom client name.
3254 * Added two new dialplan functions from libspeex for audio gain control and
3255 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
3256 rx directions of a channel from the dialplan.
3257 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
3258 based on other parameters. The default is still to search based on the
3259 forwarding station ID. However, there are new options that allow you to search
3260 based on the message desk terminal ID, or the message desk number.
3261 * TIMEOUT() has been modified to be accurate down to the millisecond.
3262 * ENUM*() functions now include the following new options:
3263 - 'u' returns the full URI and does not strip off the URI-scheme.
3264 - 's' triggers ISN specific rewriting
3265 - 'i' looks for branches into an Infrastructure ENUM tree
3266 - 'd' for a direct DNS lookup without any flipping of digits.
3267 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
3268 * CHANNEL() now has options for the maximum, minimum, and standard or normal
3269 deviation of jitter, rtt, and loss for a call using chan_sip.
3271 DAHDI channel driver (chan_dahdi) Changes
3272 ----------------------------------------
3273 * Channels can now be configured using named sections in chan_dahdi.conf, just
3274 like other channel drivers, including the use of templates.
3275 * The default for pridialplan has changed from 'national' to 'unknown'.
3279 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
3280 to something that matches the pattern a hint will be created using the contents
3281 and variables evaluated.
3282 * Dialplan matching has been extended to allow an extension to return to the
3283 PBX core to wait for more digits. This is done by using the new dialplan
3284 application called "Incomplete". This will permit a whole new level of
3285 extension control, by giving the administrator more control over early
3286 matches employing one of the short-circuit pattern match operators. Note
3287 that custom applications can trigger this same behavior by returning the
3288 special value AST_PBX_INCOMPLETE.
3292 * Directory now permits both first and last names to be matched at the same
3293 time. In addition, the number of digits to enter of the name can be set in
3294 the arguments to Directory; previously, you could enter only 3, regardless
3295 of how many names are in your company. For large companies, this should be
3297 * Voicemail now permits a mailbox setting to wrap around from first to last
3298 messages, if the "messagewrap" option is set to a true value.
3299 * Voicemail now permits an external script to be run, for password validation.
3300 The script should output "VALID" or "INVALID" on stdout, depending upon the
3301 wish to validate or invalidate the password given. Arguments are:
3302 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
3304 * Dial has a new option: F(context^extension^pri), which permits a callee to
3305 continue in the dialplan, at the specified label, if the caller hangs up.
3306 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
3307 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
3308 * The Jack application now has a c() option to supply a custom client name.
3309 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
3310 like the pre-existing whisper mode, except that the spy can also talk to the
3311 participant on the bridged channel as well.
3312 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
3313 to be spoken instead of the channel name or number. For more information on the
3314 use of this option, issue the command "core show application ChanSpy" from the
3316 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
3317 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
3318 words, if using the 'd' option, it is not possible to enter a number to append to
3319 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
3320 change to whisper mode, and pressing 6 will change to barge mode.
3321 * ExternalIVR now takes several options that affect the way it performs, as
3322 well as having several new commands. Please see the External IVR page on the Asterisk
3323 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
3324 * Added ability to communicate over a TCP socket instead of forking a child process for the
3325 ExternalIVR application.
3326 * ChanIsAvail has a new option, 'a', which will return all available channels instead
3327 of just the first one if you give the function more then one channel to check.
3328 * PrivacyManager now takes an option where you can specify a context where the
3329 given number will be matched. This way you have more control over who is allowed
3330 and it stops the people who blindly enter 10 digits.
3331 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
3332 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
3333 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
3334 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
3335 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
3336 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
3337 * The Dial() application no longer copies the language used by the caller to the callee's
3338 channel. If you desire for the caller's channel's language to be used for file playback
3339 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
3340 * SendImage() no longer hangs up the channel on error; instead, it sets the
3341 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
3342 'UNSUPPORTED'. This change makes SendImage() more consistent with other
3344 * Park has a new option, 's', which silences the announcement of the parking space number.
3345 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
3346 invalid input and will be assumed to mean that no timeout is desired.
3350 * Added DNS manager support to registrations for peers referencing peer entries.
3351 DNS manager runs in the background which allows DNS lookups to be run asynchronously
3352 as well as periodically updating the IP address. These properties allow for
3353 better performance as well as recovery in the event of an IP change.
3354 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
3355 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
3356 These changes also provide performance improvements for call setup and tear down.
3357 * Added ability to specify registration expiry time on a per registration basis in
3359 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
3361 * Added t38pt_usertpsource option. See sip.conf.sample for details.
3362 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
3363 * 'sip show peers' and 'sip show users' display their entries sorted in
3364 alphabetical order, as opposed to the order they were in, in the config
3366 * Videosupport now supports an additional option, "always", which always sets
3367 up video RTP ports, even on clients that don't support it. This helps with
3368 callfiles and certain transfers to ensure that if two video phones are
3369 connected, they will always share video feeds.
3373 * Existing DNS manager lookups extended to check for SRV records.
3374 * IAX2 encryption support has been improved to support periodic key rotation
3375 within a call for enhanced security. The option "keyrotate" has been
3376 provided to disable this functionality to preserve backwards compatibility
3377 with older versions of IAX2 that do not support key rotation.
3381 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
3382 data tree based on the given <path>.
3383 * New CLI command "data show providers" that will display all the registered
3385 * New CLI command, "config reload <file.conf>" which reloads any module that
3386 references that particular configuration file. Also added "config list"
3387 which shows which configuration files are in use.
3388 * New CLI commands, "pri show version" and "ss7 show version" that will
3389 display which version of libpri and libss7 are being used, respectively.
3390 A new API call was added so trunk will now have to be compiled against
3391 a versions of libpri and libss7 that have them or it will not know that
3392 these libraries exist.
3393 * The commands "core show globals", "core set global" and "core set chanvar" has
3394 been deprecated in favor of the more semanticly correct "dialplan show globals",
3395 "dialplan set chanvar" and "dialplan set global".
3396 * New CLI command "dialplan show chanvar" to list all variables associated
3397 with a given channel.
3401 * Addresses managed by DNS manager now can check to see if there is a DNS
3402 SRV record for a given domain and will use that hostname/port if present.
3404 AMI - The manager (TCP/TLS/HTTP)
3405 --------------------------------
3406 * The Status command now takes an optional list of variables to display
3407 along with channel status.
3408 * The QueueEntry event now also includes the channel's uniqueid
3412 * res_odbc no longer has a limit of 1023 total possible unshared connections,
3413 as some people were running into this limit. This limit has been increased
3418 * The TRANSFER queue log entry now includes the the caller's original
3419 position in the transferred-from queue.
3420 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
3421 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
3422 as well as an explanation about timeout options in general
3423 * Added a new option - C - for forcing the "answered elsewhere" flag on
3424 cancellation of calls in to members of the queue. This is to avoid the
3425 call to a member of a queue having the call listed as a "missed call".
3429 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
3430 adaptive capabilities. What this means in practical terms is that if your
3431 realtime table lacks critical fields, Asterisk will now emit warnings to
3432 that effect. Also, some of the realtime drivers have the ability (if
3433 configured) to automatically add those columns to the table with the
3434 correct type and length.
3438 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
3439 the 'setvar' option to cause a given audio file to be played upon completion
3440 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
3441 Skinny channels only.
3442 * You can now compile Asterisk against the Hoard Memory Allocator, see the
3443 Hoard page on the Asterisk wiki for more information:
3444 https://wiki.asterisk.org/wiki/x/pQBB
3445 * Config file variables may now be appended to, by using the '+=' append
3446 operator. This is most helpful when working with long SQL queries in
3447 func_odbc.conf, as the queries no longer need to be specified on a single
3449 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
3450 which will add a second to the billsec when the ending
3451 time is set, if the number in the microseconds field of the end time is
3452 greater than the number of microseconds in the answer time. This allows
3453 users to count the 'initiated' seconds in their billing records.
3455 ------------------------------------------------------------------------------
3456 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
3457 ------------------------------------------------------------------------------
3459 AMI - The manager (TCP/TLS/HTTP)
3460 --------------------------------
3461 * Manager has undergone a lot of changes, all of them documented
3462 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
3463 * Manager version has changed to 1.1
3464 * Added a new action 'CoreShowChannels' to list currently defined channels
3465 and some information about them.
3466 * Added a new action 'SIPshowregistry' to list SIP registrations.
3467 * Added TLS support for the manager interface and HTTP server
3468 * Added the URI redirect option for the built-in HTTP server
3469 * The output of CallerID in Manager events is now more consistent.
3470 CallerIDNum is used for number and CallerIDName for name.
3471 * Enable https support for builtin web server.
3472 See configs/http.conf.sample for details.
3473 * Added a new action, GetConfigJSON, which can return the contents of an
3474 Asterisk configuration file in JSON format. This is intended to help
3475 improve the performance of AJAX applications using the manager interface
3477 * SIP and IAX manager events now use "ChannelType" in all cases where we
3478 indicate channel driver. Previously, we used a mixture of "Channel"
3479 and "ChannelDriver" headers.
3480 * Added a "Bridge" action which allows you to bridge any two channels that
3481 are currently active on the system.
3482 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
3483 the voicemail users setup.
3484 * Added 'DBDel' and 'DBDelTree' manager commands.
3485 * cdr_manager now reports events via the "cdr" level, separating it from
3486 the very verbose "call" level.
3487 * Manager users are now stored in memory. If you change the manager account
3488 list (delete or add accounts) you need to reload manager.
3489 * Added Masquerade manager event for when a masquerade happens between
3491 * Added "manager reload" command for the CLI
3492 * Lots of commands that only provided information are now allowed under the
3493 Reporting privilege, instead of only under Call or System.
3494 * The IAX* commands now require either System or Reporting privilege, to
3495 mirror the privileges of the SIP* commands.
3496 * Added ability to retrieve list of categories in a config file.
3497 * Added ability to retrieve the content of a particular category.
3498 * Added ability to empty a context.
3499 * Created new action to create a new file.
3500 * Updated delete action to allow deletion by line number with respect to category.
3501 * Added new action insert to add new variable to category at specified line.
3502 * Updated action newcat to allow new category to be inserted in file above another
3504 * Added new event "JitterBufStats" in the IAX2 channel
3505 * Originate now requires the Originate privilege and, if you want to call out
3506 to a subshell, it requires the System privilege, as well. This was done to
3507 enhance manager security.
3508 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
3509 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
3510 or manager show command Atxfer from the CLI
3511 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
3512 details or manager show command IAXregistry from the CLI
3516 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
3517 state in the dialplan, as well as creating custom device states that are
3518 controllable from the dialplan.
3519 * Extend CALLERID() function with "pres" and "ton" parameters to
3520 fetch string representation of calling number presentation indicator
3521 and numeric representation of type of calling number value.
3522 * MailboxExists converted to dialplan function
3523 * A new option to Dial() for telling IP phones not to count the call
3524 as "missed" when dial times out and cancels.
3525 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
3526 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
3527 held for any given channel. Also, locks are automatically freed when a
3529 * Added HINT() dialplan function that allows retrieving hint information.
3530 Hints are mappings between extensions and devices for the sake of
3531 determining the state of an extension. This function can retrieve the list
3532 of devices or the name associated with a hint.
3533 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
3535 * Added SYSINFO() dialplan function which allows retrieval of system information
3536 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
3537 the existence of a dialplan target.
3538 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
3539 upper and lower case, respectively.
3540 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
3541 ID for the call (not the Asterisk call ID or unique ID), provided that the
3542 channel driver supports this. For SIP, you get the SIP call-ID for the
3543 bridged channel which you can store in the CDR with a custom field.
3547 * Added CLI permissions, config file: cli_permissions.conf
3548 default is to allow all commands for every local user/group.
3549 Also this new feature added three new CLI commands:
3550 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
3551 - cli reload permissions
3552 - cli show permissions
3553 * New CLI command "core show hint" (usage: core show hint <exten>)
3554 * New CLI command "core show settings"
3555 * Added 'core show channels count' CLI command.
3556 * Added the ability to set the core debug and verbose values on a per-file basis.
3557 * Added 'queue pause member' and 'queue unpause member' CLI commands
3558 * Ability to set process limits ("ulimit") without restarting Asterisk
3559 * Enhanced "agi debug" to print the channel name as a prefix to the debug
3560 output to make debugging on busy systems much easier.
3561 * New CLI commands "dialplan set extenpatternmatching true/false"
3562 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
3563 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
3564 listed in the startup_commands section of cli.conf will get executed.
3565 * Added a CLI command, "devstate change", which allows you to set custom device
3566 states from the func_devstate module that provides the DEVICE_STATE() function
3567 and handling of the "Custom:" devices.
3568 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
3569 sorted into the different possible callbacks, with the number of entries
3570 currently scheduled for each. Gives you a feel for how busy the sip channel
3572 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
3573 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
3574 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
3578 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
3579 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
3580 for a received call. If it is detected, the channel will jump to the
3581 'fax' extension in the dialplan.
3582 * The default SIP useragent= identifier now includes the Asterisk version
3583 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
3584 If set, and the incoming request carries authentication info,
3585 the username to match in the users list is taken from the Digest header
3586 rather than from the From: field. This feature is considered experimental.
3587 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
3588 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
3589 * The "localmask" setting was removed in version 1.2 and the reminder about it
3590 being removed is now also removed.
3591 * A new option "busylevel" for setting a level of calls where asterisk reports
3592 a device as busy, to separate it from call-limit. This value is also added
3593 to the SIP_PEER dialplan function.
3594 * A new realtime family called "sipregs" is now supported to store SIP registration
3595 data. If this family is defined, "sippeers" will be used for configuration and
3596 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
3597 registration data, as before.
3598 * The SIPPEER function have new options for port address, call and pickup groups
3599 * Added support for T.140 realtime text in SIP/RTP
3600 * The "checkmwi" option has been removed from sip.conf, as it is no longer
3601 required due to the restructuring of how MWI is handled. See the descriptions
3602 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
3603 for more information.
3604 * Added rtpdest option to CHANNEL() dialplan function.
3605 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
3606 * SIP now adds a header to the CANCEL if the call was answered by another phone
3607 in the same dial command, or if the new c option in dial() is used.
3608 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
3609 states it is not needed. For phones, however, that do require it the "registertrying" option
3610 has been added so it can be enabled.
3611 * A new option called "callcounter" (global/peer/user level) enables call counters needed
3612 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
3613 used to enable this functionality).
3614 * New settings for timer T1 and timer B on a global level or per device. This makes it
3615 possible to force timeout faster on non-responsive SIP servers. These settings are
3616 considered advanced, so don't use them unless you have a problem.
3617 * Added a dial string option to be able to set the To: header in an INVITE to any
3619 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
3620 the qualify frequency.
3621 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
3622 were not properly torn down due to network or endpoint failures during an established
3624 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
3625 and configs/sip.conf.sample for more information on how it is used.
3626 * Added a new configuration option "authfailureevents" that enables manager events when
3627 a peer can't authenticate properly.
3628 * Added DNS manager support to registrations for peers not referencing a peer entry.
3632 * Added the trunkmaxsize configuration option to chan_iax2.
3633 * Added the srvlookup option to iax.conf
3634 * Added support for OSP. The token is set and retrieved through the CHANNEL()
3637 XMPP Google Talk/Jingle changes
3638 -------------------------------
3639 * Added the bindaddr option to gtalk.conf.
3643 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
3644 * Proper codec support in chan_skinny.
3645 * Added settings for IP and Ethernet QoS requests
3649 * Added separate settings for media QoS in mgcp.conf
3651 Console Channel Driver changes
3652 ------------------------------
3653 * Added experimental support for video send & receive to chan_oss.
3654 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
3657 Phone channel changes (chan_phone)
3658 ----------------------------------
3659 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
3661 H.323 channel Changes
3662 ---------------------
3663 * H323 remote hold notification support added (by NOTIFY message
3664 and/or H.450 supplementary service)
3666 Local channel changes
3667 ---------------------
3668 * The device state functionality in the Local channel driver has been updated
3669 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
3670 to just UNKNOWN if the extension exists.
3671 * Added jitterbuffer support for chan_local. This allows you to use the
3672 generic jitterbuffer on incoming calls going to Asterisk applications.
3673 For example, this would allow you to use a jitterbuffer for an incoming
3674 SIP call to Voicemail by putting a Local channel in the middle. This
3675 feature is enabled by using the 'j' option in the Dial string to the Local
3676 channel in conjunction with the existing 'n' option for local channels.
3677 * A 'b' option has been added which causes chan_local to return the actual channel
3678 that is behind it when queried. This is useful for transfer scenarios as the
3679 actual channel will be transferred, not the Local channel.
3681 Agent channel changes
3682 ----------------------
3683 * The ackcall and endcall options are now supplemented with options acceptdtmf
3684 and enddtmf. These allow for the DTMF keypress to be configurable. The options
3685 default to their old hard-coded values ('#' and '*' respectively) so this should
3686 not break any existing agent installations.
3688 DAHDI channel driver (chan_dahdi) Changes
3689 ----------------------------------------
3690 * SS7 support (via libss7 library)
3691 * In India, some carriers transmit CID via dtmf. Some code has been added
3692 that will handle some situations. The cidstart=polarity_IN choice has been added for
3693 those carriers that transmit CID via dtmf after a polarity change.
3694 * CID matching information is now shown when doing 'dialplan show'.
3695 * Added dahdi show version CLI command.
3696 * Added setvar support to chan_dahdi.conf channel entries.
3697 * Added two new options: mwimonitor and mwimonitornotify. These options allow
3698 you to enable MWI monitoring on FXO lines. When the MWI state changes,
3699 the script specified in the mwimonitornotify option is executed. An internal
3700 event indicating the new state of the mailbox is also generated, so that
3701 the normal MWI facilities in Asterisk work as usual.
3702 * Added signalling type 'auto', which attempts to use the same signalling type
3703 for a channel as configured in DAHDI. This is primarily designed for analog
3704 ports, but will also work for digital ports that are configured for FXS or FXO
3705 signalling types. This mode is also the default now, so if your chan_dahdi.conf
3706 does not specify signalling for a channel (which is unlikely as the sample
3707 configuration file has always recommended specifying it for every channel) then
3708 the 'auto' mode will be used for that channel if possible.
3709 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
3710 state for a channel; also ensured that the DNDState Manager event is
3711 emitted no matter how the DND state is set or cleared.
3715 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
3716 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
3717 for details. This new channel driver allows you to use Nortel i2002,
3718 i2004, and i2050 phones with Asterisk.
3719 * Added a new channel driver, chan_console, which uses portaudio as a cross
3720 platform audio interface. It was written as a channel driver that would
3721 work with Mac CoreAudio, but portaudio supports a number of other audio
3722 interfaces, as well. Note that this channel driver requires v19 or higher
3723 of portaudio; older versions have a different API.
3727 * Added the ability to specify arguments to the Dial application when using
3728 the DUNDi switch in the dialplan.
3729 * Added the ability to set weights for responses dynamically. This can be
3730 done using a global variable or a dialplan function. Using the SHELL()
3731 function would allow you to have an external script set the weight for
3733 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
3734 functions will allow you to initiate a DUNDi query from the dialplan,
3735 find out how many results there are, and access each one.
3736 * Added the ability to specifiy a port for a dundi peer.
3740 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
3741 functions will allow you to initiate an ENUM lookup from the dialplan,
3742 and Asterisk will cache the results. ENUMRESULT can be used to access
3743 the results without doing multiple DNS queries.
3747 * Added the ability to customize which sound files are used for some of the
3748 prompts within the Voicemail application by changing them in voicemail.conf
3749 * Added the ability for the "voicemail show users" CLI command to show users
3750 configured by the dynamic realtime configuration method.
3751 * MWI (Message Waiting Indication) handling has been significantly
3752 restructured internally to Asterisk. It is now totally event based
3753 instead of polling based. The voicemail application will notify other
3754 modules that have subscribed to MWI events when something in the mailbox
3756 This also means that if any other entity outside of Asterisk is changing
3757 the contents of mailboxes, then the voicemail application still needs to
3758 poll for changes. Examples of situations that would require this option
3759 are web interfaces to voicemail or an email client in the case of using
3760 IMAP storage. So, two new options have been added to voicemail.conf
3761 to account for this: "pollmailboxes" and "pollfreq". See the sample
3762 configuration file for details.
3763 * Added "tw" language support
3764 * Added support for storage of greetings using an IMAP server
3765 * Added ability to customize forward, reverse, stop, and pause keys for message playback
3766 * SMDI is now enabled in voicemail using the smdienable option.
3767 * A "lockmode" option has been added to asterisk.conf to configure the file
3768 locking method used for voicemail, and potentially other things in the
3769 future. The default is the old behavior, lockfile. However, there is a
3770 new method, "flock", that uses a different method for situations where the
3771 lockfile will not work, such as on SMB/CIFS mounts.
3772 * Added the ability to backup deleted messages, to ease recovery in the case
3773 that a user accidentally deletes a message, and discovers that they need it.
3774 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
3775 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
3776 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
3777 voicemail boxes. The SMDI interface can also poll for MWI changes when some
3778 outside entity is modifying the state of the mailbox (such as IMAP storage or
3779 a web interface of some kind).
3780 * Added the support for marking messages as "urgent." There are two methods to accomplish
3781 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
3782 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
3783 the message as urgent after he has recorded a voicemail by following the voice instructions.
3784 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
3789 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
3790 used across multiple queues.
3791 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
3792 setqueueentryvar options for each queue, see queues.conf.sample for details.
3793 * Added keepstats option to queues.conf which will keep queue
3794 statistics during a reload.
3795 * setinterfacevar option in queues.conf also now sets a variable
3796 called MEMBERNAME which contains the member's name.
3797 * Added 'Strategy' field to manager event QueueParams which represents
3798 the queue strategy in use.
3799 * Added option to run macro when a queue member is connected to a caller,
3800 see queues.conf.sample for details.
3801 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
3802 does not count paused queue members as unavailable.
3803 * Added min-announce-frequency option to queues.conf which allows you to control the
3804 minimum amount of time between queue announcements for use when the caller's queue
3805 position changes frequently.
3806 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
3808 * Added ability for non-realtime queues to have realtime members
3809 * Added the "linear" strategy to queues.
3810 * Added the "wrandom" strategy to queues.
3811 * Added new channel variable QUEUE_MIN_PENALTY
3812 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
3813 rules in queuerules.conf. See configs/queuerules.conf.sample for details
3814 * Added a new parameter for member definition, called state_interface. This may be
3815 used so that a member may be called via one interface but have a different interface's
3816 device state reported.
3817 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
3818 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
3819 "manager show command QueueReset."
3820 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
3821 specified by the periodic-announce option, then one will be chosen randomly when it is time
3822 to play a periodic announcment
3823 * New configuration options: announce-position now takes two more values in addition to "yes" and
3824 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
3825 announce-position-limit. By setting announce-position to "limit" callers will only have their
3826 position announced if their position is less than what is specified by announce-position-limit.
3827 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
3828 will be told that their are more than announce-position-limit callers waiting.
3829 * Two new queue log events have been added. An ADDMEMBER event will be logged
3830 when a realtime queue member is added and a REMOVEMEMBER event will be logged
3831 when a realtime queue member is removed. Since there is no calling channel associated
3832 with these events, the string "REALTIME" is placed where the channel's unique id
3833 is typically placed.
3834 * The configuration method for the "joinempty" and "leavewhenempty" options has
3835 changed to a comma-separated list of methods of determining member availability
3836 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
3837 values are still accepted for backwards-compatibility, though.
3838 * The average talktime is now calculated on queues. This information is reported via the
3839 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
3840 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
3845 * The 'o' option to provide an optimization has been removed and its functionality
3846 has been enabled by default.
3847 * When a conference is created, the UNIQUEID of the channel that caused it to be
3848 created is stored. Then, every channel that joins the conference will have the
3849 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
3850 callers that come and go from long standing conferences.
3851 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
3852 except it does operations on a channel by name, instead of number in a conference.
3853 This is a very useful feature in combination with the 'X' option to ChanSpy.
3854 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
3856 * Added new RealTime functionality to provide support for scheduled conferencing.
3857 This includes optional messages to the caller if they attempt to join before
3858 the schedule start time, or to allow the caller to join the conference early.
3859 Also included is optional support for limiting the number of callers per
3860 RealTime conference.
3861 * Added the S() and L() options to the MeetMe application. These are pretty
3862 much identical to the S() and L() options to Dial(). They let you set
3863 timeouts for the conference, as well as have warning sounds played to
3864 let the caller know how much time is left, and when it is running out.
3865 * Added the ability to do "meetme concise" with the "meetme" CLI command.
3866 This extends the concise capabilities of this CLI command to include
3867 listing all conferences, instead of an addition to the other sub commands
3868 for the "meetme" command.
3869 * Added the ability to specify the music on hold class used to play into the
3870 conference when there is only one member and the M option is used.
3871 * Added MEETME_INFO dialplan function which provides a way to query
3872 various properties of a Meetme conference.
3873 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
3874 and *84: record in-conf
3876 Other Dialplan Application Changes
3877 ----------------------------------
3878 * Argument support for Gosub application
3879 * From the to-do lists: straighten out the app timeout args:
3880 Wait() app now really does 0.3 seconds- was truncating arg to an int.
3881 WaitExten() same as Wait().
3882 Congestion() - Now takes floating pt. argument.
3883 Busy() - now takes floating pt. argument.
3884 Read() - timeout now can be floating pt.
3885 WaitForRing() now takes floating pt timeout arg.
3886 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
3887 * Added 's' option to Page application.
3888 * Added an optional timeout argument to the Page application.
3889 * Added 'E', 'V', and 'P' commands to ExternalIVR.
3890 * Added 'o' and 'X' options to Chanspy.
3891 * Added a new dialplan application, Bridge, which allows you to bridge the
3892 calling channel to any other active channel on the system.
3893 * Added the ability to specify a music on hold class to play instead of ringing
3894 for the SLATrunk application.
3895 * The Read application no longer exits the dialplan on error. Instead, it sets
3896 READSTATUS to ERROR, which you can catch and handle separately.
3897 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
3898 of asking for verification of each name, one at a time.
3899 * Privacy() no longer uses privacy.conf, as all options are specifyable as
3900 direct options to the app.
3901 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
3903 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
3904 * The ChannelRedirect application no longer exits the dialplan if the given channel
3905 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
3906 or NOCHANNEL if the given channel was not found.
3907 * The silencethreshold setting that was previously configurable in multiple
3908 applications is now settable globally via dsp.conf.
3910 Music On Hold Changes
3911 ---------------------
3912 * A new option, "digit", has been added for music on hold classes in
3913 musiconhold.conf. If this is set for a music on hold class, a caller
3914 listening to music on hold can press this digit to switch to listening
3915 to this music on hold class.
3916 * Support for realtime music on hold has been added.
3917 * In conjunction with the realtime music on hold, a general section has
3918 been added to musiconhold.conf, its sole variable is cachertclasses. If this
3919 is set, then music on hold classes found in realtime will be cached in memory.
3923 * AEL upgraded to use the Gosub with Arguments instead
3924 of Macro application, to hopefully reduce the problems
3925 seen with the artificially low stack ceiling that
3926 Macro bumps into. Macros can only call other Macros
3927 to a depth of 7. Tests run using gosub, show depths
3928 limited only by virtual memory. A small test demonstrated
3929 recursive call depths of 100,000 without problems.
3930 -- in addition to this, all apps that allowed a macro
3931 to be called, as in Dial, queues, etc, are now allowing
3932 a gosub call in similar fashion.
3933 * AEL now generates LOCAL(argname) declarations when it
3934 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
3935 etc. That makes the arguments local in scope. The user
3936 can define their own local variables in macros, now,
3937 by saying "local myvar=someval;" or using Set() in this
3938 fashion: Set(LOCAL(myvar)=someval); ("local" is now
3940 * utils/conf2ael introduced. Will convert an extensions.conf
3941 file into extensions.ael. Very crude and unfinished, but
3942 will be improved as time goes by. Should be useful for a
3943 first pass at conversion.
3944 * aelparse will now read extensions.conf to see if a referenced
3945 macro or context is there before issueing a warning.
3946 * AEL parser sets a local channel variable ~~EXTEN~~, to
3947 preserve the value of ${EXTEN} thru switch statements.
3948 * New operator in $[...] expressions: the ~~ operator serves
3949 as a concatenation operator. AT THE MOMENT, it is really only
3950 necessary and useful in AEL, especially in if() expressions.
3951 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
3952 any enclosing double-quotes, and evaluate to the value of a
3953 concatenated with the value of b. For example if a is set to
3954 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
3955 evaluate to xyzabc .
3958 Call Features (res_features) Changes
3959 ------------------------------------
3960 * Added the parkedcalltransfers option to features.conf
3961 * Added parkedcallparking option to control one touch parking w/ parking
3963 * Added parkedcallhangup option to control disconnect feature w/ parking
3965 * Added parkedcallrecording option to control one-touch record w/ parking
3967 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
3968 parkedcalltransfers option support for multiple parking lots.
3969 * Added BRIDGE_FEATURES variable to set available features for a channel
3970 * The built-in method for doing attended transfers has been updated to
3971 include some new options that allow you to have the transferee sent
3972 back to the person that did the transfer if the transfer is not successful.
3973 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
3974 in features.conf.sample.
3975 * Added support for configuring named groups of custom call features in
3976 features.conf. This means that features can be written a single time, and
3977 then mapped into groups of features for different key mappings or easier
3979 * Updated the ParkedCall application to allow you to not specify a parking
3980 extension. If you don't specify a parking space to pick up, it will grab
3981 the first one available.
3982 * Added cli command 'features reload' to reload call features from features.conf
3983 * Moved into core asterisk binary.
3984 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
3985 * Added the ability for custom parking lots to be configured with their own
3986 parking extension with the parkext option.
3988 Language Support Changes
3989 ------------------------
3990 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
3991 * Added support for the Hungarian language for saying numbers, dates, and times.
3995 * Added SPEECH commands for speech recognition. A complete listing can be found
3997 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
3998 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
3999 does not behave as expected; the native command needs to be used, instead.
4000 * Added the ability to perform SRV lookups on fast AGI calls. To use this
4001 feature, simply use hagi: instead of agi: as the protocol portion
4002 of the URI parameter to the AGI function call in your dial plan. Also note
4003 that specifying a port number in the AGI URI will disable SRV lookups,
4004 even if you use the hagi: protocol.
4005 * No longer support MSG_OOB flag on HANGUP.
4009 * Added rotatestrategy option to logger.conf, along with two new options:
4010 "timestamp" which will use the time to name the logger files instead of
4011 sequence number; and "rotate", which rotates the names of the log files,
4012 similar to the way syslog rotates files.
4013 * Added exec_after_rotate option to logger.conf, which allows a system
4014 command to be run after rotation. This is primarily useful with
4015 rotatestrategy=rotate, to allow a limit on the number of log files kept
4016 and to ensure that the oldest log file gets deleted.
4017 * Added realtime support for the queue log
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